diff options
Diffstat (limited to 'third_party/libwebrtc/modules')
243 files changed, 2600 insertions, 4488 deletions
diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build index 347559a342..dfff987043 100644 --- a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build +++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/BUILD.gn b/third_party/libwebrtc/modules/audio_coding/BUILD.gn index 3e4d7e0c25..ddd1fd2656 100644 --- a/third_party/libwebrtc/modules/audio_coding/BUILD.gn +++ b/third_party/libwebrtc/modules/audio_coding/BUILD.gn @@ -618,7 +618,6 @@ rtc_library("audio_network_adaptor") { "../../common_audio", "../../logging:rtc_event_audio", "../../rtc_base:checks", - "../../rtc_base:ignore_wundef", "../../rtc_base:logging", "../../rtc_base:protobuf_utils", "../../rtc_base:safe_conversions", @@ -957,7 +956,6 @@ rtc_library("audio_coding_modules_tests_shared") { "../../api/audio_codecs:builtin_audio_encoder_factory", "../../api/neteq:neteq_api", "../../rtc_base:checks", - "../../rtc_base:ignore_wundef", "../../rtc_base:ssl", "../../rtc_base:stringutils", "../../system_wrappers", @@ -1644,6 +1642,7 @@ if (rtc_include_tests) { "neteq/mock/mock_expand.h", "neteq/mock/mock_histogram.h", "neteq/mock/mock_neteq_controller.h", + "neteq/mock/mock_packet_arrival_history.h", "neteq/mock/mock_packet_buffer.h", "neteq/mock/mock_red_payload_splitter.h", "neteq/mock/mock_statistics_calculator.h", @@ -1717,7 +1716,6 @@ if (rtc_include_tests) { "../../logging:rtc_event_audio", "../../modules/rtp_rtcp:rtp_rtcp_format", "../../rtc_base:checks", - "../../rtc_base:ignore_wundef", "../../rtc_base:macromagic", "../../rtc_base:platform_thread", "../../rtc_base:refcount", diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc index 210244154a..2d9ea91106 100644 --- a/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -707,7 +707,7 @@ class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {}; TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); - Run(/*audio_checksum_ref=*/"69118ed438ac76252d023e0463819471", + Run(/*audio_checksum_ref=*/"3e43fd5d3c73a59e8118e68fbfafe2c7", /*payload_checksum_ref=*/"c1edd36339ce0326cc4550041ad719a0", /*expected_packets=*/100, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -715,7 +715,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160)); - Run(/*audio_checksum_ref=*/"f95c87bdd33f631bcf80f4b19445bbd2", + Run(/*audio_checksum_ref=*/"608750138315cbab33d76d38e8367807", /*payload_checksum_ref=*/"ad786526383178b08d80d6eee06e9bad", /*expected_packets=*/100, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -723,7 +723,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320)); - Run(/*audio_checksum_ref=*/"c50244419c5c3a2f04cc69a022c266a2", + Run(/*audio_checksum_ref=*/"02e9927ef5e4d2cd792a5df0bdee5e19", /*payload_checksum_ref=*/"5ef82ea885e922263606c6fdbc49f651", /*expected_packets=*/100, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -731,7 +731,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80)); - Run(/*audio_checksum_ref=*/"4fccf4cc96f1e8e8de4b9fadf62ded9e", + Run(/*audio_checksum_ref=*/"4ff38de045b19f64de9c7e229ba36317", /*payload_checksum_ref=*/"62ce5adb0d4965d0a52ec98ae7f98974", /*expected_packets=*/100, /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput); @@ -739,7 +739,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160)); - Run(/*audio_checksum_ref=*/"e15e388d9d4af8c02a59fe1552fedee3", + Run(/*audio_checksum_ref=*/"1ee35394cfca78ad6d55468441af36fa", /*payload_checksum_ref=*/"41ca8edac4b8c71cd54fd9f25ec14870", /*expected_packets=*/100, /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput); @@ -747,7 +747,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320)); - Run(/*audio_checksum_ref=*/"b240520c0d05003fde7a174ae5957286", + Run(/*audio_checksum_ref=*/"19cae34730a0f6a17cf4e76bf21b69d6", /*payload_checksum_ref=*/"50e58502fb04421bf5b857dda4c96879", /*expected_packets=*/100, /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput); @@ -763,7 +763,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160)); - Run(/*audio_checksum_ref=*/"47eb60e855eb12d1b0e6da9c975754a4", + Run(/*audio_checksum_ref=*/"ae259cab624095270b7369e53a7b53a3", /*payload_checksum_ref=*/"6ad745e55aa48981bfc790d0eeef2dd1", /*expected_packets=*/50, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -779,7 +779,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160)); - Run(/*audio_checksum_ref=*/"a84d75e098d87ab6b260687eb4b612a2", + Run(/*audio_checksum_ref=*/"f2e81d2531a805c40e61da5106b50006", /*payload_checksum_ref=*/"92b282c83efd20e7eeef52ba40842cf7", /*expected_packets=*/50, /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput); @@ -789,7 +789,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) { defined(WEBRTC_ARCH_X86_64) TEST_F(AcmSenderBitExactnessOldApi, Ilbc_30ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240)); - Run(/*audio_checksum_ref=*/"b14dba0de36efa5ec88a32c0b320b70f", + Run(/*audio_checksum_ref=*/"a739434bec8a754e9356ce2115603ce5", /*payload_checksum_ref=*/"cfae2e9f6aba96e145f2bcdd5050ce78", /*expected_packets=*/33, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -799,7 +799,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Ilbc_30ms) { #if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); - Run(/*audio_checksum_ref=*/"f5264affff25cf2cbd2e1e8a5217f9a3", + Run(/*audio_checksum_ref=*/"b875d9a3e41f5470857bdff02e3b368f", /*payload_checksum_ref=*/"fc68a87e1380614e658087cb35d5ca10", /*expected_packets=*/50, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -809,7 +809,7 @@ TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) { #if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) TEST_F(AcmSenderBitExactnessOldApi, G722_stereo_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160)); - Run(/*audio_checksum_ref=*/"be0b8528ff9db3a2219f55ddd36faf7f", + Run(/*audio_checksum_ref=*/"02c427d73363b2f37853a0dd17fe1aba", /*payload_checksum_ref=*/"66516152eeaa1e650ad94ff85f668dac", /*expected_packets=*/50, /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput); @@ -897,8 +897,8 @@ TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) { ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder( AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120)); const std::string audio_maybe_sse = - "1010e60ad34cee73c939edaf563d0593" - "|c05b4523d4c3fad2bab96d2a56baa2d0"; + "cb644fc17d9666a0f5986eef24818159" + "|4a74024473c7c729543c2790829b1e42"; const std::string payload_maybe_sse = "ea48d94e43217793af9b7e15ece94e54" diff --git a/third_party/libwebrtc/modules/audio_coding/audio_coding_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_coding_gn/moz.build index 4dad1217d0..88fa77a0e2 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_coding_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/audio_coding_gn/moz.build @@ -203,7 +203,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -213,10 +212,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/audio_coding_module_typedefs_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_coding_module_typedefs_gn/moz.build index 704026c845..851dd7b58e 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_coding_module_typedefs_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/audio_coding_module_typedefs_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/audio_coding_opus_common_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_coding_opus_common_gn/moz.build index bbb1557baa..e509916cfd 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_coding_opus_common_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/audio_coding_opus_common_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/audio_encoder_cng_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_encoder_cng_gn/moz.build index 75153f3221..7829419065 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_encoder_cng_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/audio_encoder_cng_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc index 42dd8a8786..793c73a380 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc +++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc @@ -24,18 +24,16 @@ #include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h" #include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h" #include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/logging.h" #include "rtc_base/time_utils.h" #if WEBRTC_ENABLE_PROTOBUF -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h" #else #include "modules/audio_coding/audio_network_adaptor/config.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() + #endif namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc index 3e6ecf6def..f399511757 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc @@ -17,17 +17,14 @@ #include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h" #include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h" #include "rtc_base/fake_clock.h" -#include "rtc_base/ignore_wundef.h" #include "test/gtest.h" #if WEBRTC_ENABLE_PROTOBUF -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h" #else #include "modules/audio_coding/audio_network_adaptor/config.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() #endif namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc index 2616706ee5..5ffbee219c 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc +++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc @@ -14,18 +14,15 @@ #include "absl/types/optional.h" #include "rtc_base/checks.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/system/file_wrapper.h" #if WEBRTC_ENABLE_PROTOBUF -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" #else #include "modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() #endif namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h index 8fdf2f7728..fd3a64dbb1 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h +++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h @@ -15,16 +15,14 @@ #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/system/file_wrapper.h" + #if WEBRTC_ENABLE_PROTOBUF -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h" #else #include "modules/audio_coding/audio_network_adaptor/config.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() #endif namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_config_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_config_gn/moz.build index b9d3c55453..de87e8b033 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_config_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_config_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_gn/moz.build index 7d446965f1..8a371a9aaf 100644 --- a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_gn/moz.build @@ -209,7 +209,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -219,10 +218,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/default_neteq_factory_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/default_neteq_factory_gn/moz.build index aea0a80ed4..d7928549d7 100644 --- a/third_party/libwebrtc/modules/audio_coding/default_neteq_factory_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/default_neteq_factory_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/g711_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g711_c_gn/moz.build index 575478702e..bedb8fc477 100644 --- a/third_party/libwebrtc/modules/audio_coding/g711_c_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/g711_c_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/g711_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g711_gn/moz.build index fa25fde0bd..103d89c6d8 100644 --- a/third_party/libwebrtc/modules/audio_coding/g711_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/g711_gn/moz.build @@ -196,7 +196,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -206,10 +205,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/g722_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g722_c_gn/moz.build index 4821c2bd82..48137ada85 100644 --- a/third_party/libwebrtc/modules/audio_coding/g722_c_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/g722_c_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/g722_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g722_gn/moz.build index 0a56f32af0..81eb870466 100644 --- a/third_party/libwebrtc/modules/audio_coding/g722_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/g722_gn/moz.build @@ -196,7 +196,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -206,10 +205,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/ilbc_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/ilbc_c_gn/moz.build index 43d69c7662..d3aa4e0018 100644 --- a/third_party/libwebrtc/modules/audio_coding/ilbc_c_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/ilbc_c_gn/moz.build @@ -267,7 +267,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -277,10 +276,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/ilbc_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/ilbc_gn/moz.build index c4b3b4cd13..9a397a1fdc 100644 --- a/third_party/libwebrtc/modules/audio_coding/ilbc_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/ilbc_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/isac_bwinfo_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/isac_bwinfo_gn/moz.build index 4f4a5c0e7e..fdfc4fc855 100644 --- a/third_party/libwebrtc/modules/audio_coding/isac_bwinfo_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/isac_bwinfo_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/isac_vad_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/isac_vad_gn/moz.build index a5cc52279a..1b599c5e51 100644 --- a/third_party/libwebrtc/modules/audio_coding/isac_vad_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/isac_vad_gn/moz.build @@ -187,7 +187,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -197,10 +196,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/legacy_encoded_audio_frame_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/legacy_encoded_audio_frame_gn/moz.build index 78b7338ddd..b884cb8d99 100644 --- a/third_party/libwebrtc/modules/audio_coding/legacy_encoded_audio_frame_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/legacy_encoded_audio_frame_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc index fd4f2f5a20..6648fd8709 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc @@ -95,10 +95,14 @@ DecisionLogic::DecisionLogic(NetEqController::Config config) DecisionLogic::DecisionLogic( NetEqController::Config config, std::unique_ptr<DelayManager> delay_manager, - std::unique_ptr<BufferLevelFilter> buffer_level_filter) + std::unique_ptr<BufferLevelFilter> buffer_level_filter, + std::unique_ptr<PacketArrivalHistory> packet_arrival_history) : delay_manager_(std::move(delay_manager)), buffer_level_filter_(std::move(buffer_level_filter)), - packet_arrival_history_(config_.packet_history_size_ms), + packet_arrival_history_(packet_arrival_history + ? std::move(packet_arrival_history) + : std::make_unique<PacketArrivalHistory>( + config_.packet_history_size_ms)), tick_timer_(config.tick_timer), disallow_time_stretching_(!config.allow_time_stretching), timescale_countdown_( @@ -115,7 +119,7 @@ void DecisionLogic::SoftReset() { time_stretched_cn_samples_ = 0; delay_manager_->Reset(); buffer_level_filter_->Reset(); - packet_arrival_history_.Reset(); + packet_arrival_history_->Reset(); } void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) { @@ -124,7 +128,7 @@ void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) { fs_hz == 48000); sample_rate_khz_ = fs_hz / 1000; output_size_samples_ = output_size_samples; - packet_arrival_history_.set_sample_rate(fs_hz); + packet_arrival_history_->set_sample_rate(fs_hz); } NetEq::Operation DecisionLogic::GetDecision(const NetEqStatus& status, @@ -218,15 +222,15 @@ absl::optional<int> DecisionLogic::PacketArrived( delay_manager_->SetPacketAudioLength(packet_length_samples_ * 1000 / fs_hz); } int64_t time_now_ms = tick_timer_->ticks() * tick_timer_->ms_per_tick(); - packet_arrival_history_.Insert(info.main_timestamp, time_now_ms); - if (packet_arrival_history_.size() < 2) { + packet_arrival_history_->Insert(info.main_timestamp, time_now_ms); + if (packet_arrival_history_->size() < 2) { // No meaningful delay estimate unless at least 2 packets have arrived. return absl::nullopt; } int arrival_delay_ms = - packet_arrival_history_.GetDelayMs(info.main_timestamp, time_now_ms); + packet_arrival_history_->GetDelayMs(info.main_timestamp, time_now_ms); bool reordered = - !packet_arrival_history_.IsNewestRtpTimestamp(info.main_timestamp); + !packet_arrival_history_->IsNewestRtpTimestamp(info.main_timestamp); delay_manager_->Update(arrival_delay_ms, reordered); return arrival_delay_ms; } @@ -306,10 +310,10 @@ NetEq::Operation DecisionLogic::ExpectedPacketAvailable( !status.play_dtmf) { if (config_.enable_stable_delay_mode) { const int playout_delay_ms = GetPlayoutDelayMs(status); - const int low_limit = TargetLevelMs(); - const int high_limit = low_limit + - packet_arrival_history_.GetMaxDelayMs() + - kDelayAdjustmentGranularityMs; + const int64_t low_limit = TargetLevelMs(); + const int64_t high_limit = low_limit + + packet_arrival_history_->GetMaxDelayMs() + + kDelayAdjustmentGranularityMs; if (playout_delay_ms >= high_limit * 4) { return NetEq::Operation::kFastAccelerate; } @@ -460,7 +464,7 @@ int DecisionLogic::GetPlayoutDelayMs( NetEqController::NetEqStatus status) const { uint32_t playout_timestamp = status.target_timestamp - status.sync_buffer_samples; - return packet_arrival_history_.GetDelayMs( + return packet_arrival_history_->GetDelayMs( playout_timestamp, tick_timer_->ticks() * tick_timer_->ms_per_tick()); } diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h index d96fbecd6a..a6b02c69cd 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h +++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h @@ -27,9 +27,11 @@ namespace webrtc { class DecisionLogic : public NetEqController { public: DecisionLogic(NetEqController::Config config); - DecisionLogic(NetEqController::Config config, - std::unique_ptr<DelayManager> delay_manager, - std::unique_ptr<BufferLevelFilter> buffer_level_filter); + DecisionLogic( + NetEqController::Config config, + std::unique_ptr<DelayManager> delay_manager, + std::unique_ptr<BufferLevelFilter> buffer_level_filter, + std::unique_ptr<PacketArrivalHistory> packet_arrival_history = nullptr); ~DecisionLogic() override; @@ -154,16 +156,16 @@ class DecisionLogic : public NetEqController { struct Config { Config(); - bool enable_stable_delay_mode = false; - bool combine_concealment_decision = false; + bool enable_stable_delay_mode = true; + bool combine_concealment_decision = true; int deceleration_target_level_offset_ms = 85; int packet_history_size_ms = 2000; - absl::optional<int> cng_timeout_ms; + absl::optional<int> cng_timeout_ms = 1000; }; Config config_; std::unique_ptr<DelayManager> delay_manager_; std::unique_ptr<BufferLevelFilter> buffer_level_filter_; - PacketArrivalHistory packet_arrival_history_; + std::unique_ptr<PacketArrivalHistory> packet_arrival_history_; const TickTimer* tick_timer_; int sample_rate_khz_; size_t output_size_samples_; diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc index 97e20dd883..9e9902af50 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc @@ -18,6 +18,7 @@ #include "modules/audio_coding/neteq/delay_manager.h" #include "modules/audio_coding/neteq/mock/mock_buffer_level_filter.h" #include "modules/audio_coding/neteq/mock/mock_delay_manager.h" +#include "modules/audio_coding/neteq/mock/mock_packet_arrival_history.h" #include "test/field_trial.h" #include "test/gtest.h" @@ -47,6 +48,7 @@ NetEqController::NetEqStatus CreateNetEqStatus(NetEq::Mode last_mode, return status; } +using ::testing::_; using ::testing::Return; } // namespace @@ -54,8 +56,6 @@ using ::testing::Return; class DecisionLogicTest : public ::testing::Test { protected: DecisionLogicTest() { - test::ScopedFieldTrials trials( - "WebRTC-Audio-NetEqDecisionLogicConfig/cng_timeout_ms:1000/"); NetEqController::Config config; config.tick_timer = &tick_timer_; config.allow_time_stretching = true; @@ -64,8 +64,11 @@ class DecisionLogicTest : public ::testing::Test { mock_delay_manager_ = delay_manager.get(); auto buffer_level_filter = std::make_unique<MockBufferLevelFilter>(); mock_buffer_level_filter_ = buffer_level_filter.get(); + auto packet_arrival_history = std::make_unique<MockPacketArrivalHistory>(); + mock_packet_arrival_history_ = packet_arrival_history.get(); decision_logic_ = std::make_unique<DecisionLogic>( - config, std::move(delay_manager), std::move(buffer_level_filter)); + config, std::move(delay_manager), std::move(buffer_level_filter), + std::move(packet_arrival_history)); decision_logic_->SetSampleRate(kSampleRate, kOutputSizeSamples); } @@ -73,13 +76,16 @@ class DecisionLogicTest : public ::testing::Test { std::unique_ptr<DecisionLogic> decision_logic_; MockDelayManager* mock_delay_manager_; MockBufferLevelFilter* mock_buffer_level_filter_; + MockPacketArrivalHistory* mock_packet_arrival_history_; }; TEST_F(DecisionLogicTest, NormalOperation) { EXPECT_CALL(*mock_delay_manager_, TargetDelayMs()) .WillRepeatedly(Return(100)); - EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level()) - .WillRepeatedly(Return(90 * kSamplesPerMs)); + EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _)) + .WillRepeatedly(Return(100)); + EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs()) + .WillRepeatedly(Return(0)); bool reset_decoder = false; tick_timer_.Increment(kMinTimescaleInterval + 1); @@ -92,8 +98,10 @@ TEST_F(DecisionLogicTest, NormalOperation) { TEST_F(DecisionLogicTest, Accelerate) { EXPECT_CALL(*mock_delay_manager_, TargetDelayMs()) .WillRepeatedly(Return(100)); - EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level()) - .WillRepeatedly(Return(110 * kSamplesPerMs)); + EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _)) + .WillRepeatedly(Return(150)); + EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs()) + .WillRepeatedly(Return(0)); bool reset_decoder = false; tick_timer_.Increment(kMinTimescaleInterval + 1); @@ -106,8 +114,10 @@ TEST_F(DecisionLogicTest, Accelerate) { TEST_F(DecisionLogicTest, FastAccelerate) { EXPECT_CALL(*mock_delay_manager_, TargetDelayMs()) .WillRepeatedly(Return(100)); - EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level()) - .WillRepeatedly(Return(400 * kSamplesPerMs)); + EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _)) + .WillRepeatedly(Return(500)); + EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs()) + .WillRepeatedly(Return(0)); bool reset_decoder = false; tick_timer_.Increment(kMinTimescaleInterval + 1); @@ -120,8 +130,10 @@ TEST_F(DecisionLogicTest, FastAccelerate) { TEST_F(DecisionLogicTest, PreemptiveExpand) { EXPECT_CALL(*mock_delay_manager_, TargetDelayMs()) .WillRepeatedly(Return(100)); - EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level()) - .WillRepeatedly(Return(50 * kSamplesPerMs)); + EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _)) + .WillRepeatedly(Return(50)); + EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs()) + .WillRepeatedly(Return(0)); bool reset_decoder = false; tick_timer_.Increment(kMinTimescaleInterval + 1); @@ -131,20 +143,6 @@ TEST_F(DecisionLogicTest, PreemptiveExpand) { EXPECT_FALSE(reset_decoder); } -TEST_F(DecisionLogicTest, DecelerationTargetLevelOffset) { - EXPECT_CALL(*mock_delay_manager_, TargetDelayMs()) - .WillRepeatedly(Return(500)); - EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level()) - .WillRepeatedly(Return(400 * kSamplesPerMs)); - - bool reset_decoder = false; - tick_timer_.Increment(kMinTimescaleInterval + 1); - EXPECT_EQ(decision_logic_->GetDecision( - CreateNetEqStatus(NetEq::Mode::kNormal, 400), &reset_decoder), - NetEq::Operation::kPreemptiveExpand); - EXPECT_FALSE(reset_decoder); -} - TEST_F(DecisionLogicTest, PostponeDecodeAfterExpand) { EXPECT_CALL(*mock_delay_manager_, TargetDelayMs()) .WillRepeatedly(Return(500)); @@ -170,7 +168,7 @@ TEST_F(DecisionLogicTest, TimeStrechComfortNoise) { { bool reset_decoder = false; // Below target window. - auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 400); + auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 200); status.generated_noise_samples = 400 * kSamplesPerMs; status.next_packet->timestamp = status.target_timestamp + 400 * kSamplesPerMs; @@ -189,18 +187,6 @@ TEST_F(DecisionLogicTest, TimeStrechComfortNoise) { EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder), NetEq::Operation::kNormal); EXPECT_FALSE(reset_decoder); - - // The buffer level filter should be adjusted with the number of samples - // that was skipped. - int timestamp_leap = status.next_packet->timestamp - - status.target_timestamp - - status.generated_noise_samples; - EXPECT_CALL(*mock_buffer_level_filter_, - Update(400 * kSamplesPerMs, timestamp_leap)); - EXPECT_EQ(decision_logic_->GetDecision( - CreateNetEqStatus(NetEq::Mode::kNormal, 400), &reset_decoder), - NetEq::Operation::kNormal); - EXPECT_FALSE(reset_decoder); } } diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h new file mode 100644 index 0000000000..1b2080cd94 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h @@ -0,0 +1,32 @@ +/* + * Copyright 2023 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ +#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ + +#include "modules/audio_coding/neteq/packet_arrival_history.h" +#include "test/gmock.h" + +namespace webrtc { + +class MockPacketArrivalHistory : public PacketArrivalHistory { + public: + MockPacketArrivalHistory() : PacketArrivalHistory(0) {} + + MOCK_METHOD(int, + GetDelayMs, + (uint32_t rtp_timestamp, int64_t time_ms), + (const override)); + MOCK_METHOD(int, GetMaxDelayMs, (), (const override)); +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h index 48357ea466..fa44f606fc 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h +++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h @@ -18,39 +18,15 @@ namespace webrtc { class MockPacketBuffer : public PacketBuffer { public: - MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer) - : PacketBuffer(max_number_of_packets, tick_timer) {} + MockPacketBuffer(size_t max_number_of_packets, + const TickTimer* tick_timer, + StatisticsCalculator* stats) + : PacketBuffer(max_number_of_packets, tick_timer, stats) {} ~MockPacketBuffer() override { Die(); } MOCK_METHOD(void, Die, ()); - MOCK_METHOD(void, Flush, (StatisticsCalculator * stats), (override)); - MOCK_METHOD(void, - PartialFlush, - (int target_level_ms, - size_t sample_rate, - size_t last_decoded_length, - StatisticsCalculator* stats), - (override)); + MOCK_METHOD(void, Flush, (), (override)); MOCK_METHOD(bool, Empty, (), (const, override)); - MOCK_METHOD(int, - InsertPacket, - (Packet && packet, - StatisticsCalculator* stats, - size_t last_decoded_length, - size_t sample_rate, - int target_level_ms, - const DecoderDatabase& decoder_database), - (override)); - MOCK_METHOD(int, - InsertPacketList, - (PacketList * packet_list, - const DecoderDatabase& decoder_database, - absl::optional<uint8_t>* current_rtp_payload_type, - absl::optional<uint8_t>* current_cng_rtp_payload_type, - StatisticsCalculator* stats, - size_t last_decoded_length, - size_t sample_rate, - int target_level_ms), - (override)); + MOCK_METHOD(int, InsertPacket, (Packet && packet), (override)); MOCK_METHOD(int, NextTimestamp, (uint32_t * next_timestamp), @@ -61,19 +37,14 @@ class MockPacketBuffer : public PacketBuffer { (const, override)); MOCK_METHOD(const Packet*, PeekNextPacket, (), (const, override)); MOCK_METHOD(absl::optional<Packet>, GetNextPacket, (), (override)); - MOCK_METHOD(int, - DiscardNextPacket, - (StatisticsCalculator * stats), - (override)); + MOCK_METHOD(int, DiscardNextPacket, (), (override)); MOCK_METHOD(void, DiscardOldPackets, - (uint32_t timestamp_limit, - uint32_t horizon_samples, - StatisticsCalculator* stats), + (uint32_t timestamp_limit, uint32_t horizon_samples), (override)); MOCK_METHOD(void, DiscardAllOldPackets, - (uint32_t timestamp_limit, StatisticsCalculator* stats), + (uint32_t timestamp_limit), (override)); MOCK_METHOD(size_t, NumPacketsInBuffer, (), (const, override)); }; diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc index 52e8cbad3a..e5c8bf6c08 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc @@ -70,6 +70,62 @@ std::unique_ptr<NetEqController> CreateNetEqController( return controller_factory.CreateNetEqController(config); } +void SetAudioFrameActivityAndType(bool vad_enabled, + NetEqImpl::OutputType type, + AudioFrame::VADActivity last_vad_activity, + AudioFrame* audio_frame) { + switch (type) { + case NetEqImpl::OutputType::kNormalSpeech: { + audio_frame->speech_type_ = AudioFrame::kNormalSpeech; + audio_frame->vad_activity_ = AudioFrame::kVadActive; + break; + } + case NetEqImpl::OutputType::kVadPassive: { + // This should only be reached if the VAD is enabled. + RTC_DCHECK(vad_enabled); + audio_frame->speech_type_ = AudioFrame::kNormalSpeech; + audio_frame->vad_activity_ = AudioFrame::kVadPassive; + break; + } + case NetEqImpl::OutputType::kCNG: { + audio_frame->speech_type_ = AudioFrame::kCNG; + audio_frame->vad_activity_ = AudioFrame::kVadPassive; + break; + } + case NetEqImpl::OutputType::kPLC: { + audio_frame->speech_type_ = AudioFrame::kPLC; + audio_frame->vad_activity_ = last_vad_activity; + break; + } + case NetEqImpl::OutputType::kPLCCNG: { + audio_frame->speech_type_ = AudioFrame::kPLCCNG; + audio_frame->vad_activity_ = AudioFrame::kVadPassive; + break; + } + case NetEqImpl::OutputType::kCodecPLC: { + audio_frame->speech_type_ = AudioFrame::kCodecPLC; + audio_frame->vad_activity_ = last_vad_activity; + break; + } + default: + RTC_DCHECK_NOTREACHED(); + } + if (!vad_enabled) { + // Always set kVadUnknown when receive VAD is inactive. + audio_frame->vad_activity_ = AudioFrame::kVadUnknown; + } +} + +// Returns true if both payload types are known to the decoder database, and +// have the same sample rate. +bool EqualSampleRates(uint8_t pt1, + uint8_t pt2, + const DecoderDatabase& decoder_database) { + auto* di1 = decoder_database.GetDecoderInfo(pt1); + auto* di2 = decoder_database.GetDecoderInfo(pt2); + return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz(); +} + } // namespace NetEqImpl::Dependencies::Dependencies( @@ -84,8 +140,9 @@ NetEqImpl::Dependencies::Dependencies( new DecoderDatabase(decoder_factory, config.codec_pair_id)), dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)), dtmf_tone_generator(new DtmfToneGenerator), - packet_buffer( - new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())), + packet_buffer(new PacketBuffer(config.max_packets_in_buffer, + tick_timer.get(), + stats.get())), neteq_controller( CreateNetEqController(controller_factory, config.min_delay_ms, @@ -182,54 +239,6 @@ void NetEqImpl::InsertEmptyPacket(const RTPHeader& rtp_header) { controller_->RegisterEmptyPacket(); } -namespace { -void SetAudioFrameActivityAndType(bool vad_enabled, - NetEqImpl::OutputType type, - AudioFrame::VADActivity last_vad_activity, - AudioFrame* audio_frame) { - switch (type) { - case NetEqImpl::OutputType::kNormalSpeech: { - audio_frame->speech_type_ = AudioFrame::kNormalSpeech; - audio_frame->vad_activity_ = AudioFrame::kVadActive; - break; - } - case NetEqImpl::OutputType::kVadPassive: { - // This should only be reached if the VAD is enabled. - RTC_DCHECK(vad_enabled); - audio_frame->speech_type_ = AudioFrame::kNormalSpeech; - audio_frame->vad_activity_ = AudioFrame::kVadPassive; - break; - } - case NetEqImpl::OutputType::kCNG: { - audio_frame->speech_type_ = AudioFrame::kCNG; - audio_frame->vad_activity_ = AudioFrame::kVadPassive; - break; - } - case NetEqImpl::OutputType::kPLC: { - audio_frame->speech_type_ = AudioFrame::kPLC; - audio_frame->vad_activity_ = last_vad_activity; - break; - } - case NetEqImpl::OutputType::kPLCCNG: { - audio_frame->speech_type_ = AudioFrame::kPLCCNG; - audio_frame->vad_activity_ = AudioFrame::kVadPassive; - break; - } - case NetEqImpl::OutputType::kCodecPLC: { - audio_frame->speech_type_ = AudioFrame::kCodecPLC; - audio_frame->vad_activity_ = last_vad_activity; - break; - } - default: - RTC_DCHECK_NOTREACHED(); - } - if (!vad_enabled) { - // Always set kVadUnknown when receive VAD is inactive. - audio_frame->vad_activity_ = AudioFrame::kVadUnknown; - } -} -} // namespace - int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted, int* current_sample_rate_hz, @@ -265,7 +274,7 @@ void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) { const std::vector<int> changed_payload_types = decoder_database_->SetCodecs(codecs); for (const int pt : changed_payload_types) { - packet_buffer_->DiscardPacketsWithPayloadType(pt, stats_.get()); + packet_buffer_->DiscardPacketsWithPayloadType(pt); } } @@ -283,8 +292,7 @@ int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) { MutexLock lock(&mutex_); int ret = decoder_database_->Remove(rtp_payload_type); if (ret == DecoderDatabase::kOK || ret == DecoderDatabase::kDecoderNotFound) { - packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type, - stats_.get()); + packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type); return kOK; } return kFail; @@ -441,7 +449,7 @@ absl::optional<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat( void NetEqImpl::FlushBuffers() { MutexLock lock(&mutex_); RTC_LOG(LS_VERBOSE) << "FlushBuffers"; - packet_buffer_->Flush(stats_.get()); + packet_buffer_->Flush(); RTC_DCHECK(sync_buffer_.get()); RTC_DCHECK(expand_.get()); sync_buffer_->Flush(); @@ -542,7 +550,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header, // the packet has been successfully inserted into the packet buffer. // Flush the packet buffer and DTMF buffer. - packet_buffer_->Flush(stats_.get()); + packet_buffer_->Flush(); dtmf_buffer_->Flush(); // Update audio buffer timestamp. @@ -681,26 +689,25 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header, number_of_primary_packets); } - // Insert packets in buffer. - const int target_level_ms = controller_->TargetLevelMs(); - const int ret = packet_buffer_->InsertPacketList( - &parsed_packet_list, *decoder_database_, ¤t_rtp_payload_type_, - ¤t_cng_rtp_payload_type_, stats_.get(), decoder_frame_length_, - last_output_sample_rate_hz_, target_level_ms); bool buffer_flush_occured = false; - if (ret == PacketBuffer::kFlushed) { + for (Packet& packet : parsed_packet_list) { + if (MaybeChangePayloadType(packet.payload_type)) { + packet_buffer_->Flush(); + buffer_flush_occured = true; + } + int return_val = packet_buffer_->InsertPacket(std::move(packet)); + if (return_val == PacketBuffer::kFlushed) { + buffer_flush_occured = true; + } else if (return_val != PacketBuffer::kOK) { + // An error occurred. + return kOtherError; + } + } + + if (buffer_flush_occured) { // Reset DSP timestamp etc. if packet buffer flushed. new_codec_ = true; update_sample_rate_and_channels = true; - buffer_flush_occured = true; - } else if (ret == PacketBuffer::kPartialFlush) { - // Forward sync buffer timestamp - timestamp_ = packet_buffer_->PeekNextPacket()->timestamp; - sync_buffer_->IncreaseEndTimestamp(timestamp_ - - sync_buffer_->end_timestamp()); - buffer_flush_occured = true; - } else if (ret != PacketBuffer::kOK) { - return kOtherError; } if (first_packet_) { @@ -767,6 +774,31 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header, return 0; } +bool NetEqImpl::MaybeChangePayloadType(uint8_t payload_type) { + bool changed = false; + if (decoder_database_->IsComfortNoise(payload_type)) { + if (current_cng_rtp_payload_type_ && + *current_cng_rtp_payload_type_ != payload_type) { + // New CNG payload type implies new codec type. + current_rtp_payload_type_ = absl::nullopt; + changed = true; + } + current_cng_rtp_payload_type_ = payload_type; + } else if (!decoder_database_->IsDtmf(payload_type)) { + // This must be speech. + if ((current_rtp_payload_type_ && + *current_rtp_payload_type_ != payload_type) || + (current_cng_rtp_payload_type_ && + !EqualSampleRates(payload_type, *current_cng_rtp_payload_type_, + *decoder_database_))) { + current_cng_rtp_payload_type_ = absl::nullopt; + changed = true; + } + current_rtp_payload_type_ = payload_type; + } + return changed; +} + int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted, absl::optional<Operation> action_override) { @@ -1037,8 +1069,7 @@ int NetEqImpl::GetDecision(Operation* operation, uint32_t end_timestamp = sync_buffer_->end_timestamp(); if (!new_codec_) { const uint32_t five_seconds_samples = 5 * fs_hz_; - packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples, - stats_.get()); + packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples); } const Packet* packet = packet_buffer_->PeekNextPacket(); @@ -1058,14 +1089,12 @@ int NetEqImpl::GetDecision(Operation* operation, (end_timestamp >= packet->timestamp || end_timestamp + generated_noise_samples > packet->timestamp)) { // Don't use this packet, discard it. - if (packet_buffer_->DiscardNextPacket(stats_.get()) != - PacketBuffer::kOK) { + if (packet_buffer_->DiscardNextPacket() != PacketBuffer::kOK) { RTC_DCHECK_NOTREACHED(); // Must be ok by design. } // Check buffer again. if (!new_codec_) { - packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_, - stats_.get()); + packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_); } packet = packet_buffer_->PeekNextPacket(); } @@ -2024,7 +2053,7 @@ int NetEqImpl::ExtractPackets(size_t required_samples, // we could end up in the situation where we never decode anything, since // all incoming packets are considered too old but the buffer will also // never be flooded and flushed. - packet_buffer_->DiscardAllOldPackets(timestamp_, stats_.get()); + packet_buffer_->DiscardAllOldPackets(timestamp_); } return rtc::dchecked_cast<int>(extracted_samples); diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h index f27738bcbf..f8f2b06410 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h +++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h @@ -27,6 +27,7 @@ #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/expand_uma_logger.h" #include "modules/audio_coding/neteq/packet.h" +#include "modules/audio_coding/neteq/packet_buffer.h" #include "modules/audio_coding/neteq/random_vector.h" #include "modules/audio_coding/neteq/statistics_calculator.h" #include "rtc_base/synchronization/mutex.h" @@ -46,7 +47,6 @@ class Expand; class Merge; class NackTracker; class Normal; -class PacketBuffer; class RedPayloadSplitter; class PostDecodeVad; class PreemptiveExpand; @@ -215,6 +215,12 @@ class NetEqImpl : public webrtc::NetEq { rtc::ArrayView<const uint8_t> payload) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); + // Returns true if the payload type changed (this should be followed by + // resetting various state). Returns false if the current payload type is + // unknown or equal to `payload_type`. + bool MaybeChangePayloadType(uint8_t payload_type) + RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); + // Delivers 10 ms of audio data. The data is written to `audio_frame`. // Returns 0 on success, otherwise an error code. int GetAudioInternal(AudioFrame* audio_frame, diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc index e61cd52502..8309dafb58 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc @@ -108,8 +108,8 @@ class NetEqImplTest : public ::testing::Test { dtmf_tone_generator_ = deps.dtmf_tone_generator.get(); if (use_mock_packet_buffer_) { - std::unique_ptr<MockPacketBuffer> mock( - new MockPacketBuffer(config_.max_packets_in_buffer, tick_timer_)); + std::unique_ptr<MockPacketBuffer> mock(new MockPacketBuffer( + config_.max_packets_in_buffer, tick_timer_, deps.stats.get())); mock_packet_buffer_ = mock.get(); deps.packet_buffer = std::move(mock); } @@ -120,7 +120,6 @@ class NetEqImplTest : public ::testing::Test { mock_neteq_controller_ = mock.get(); deps.neteq_controller = std::move(mock); } else { - deps.stats = std::make_unique<StatisticsCalculator>(); NetEqController::Config controller_config; controller_config.tick_timer = tick_timer_; controller_config.base_min_delay_ms = config_.min_delay_ms; @@ -329,15 +328,10 @@ TEST_F(NetEqImplTest, InsertPacket) { // Expectations for packet buffer. EXPECT_CALL(*mock_packet_buffer_, Empty()) .WillOnce(Return(false)); // Called once after first packet is inserted. - EXPECT_CALL(*mock_packet_buffer_, Flush(_)).Times(1); - EXPECT_CALL(*mock_packet_buffer_, InsertPacketList(_, _, _, _, _, _, _, _)) + EXPECT_CALL(*mock_packet_buffer_, Flush()).Times(1); + EXPECT_CALL(*mock_packet_buffer_, InsertPacket(_)) .Times(2) - .WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType), - WithArg<0>(Invoke(DeletePacketsAndReturnOk)))); - // SetArgPointee<2>(kPayloadType) means that the third argument (zero-based - // index) is a pointer, and the variable pointed to is set to kPayloadType. - // Also invoke the function DeletePacketsAndReturnOk to properly delete all - // packets in the list (to avoid memory leaks in the test). + .WillRepeatedly(Return(PacketBuffer::kOK)); EXPECT_CALL(*mock_packet_buffer_, PeekNextPacket()) .Times(1) .WillOnce(Return(&fake_packet)); @@ -1246,12 +1240,15 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) { EXPECT_EQ(kChannels, output.num_channels_); EXPECT_THAT(output.packet_infos_, IsEmpty()); - // Second call to GetAudio will decode the packet that is ok. No errors are - // expected. - EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted)); - EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels); - EXPECT_EQ(kChannels, output.num_channels_); - EXPECT_THAT(output.packet_infos_, SizeIs(1)); + // Call GetAudio until the next packet is decoded. + int calls = 0; + int kTimeout = 10; + while (output.packet_infos_.empty() && calls < kTimeout) { + EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted)); + EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels); + EXPECT_EQ(kChannels, output.num_channels_); + } + EXPECT_LT(calls, kTimeout); // Die isn't called through NiceMock (since it's called by the // MockAudioDecoder constructor), so it needs to be mocked explicitly. @@ -1640,6 +1637,74 @@ TEST_F(NetEqImplTest, NoCrashWith1000Channels) { } } +// The test first inserts a packet with narrow-band CNG, then a packet with +// wide-band speech. The expected behavior is to detect a change in sample rate, +// even though no speech packet has been inserted before, and flush out the CNG +// packet. +TEST_F(NetEqImplTest, CngFirstThenSpeechWithNewSampleRate) { + UseNoMocks(); + CreateInstance(); + constexpr int kCnPayloadType = 7; + neteq_->RegisterPayloadType(kCnPayloadType, SdpAudioFormat("cn", 8000, 1)); + constexpr int kSpeechPayloadType = 8; + neteq_->RegisterPayloadType(kSpeechPayloadType, + SdpAudioFormat("l16", 16000, 1)); + + RTPHeader header; + header.payloadType = kCnPayloadType; + uint8_t payload[320] = {0}; + + EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK); + EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 0u); + + header.payloadType = kSpeechPayloadType; + header.timestamp += 160; + EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK); + // CN packet should be discarded, since it does not match the + // new speech sample rate. + EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 1u); + + // Next decoded packet should be speech. + AudioFrame audio_frame; + bool muted; + EXPECT_EQ(neteq_->GetAudio(&audio_frame, &muted), NetEq::kOK); + EXPECT_EQ(audio_frame.sample_rate_hz(), 16000); + EXPECT_EQ(audio_frame.speech_type_, AudioFrame::SpeechType::kNormalSpeech); +} + +TEST_F(NetEqImplTest, InsertPacketChangePayloadType) { + UseNoMocks(); + CreateInstance(); + constexpr int kPcmuPayloadType = 7; + neteq_->RegisterPayloadType(kPcmuPayloadType, + SdpAudioFormat("pcmu", 8000, 1)); + constexpr int kPcmaPayloadType = 8; + neteq_->RegisterPayloadType(kPcmaPayloadType, + SdpAudioFormat("pcma", 8000, 1)); + + RTPHeader header; + header.payloadType = kPcmuPayloadType; + header.timestamp = 1234; + uint8_t payload[160] = {0}; + + EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK); + EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 0u); + + header.payloadType = kPcmaPayloadType; + header.timestamp += 80; + EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK); + // The previous packet should be discarded since the codec changed. + EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 1u); + + // Next decoded packet should be speech. + AudioFrame audio_frame; + bool muted; + EXPECT_EQ(neteq_->GetAudio(&audio_frame, &muted), NetEq::kOK); + EXPECT_EQ(audio_frame.sample_rate_hz(), 8000); + EXPECT_EQ(audio_frame.speech_type_, AudioFrame::SpeechType::kNormalSpeech); + // TODO(jakobi): check active decoder. +} + class Decoder120ms : public AudioDecoder { public: Decoder120ms(int sample_rate_hz, SpeechType speech_type) diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc index a669ad727e..da516982c7 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc @@ -273,15 +273,16 @@ class NetEqNetworkStatsTest { // Next we introduce packet losses. SetPacketLossRate(0.1); - expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 898; + expects.expand_rate = expects.speech_expand_rate = kLargerThan; RunTest(50, expects); // Next we enable FEC. decoder_->set_fec_enabled(true); // If FEC fills in the lost packets, no packet loss will be counted. + expects.expand_rate = expects.speech_expand_rate = kEqual; expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 0; - expects.stats_ref.secondary_decoded_rate = 2006; - expects.stats_ref.secondary_discarded_rate = 14336; + expects.secondary_decoded_rate = kLargerThan; + expects.secondary_discarded_rate = kLargerThan; RunTest(50, expects); } diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc index 77bd5b5035..aec7e580ec 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc @@ -31,7 +31,6 @@ #include "modules/include/module_common_types_public.h" #include "modules/rtp_rtcp/include/rtcp_statistics.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/message_digest.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/strings/string_builder.h" @@ -77,11 +76,11 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); const std::string output_checksum = - "fec6827bb9ee0b21770bbbb4a3a6f8823bf537dc|" - "3610cc7be4b3407b9c273b1299ab7f8f47cca96b"; + "2efdbea92c3fb2383c59f89d881efec9f94001d0|" + "a6831b946b59913852ae3e53f99fa8f209bb23cd"; const std::string network_stats_checksum = - "3d043e47e5f4bb81d37e7bce8c44bf802965c853|" + "dfaf4399fd60293405290476ccf1c05c807c71a0|" "076662525572dba753b11578330bd491923f7f5e"; DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum, @@ -99,11 +98,11 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) { webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp"); const std::string output_checksum = - "b3c4899eab5378ef5e54f2302948872149f6ad5e|" - "589e975ec31ea13f302457fea1425be9380ffb96"; + "7eddce841cbfa500964c91cdae78b01b9f448948|" + "5d13affec87bf4cc8c7667f0cd0d25e1ad09c7c3"; const std::string network_stats_checksum = - "dc8447b9fee1a21fd5d1f4045d62b982a3fb0215"; + "92b0fdcbf8bb9354d40140b7312f2fb76a078555"; DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum, absl::GetFlag(FLAGS_gen_ref)); @@ -165,7 +164,7 @@ TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) { const double kDriftFactor = 1000.0 / (1000.0 + 25.0); const double kNetworkFreezeTimeMs = 0.0; const bool kGetAudioDuringFreezeRecovery = false; - const int kDelayToleranceMs = 20; + const int kDelayToleranceMs = 60; const int kMaxTimeToSpeechMs = 100; LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs, kGetAudioDuringFreezeRecovery, kDelayToleranceMs, @@ -495,7 +494,7 @@ TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { timestamp += kCngPeriodSamples; uint32_t first_speech_timestamp = timestamp; // Insert speech again. - for (int i = 0; i < 3; ++i) { + for (int i = 0; i < 4; ++i) { PopulateRtpInfo(seq_no, timestamp, &rtp_info); ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload)); ++seq_no; @@ -700,8 +699,7 @@ TEST_F(NetEqDecodingTestWithMutedState, MutedStateOldPacket) { for (int i = 0; i < 5; ++i) { InsertPacket(kSamples * (i - 1000)); } - EXPECT_FALSE(GetAudioReturnMuted()); - EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_); + GetAudioUntilNormal(); } // Verifies that NetEq doesn't enter muted state when CNG mode is active and the diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h index cad362b469..722caf5688 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h +++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h @@ -26,6 +26,7 @@ namespace webrtc { class PacketArrivalHistory { public: explicit PacketArrivalHistory(int window_size_ms); + virtual ~PacketArrivalHistory() = default; // Insert packet with `rtp_timestamp` and `arrival_time_ms` into the history. void Insert(uint32_t rtp_timestamp, int64_t arrival_time_ms); @@ -34,10 +35,10 @@ class PacketArrivalHistory { // `(time_ms - p.arrival_time_ms) - (rtp_timestamp - p.rtp_timestamp)` // where `p` is chosen as the packet arrival in the history that maximizes the // delay. - int GetDelayMs(uint32_t rtp_timestamp, int64_t time_ms) const; + virtual int GetDelayMs(uint32_t rtp_timestamp, int64_t time_ms) const; // Get the maximum packet arrival delay observed in the history. - int GetMaxDelayMs() const; + virtual int GetMaxDelayMs() const; bool IsNewestRtpTimestamp(uint32_t rtp_timestamp) const; diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc index 9bfa908ab9..47c391a18f 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc @@ -44,53 +44,14 @@ class NewTimestampIsLarger { const Packet& new_packet_; }; -// Returns true if both payload types are known to the decoder database, and -// have the same sample rate. -bool EqualSampleRates(uint8_t pt1, - uint8_t pt2, - const DecoderDatabase& decoder_database) { - auto* di1 = decoder_database.GetDecoderInfo(pt1); - auto* di2 = decoder_database.GetDecoderInfo(pt2); - return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz(); -} - -void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) { - RTC_CHECK(stats); - if (codec_level > 0) { - stats->SecondaryPacketsDiscarded(1); - } else { - stats->PacketsDiscarded(1); - } -} - -absl::optional<SmartFlushingConfig> GetSmartflushingConfig() { - absl::optional<SmartFlushingConfig> result; - std::string field_trial_string = - field_trial::FindFullName("WebRTC-Audio-NetEqSmartFlushing"); - result = SmartFlushingConfig(); - bool enabled = false; - auto parser = StructParametersParser::Create( - "enabled", &enabled, "target_level_threshold_ms", - &result->target_level_threshold_ms, "target_level_multiplier", - &result->target_level_multiplier); - parser->Parse(field_trial_string); - if (!enabled) { - return absl::nullopt; - } - RTC_LOG(LS_INFO) << "Using smart flushing, target_level_threshold_ms: " - << result->target_level_threshold_ms - << ", target_level_multiplier: " - << result->target_level_multiplier; - return result; -} - } // namespace PacketBuffer::PacketBuffer(size_t max_number_of_packets, - const TickTimer* tick_timer) - : smart_flushing_config_(GetSmartflushingConfig()), - max_number_of_packets_(max_number_of_packets), - tick_timer_(tick_timer) {} + const TickTimer* tick_timer, + StatisticsCalculator* stats) + : max_number_of_packets_(max_number_of_packets), + tick_timer_(tick_timer), + stats_(stats) {} // Destructor. All packets in the buffer will be destroyed. PacketBuffer::~PacketBuffer() { @@ -98,45 +59,19 @@ PacketBuffer::~PacketBuffer() { } // Flush the buffer. All packets in the buffer will be destroyed. -void PacketBuffer::Flush(StatisticsCalculator* stats) { +void PacketBuffer::Flush() { for (auto& p : buffer_) { - LogPacketDiscarded(p.priority.codec_level, stats); + LogPacketDiscarded(p.priority.codec_level); } buffer_.clear(); - stats->FlushedPacketBuffer(); -} - -void PacketBuffer::PartialFlush(int target_level_ms, - size_t sample_rate, - size_t last_decoded_length, - StatisticsCalculator* stats) { - // Make sure that at least half the packet buffer capacity will be available - // after the flush. This is done to avoid getting stuck if the target level is - // very high. - int target_level_samples = - std::min(target_level_ms * sample_rate / 1000, - max_number_of_packets_ * last_decoded_length / 2); - // We should avoid flushing to very low levels. - target_level_samples = std::max( - target_level_samples, smart_flushing_config_->target_level_threshold_ms); - while (GetSpanSamples(last_decoded_length, sample_rate, false) > - static_cast<size_t>(target_level_samples) || - buffer_.size() > max_number_of_packets_ / 2) { - LogPacketDiscarded(PeekNextPacket()->priority.codec_level, stats); - buffer_.pop_front(); - } + stats_->FlushedPacketBuffer(); } bool PacketBuffer::Empty() const { return buffer_.empty(); } -int PacketBuffer::InsertPacket(Packet&& packet, - StatisticsCalculator* stats, - size_t last_decoded_length, - size_t sample_rate, - int target_level_ms, - const DecoderDatabase& decoder_database) { +int PacketBuffer::InsertPacket(Packet&& packet) { if (packet.empty()) { RTC_LOG(LS_WARNING) << "InsertPacket invalid packet"; return kInvalidPacket; @@ -149,32 +84,11 @@ int PacketBuffer::InsertPacket(Packet&& packet, packet.waiting_time = tick_timer_->GetNewStopwatch(); - // Perform a smart flush if the buffer size exceeds a multiple of the target - // level. - const size_t span_threshold = - smart_flushing_config_ - ? smart_flushing_config_->target_level_multiplier * - std::max(smart_flushing_config_->target_level_threshold_ms, - target_level_ms) * - sample_rate / 1000 - : 0; - const bool smart_flush = - smart_flushing_config_.has_value() && - GetSpanSamples(last_decoded_length, sample_rate, false) >= span_threshold; - if (buffer_.size() >= max_number_of_packets_ || smart_flush) { - size_t buffer_size_before_flush = buffer_.size(); - if (smart_flushing_config_.has_value()) { - // Flush down to the target level. - PartialFlush(target_level_ms, sample_rate, last_decoded_length, stats); - return_val = kPartialFlush; - } else { - // Buffer is full. - Flush(stats); - return_val = kFlushed; - } - RTC_LOG(LS_WARNING) << "Packet buffer flushed, " - << (buffer_size_before_flush - buffer_.size()) - << " packets discarded."; + if (buffer_.size() >= max_number_of_packets_) { + // Buffer is full. + Flush(); + return_val = kFlushed; + RTC_LOG(LS_WARNING) << "Packet buffer flushed."; } // Get an iterator pointing to the place in the buffer where the new packet @@ -187,7 +101,7 @@ int PacketBuffer::InsertPacket(Packet&& packet, // timestamp as `rit`, which has a higher priority, do not insert the new // packet to list. if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) { - LogPacketDiscarded(packet.priority.codec_level, stats); + LogPacketDiscarded(packet.priority.codec_level); return return_val; } @@ -196,7 +110,7 @@ int PacketBuffer::InsertPacket(Packet&& packet, // packet. PacketList::iterator it = rit.base(); if (it != buffer_.end() && packet.timestamp == it->timestamp) { - LogPacketDiscarded(it->priority.codec_level, stats); + LogPacketDiscarded(it->priority.codec_level); it = buffer_.erase(it); } buffer_.insert(it, std::move(packet)); // Insert the packet at that position. @@ -204,57 +118,6 @@ int PacketBuffer::InsertPacket(Packet&& packet, return return_val; } -int PacketBuffer::InsertPacketList( - PacketList* packet_list, - const DecoderDatabase& decoder_database, - absl::optional<uint8_t>* current_rtp_payload_type, - absl::optional<uint8_t>* current_cng_rtp_payload_type, - StatisticsCalculator* stats, - size_t last_decoded_length, - size_t sample_rate, - int target_level_ms) { - RTC_DCHECK(stats); - bool flushed = false; - for (auto& packet : *packet_list) { - if (decoder_database.IsComfortNoise(packet.payload_type)) { - if (*current_cng_rtp_payload_type && - **current_cng_rtp_payload_type != packet.payload_type) { - // New CNG payload type implies new codec type. - *current_rtp_payload_type = absl::nullopt; - Flush(stats); - flushed = true; - } - *current_cng_rtp_payload_type = packet.payload_type; - } else if (!decoder_database.IsDtmf(packet.payload_type)) { - // This must be speech. - if ((*current_rtp_payload_type && - **current_rtp_payload_type != packet.payload_type) || - (*current_cng_rtp_payload_type && - !EqualSampleRates(packet.payload_type, - **current_cng_rtp_payload_type, - decoder_database))) { - *current_cng_rtp_payload_type = absl::nullopt; - Flush(stats); - flushed = true; - } - *current_rtp_payload_type = packet.payload_type; - } - int return_val = - InsertPacket(std::move(packet), stats, last_decoded_length, sample_rate, - target_level_ms, decoder_database); - if (return_val == kFlushed) { - // The buffer flushed, but this is not an error. We can still continue. - flushed = true; - } else if (return_val != kOK) { - // An error occurred. Delete remaining packets in list and return. - packet_list->clear(); - return return_val; - } - } - packet_list->clear(); - return flushed ? kFlushed : kOK; -} - int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const { if (Empty()) { return kBufferEmpty; @@ -303,43 +166,40 @@ absl::optional<Packet> PacketBuffer::GetNextPacket() { return packet; } -int PacketBuffer::DiscardNextPacket(StatisticsCalculator* stats) { +int PacketBuffer::DiscardNextPacket() { if (Empty()) { return kBufferEmpty; } // Assert that the packet sanity checks in InsertPacket method works. const Packet& packet = buffer_.front(); RTC_DCHECK(!packet.empty()); - LogPacketDiscarded(packet.priority.codec_level, stats); + LogPacketDiscarded(packet.priority.codec_level); buffer_.pop_front(); return kOK; } void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit, - uint32_t horizon_samples, - StatisticsCalculator* stats) { - buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) { + uint32_t horizon_samples) { + buffer_.remove_if([this, timestamp_limit, horizon_samples](const Packet& p) { if (timestamp_limit == p.timestamp || !IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) { return false; } - LogPacketDiscarded(p.priority.codec_level, stats); + LogPacketDiscarded(p.priority.codec_level); return true; }); } -void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit, - StatisticsCalculator* stats) { - DiscardOldPackets(timestamp_limit, 0, stats); +void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) { + DiscardOldPackets(timestamp_limit, 0); } -void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type, - StatisticsCalculator* stats) { - buffer_.remove_if([payload_type, stats](const Packet& p) { +void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type) { + buffer_.remove_if([this, payload_type](const Packet& p) { if (p.payload_type != payload_type) { return false; } - LogPacketDiscarded(p.priority.codec_level, stats); + LogPacketDiscarded(p.priority.codec_level); return true; }); } @@ -404,4 +264,12 @@ bool PacketBuffer::ContainsDtxOrCngPacket( return false; } +void PacketBuffer::LogPacketDiscarded(int codec_level) { + if (codec_level > 0) { + stats_->SecondaryPacketsDiscarded(1); + } else { + stats_->PacketsDiscarded(1); + } +} + } // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h index 1eef64a02c..795dd4e812 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h +++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h @@ -21,14 +21,6 @@ namespace webrtc { class DecoderDatabase; class StatisticsCalculator; class TickTimer; -struct SmartFlushingConfig { - // When calculating the flushing threshold, the maximum between the target - // level and this value is used. - int target_level_threshold_ms = 500; - // A smart flush is triggered when the packet buffer contains a multiple of - // the target level. - int target_level_multiplier = 3; -}; // This is the actual buffer holding the packets before decoding. class PacketBuffer { @@ -36,7 +28,6 @@ class PacketBuffer { enum BufferReturnCodes { kOK = 0, kFlushed, - kPartialFlush, kNotFound, kBufferEmpty, kInvalidPacket, @@ -45,7 +36,9 @@ class PacketBuffer { // Constructor creates a buffer which can hold a maximum of // `max_number_of_packets` packets. - PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer); + PacketBuffer(size_t max_number_of_packets, + const TickTimer* tick_timer, + StatisticsCalculator* stats); // Deletes all packets in the buffer before destroying the buffer. virtual ~PacketBuffer(); @@ -54,13 +47,7 @@ class PacketBuffer { PacketBuffer& operator=(const PacketBuffer&) = delete; // Flushes the buffer and deletes all packets in it. - virtual void Flush(StatisticsCalculator* stats); - - // Partial flush. Flush packets but leave some packets behind. - virtual void PartialFlush(int target_level_ms, - size_t sample_rate, - size_t last_decoded_length, - StatisticsCalculator* stats); + virtual void Flush(); // Returns true for an empty buffer. virtual bool Empty() const; @@ -69,30 +56,7 @@ class PacketBuffer { // the packet object. // Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer // was flushed due to overfilling. - virtual int InsertPacket(Packet&& packet, - StatisticsCalculator* stats, - size_t last_decoded_length, - size_t sample_rate, - int target_level_ms, - const DecoderDatabase& decoder_database); - - // Inserts a list of packets into the buffer. The buffer will take over - // ownership of the packet objects. - // Returns PacketBuffer::kOK if all packets were inserted successfully. - // If the buffer was flushed due to overfilling, only a subset of the list is - // inserted, and PacketBuffer::kFlushed is returned. - // The last three parameters are included for legacy compatibility. - // TODO(hlundin): Redesign to not use current_*_payload_type and - // decoder_database. - virtual int InsertPacketList( - PacketList* packet_list, - const DecoderDatabase& decoder_database, - absl::optional<uint8_t>* current_rtp_payload_type, - absl::optional<uint8_t>* current_cng_rtp_payload_type, - StatisticsCalculator* stats, - size_t last_decoded_length, - size_t sample_rate, - int target_level_ms); + virtual int InsertPacket(Packet&& packet); // Gets the timestamp for the first packet in the buffer and writes it to the // output variable `next_timestamp`. @@ -119,7 +83,7 @@ class PacketBuffer { // Discards the first packet in the buffer. The packet is deleted. // Returns PacketBuffer::kBufferEmpty if the buffer is empty, // PacketBuffer::kOK otherwise. - virtual int DiscardNextPacket(StatisticsCalculator* stats); + virtual int DiscardNextPacket(); // Discards all packets that are (strictly) older than timestamp_limit, // but newer than timestamp_limit - horizon_samples. Setting horizon_samples @@ -127,16 +91,13 @@ class PacketBuffer { // is, if a packet is more than 2^31 timestamps into the future compared with // timestamp_limit (including wrap-around), it is considered old. virtual void DiscardOldPackets(uint32_t timestamp_limit, - uint32_t horizon_samples, - StatisticsCalculator* stats); + uint32_t horizon_samples); // Discards all packets that are (strictly) older than timestamp_limit. - virtual void DiscardAllOldPackets(uint32_t timestamp_limit, - StatisticsCalculator* stats); + virtual void DiscardAllOldPackets(uint32_t timestamp_limit); // Removes all packets with a specific payload type from the buffer. - virtual void DiscardPacketsWithPayloadType(uint8_t payload_type, - StatisticsCalculator* stats); + virtual void DiscardPacketsWithPayloadType(uint8_t payload_type); // Returns the number of packets in the buffer, including duplicates and // redundant packets. @@ -171,10 +132,12 @@ class PacketBuffer { } private: - absl::optional<SmartFlushingConfig> smart_flushing_config_; + void LogPacketDiscarded(int codec_level); + size_t max_number_of_packets_; PacketList buffer_; const TickTimer* tick_timer_; + StatisticsCalculator* stats_; }; } // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc index b0079645ff..8f307a9eaf 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc @@ -108,26 +108,23 @@ namespace webrtc { TEST(PacketBuffer, CreateAndDestroy) { TickTimer tick_timer; - PacketBuffer* buffer = new PacketBuffer(10, &tick_timer); // 10 packets. + StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer* buffer = + new PacketBuffer(10, &tick_timer, &mock_stats); // 10 packets. EXPECT_TRUE(buffer->Empty()); delete buffer; } TEST(PacketBuffer, InsertPacket) { TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. - PacketGenerator gen(17u, 4711u, 0, 10); StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(10, &tick_timer, &mock_stats); // 10 packets. + PacketGenerator gen(17u, 4711u, 0, 10); MockDecoderDatabase decoder_database; const int payload_len = 100; const Packet packet = gen.NextPacket(payload_len, nullptr); - EXPECT_EQ(0, buffer.InsertPacket(/*packet=*/packet.Clone(), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/10000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + EXPECT_EQ(0, buffer.InsertPacket(/*packet=*/packet.Clone())); uint32_t next_ts; EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts)); EXPECT_EQ(4711u, next_ts); @@ -144,28 +141,22 @@ TEST(PacketBuffer, InsertPacket) { // Test to flush buffer. TEST(PacketBuffer, FlushBuffer) { TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. + StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(10, &tick_timer, &mock_stats); // 10 packets. PacketGenerator gen(0, 0, 0, 10); const int payload_len = 10; - StrictMock<MockStatisticsCalculator> mock_stats; MockDecoderDatabase decoder_database; // Insert 10 small packets; should be ok. for (int i = 0; i < 10; ++i) { - EXPECT_EQ( - PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(/*packet=*/gen.NextPacket( + payload_len, nullptr))); } EXPECT_EQ(10u, buffer.NumPacketsInBuffer()); EXPECT_FALSE(buffer.Empty()); EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10); - buffer.Flush(&mock_stats); + buffer.Flush(); // Buffer should delete the payloads itself. EXPECT_EQ(0u, buffer.NumPacketsInBuffer()); EXPECT_TRUE(buffer.Empty()); @@ -175,23 +166,17 @@ TEST(PacketBuffer, FlushBuffer) { // Test to fill the buffer over the limits, and verify that it flushes. TEST(PacketBuffer, OverfillBuffer) { TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. - PacketGenerator gen(0, 0, 0, 10); StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(10, &tick_timer, &mock_stats); // 10 packets. + PacketGenerator gen(0, 0, 0, 10); MockDecoderDatabase decoder_database; // Insert 10 small packets; should be ok. const int payload_len = 10; int i; for (i = 0; i < 10; ++i) { - EXPECT_EQ( - PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(/*packet=*/gen.NextPacket( + payload_len, nullptr))); } EXPECT_EQ(10u, buffer.NumPacketsInBuffer()); uint32_t next_ts; @@ -202,12 +187,7 @@ TEST(PacketBuffer, OverfillBuffer) { const Packet packet = gen.NextPacket(payload_len, nullptr); // Insert 11th packet; should flush the buffer and insert it after flushing. EXPECT_EQ(PacketBuffer::kFlushed, - buffer.InsertPacket(/*packet=*/packet.Clone(), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + buffer.InsertPacket(/*packet=*/packet.Clone())); EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts)); // Expect last inserted packet to be first in line. @@ -216,190 +196,11 @@ TEST(PacketBuffer, OverfillBuffer) { EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. } -// Test a partial buffer flush. -TEST(PacketBuffer, PartialFlush) { - // Use a field trial to configure smart flushing. - test::ScopedFieldTrials field_trials( - "WebRTC-Audio-NetEqSmartFlushing/enabled:true," - "target_level_threshold_ms:0,target_level_multiplier:2/"); - TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. - PacketGenerator gen(0, 0, 0, 10); - const int payload_len = 10; - StrictMock<MockStatisticsCalculator> mock_stats; - MockDecoderDatabase decoder_database; - - // Insert 10 small packets; should be ok. - for (int i = 0; i < 10; ++i) { - EXPECT_EQ( - PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/100, - /*decoder_database=*/decoder_database)); - } - EXPECT_EQ(10u, buffer.NumPacketsInBuffer()); - EXPECT_FALSE(buffer.Empty()); - - EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(7); - buffer.PartialFlush(/*target_level_ms=*/30, - /*sample_rate=*/1000, - /*last_decoded_length=*/payload_len, - /*stats=*/&mock_stats); - // There should still be some packets left in the buffer. - EXPECT_EQ(3u, buffer.NumPacketsInBuffer()); - EXPECT_FALSE(buffer.Empty()); - EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. -} - -// Test to fill the buffer over the limits, and verify that the smart flush -// functionality works as expected. -TEST(PacketBuffer, SmartFlushOverfillBuffer) { - // Use a field trial to configure smart flushing. - test::ScopedFieldTrials field_trials( - "WebRTC-Audio-NetEqSmartFlushing/enabled:true," - "target_level_threshold_ms:0,target_level_multiplier:2/"); - TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. - PacketGenerator gen(0, 0, 0, 10); - StrictMock<MockStatisticsCalculator> mock_stats; - MockDecoderDatabase decoder_database; - - // Insert 10 small packets; should be ok. - const int payload_len = 10; - int i; - for (i = 0; i < 10; ++i) { - EXPECT_EQ( - PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/100, - /*decoder_database=*/decoder_database)); - } - EXPECT_EQ(10u, buffer.NumPacketsInBuffer()); - uint32_t next_ts; - EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts)); - EXPECT_EQ(0u, next_ts); // Expect first inserted packet to be first in line. - - const Packet packet = gen.NextPacket(payload_len, nullptr); - EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(6); - // Insert 11th packet; should cause a partial flush and insert the packet - // after flushing. - EXPECT_EQ(PacketBuffer::kPartialFlush, - buffer.InsertPacket(/*packet=*/packet.Clone(), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/40, - /*decoder_database=*/decoder_database)); - EXPECT_EQ(5u, buffer.NumPacketsInBuffer()); - EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. -} - -// Test inserting a list of packets. -TEST(PacketBuffer, InsertPacketList) { - TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. - PacketGenerator gen(0, 0, 0, 10); - PacketList list; - const int payload_len = 10; - - // Insert 10 small packets. - for (int i = 0; i < 10; ++i) { - list.push_back(gen.NextPacket(payload_len, nullptr)); - } - - MockDecoderDatabase decoder_database; - auto factory = CreateBuiltinAudioDecoderFactory(); - const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1), - absl::nullopt, factory.get()); - EXPECT_CALL(decoder_database, GetDecoderInfo(0)) - .WillRepeatedly(Return(&info)); - - StrictMock<MockStatisticsCalculator> mock_stats; - - absl::optional<uint8_t> current_pt; - absl::optional<uint8_t> current_cng_pt; - EXPECT_EQ( - PacketBuffer::kOK, - buffer.InsertPacketList(/*packet_list=*/&list, - /*decoder_database=*/decoder_database, - /*current_rtp_payload_type=*/¤t_pt, - /*current_cng_rtp_payload_type=*/¤t_cng_pt, - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/30)); - EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list. - EXPECT_EQ(10u, buffer.NumPacketsInBuffer()); - EXPECT_EQ(0, current_pt); // Current payload type changed to 0. - EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type not changed. - - EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. -} - -// Test inserting a list of packets. Last packet is of a different payload type. -// Expecting the buffer to flush. -// TODO(hlundin): Remove this test when legacy operation is no longer needed. -TEST(PacketBuffer, InsertPacketListChangePayloadType) { - TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. - PacketGenerator gen(0, 0, 0, 10); - PacketList list; - const int payload_len = 10; - - // Insert 10 small packets. - for (int i = 0; i < 10; ++i) { - list.push_back(gen.NextPacket(payload_len, nullptr)); - } - // Insert 11th packet of another payload type (not CNG). - { - Packet packet = gen.NextPacket(payload_len, nullptr); - packet.payload_type = 1; - list.push_back(std::move(packet)); - } - - MockDecoderDatabase decoder_database; - auto factory = CreateBuiltinAudioDecoderFactory(); - const DecoderDatabase::DecoderInfo info0(SdpAudioFormat("pcmu", 8000, 1), - absl::nullopt, factory.get()); - EXPECT_CALL(decoder_database, GetDecoderInfo(0)) - .WillRepeatedly(Return(&info0)); - const DecoderDatabase::DecoderInfo info1(SdpAudioFormat("pcma", 8000, 1), - absl::nullopt, factory.get()); - EXPECT_CALL(decoder_database, GetDecoderInfo(1)) - .WillRepeatedly(Return(&info1)); - - StrictMock<MockStatisticsCalculator> mock_stats; - - absl::optional<uint8_t> current_pt; - absl::optional<uint8_t> current_cng_pt; - EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10); - EXPECT_EQ( - PacketBuffer::kFlushed, - buffer.InsertPacketList(/*packet_list=*/&list, - /*decoder_database=*/decoder_database, - /*current_rtp_payload_type=*/¤t_pt, - /*current_cng_rtp_payload_type=*/¤t_cng_pt, - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/30)); - EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list. - EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); // Only the last packet. - EXPECT_EQ(1, current_pt); // Current payload type changed to 1. - EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type not changed. - - EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. -} TEST(PacketBuffer, ExtractOrderRedundancy) { TickTimer tick_timer; - PacketBuffer buffer(100, &tick_timer); // 100 packets. + StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(100, &tick_timer, &mock_stats); // 100 packets. const int kPackets = 18; const int kFrameSize = 10; const int kPayloadLength = 10; @@ -423,8 +224,6 @@ TEST(PacketBuffer, ExtractOrderRedundancy) { PacketGenerator gen(0, 0, 0, kFrameSize); - StrictMock<MockStatisticsCalculator> mock_stats; - // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction // check ensures that exactly one call to PacketsDiscarded happens in each // DiscardNextPacket call. @@ -444,12 +243,7 @@ TEST(PacketBuffer, ExtractOrderRedundancy) { } EXPECT_CALL(check, Call(i)); EXPECT_EQ(PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/packet.Clone(), - /*stats=*/&mock_stats, - /*last_decoded_length=*/kPayloadLength, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + buffer.InsertPacket(/*packet=*/packet.Clone())); if (packet_facts[i].extract_order >= 0) { expect_order[packet_facts[i].extract_order] = std::move(packet); } @@ -468,25 +262,20 @@ TEST(PacketBuffer, ExtractOrderRedundancy) { TEST(PacketBuffer, DiscardPackets) { TickTimer tick_timer; - PacketBuffer buffer(100, &tick_timer); // 100 packets. + StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(100, &tick_timer, &mock_stats); // 100 packets. const uint16_t start_seq_no = 17; const uint32_t start_ts = 4711; const uint32_t ts_increment = 10; PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment); PacketList list; const int payload_len = 10; - StrictMock<MockStatisticsCalculator> mock_stats; MockDecoderDatabase decoder_database; constexpr int kTotalPackets = 10; // Insert 10 small packets. for (int i = 0; i < kTotalPackets; ++i) { - buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database); + buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr)); } EXPECT_EQ(10u, buffer.NumPacketsInBuffer()); @@ -507,7 +296,7 @@ TEST(PacketBuffer, DiscardPackets) { EXPECT_EQ(current_ts, ts); EXPECT_CALL(mock_stats, PacketsDiscarded(1)); EXPECT_CALL(check, Call(i)); - EXPECT_EQ(PacketBuffer::kOK, buffer.DiscardNextPacket(&mock_stats)); + EXPECT_EQ(PacketBuffer::kOK, buffer.DiscardNextPacket()); current_ts += ts_increment; check.Call(i); } @@ -520,7 +309,7 @@ TEST(PacketBuffer, DiscardPackets) { .Times(kRemainingPackets - kSkipPackets); EXPECT_CALL(check, Call(17)); // Arbitrary id number. buffer.DiscardOldPackets(start_ts + kTotalPackets * ts_increment, - kRemainingPackets * ts_increment, &mock_stats); + kRemainingPackets * ts_increment); check.Call(17); // Same arbitrary id number. EXPECT_EQ(kSkipPackets, buffer.NumPacketsInBuffer()); @@ -530,8 +319,7 @@ TEST(PacketBuffer, DiscardPackets) { // Discard all remaining packets. EXPECT_CALL(mock_stats, PacketsDiscarded(kSkipPackets)); - buffer.DiscardAllOldPackets(start_ts + kTotalPackets * ts_increment, - &mock_stats); + buffer.DiscardAllOldPackets(start_ts + kTotalPackets * ts_increment); EXPECT_TRUE(buffer.Empty()); EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. @@ -539,7 +327,8 @@ TEST(PacketBuffer, DiscardPackets) { TEST(PacketBuffer, Reordering) { TickTimer tick_timer; - PacketBuffer buffer(100, &tick_timer); // 100 packets. + StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(100, &tick_timer, &mock_stats); // 100 packets. const uint16_t start_seq_no = 17; const uint32_t start_ts = 4711; const uint32_t ts_increment = 10; @@ -559,27 +348,9 @@ TEST(PacketBuffer, Reordering) { } } - MockDecoderDatabase decoder_database; - auto factory = CreateBuiltinAudioDecoderFactory(); - const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1), - absl::nullopt, factory.get()); - EXPECT_CALL(decoder_database, GetDecoderInfo(0)) - .WillRepeatedly(Return(&info)); - absl::optional<uint8_t> current_pt; - absl::optional<uint8_t> current_cng_pt; - - StrictMock<MockStatisticsCalculator> mock_stats; - - EXPECT_EQ( - PacketBuffer::kOK, - buffer.InsertPacketList(/*packet_list=*/&list, - /*decoder_database=*/decoder_database, - /*current_rtp_payload_type=*/¤t_pt, - /*current_cng_rtp_payload_type=*/¤t_cng_pt, - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/30)); + for (Packet& packet : list) { + EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(std::move(packet))); + } EXPECT_EQ(10u, buffer.NumPacketsInBuffer()); // Extract them and make sure that come out in the right order. @@ -591,86 +362,6 @@ TEST(PacketBuffer, Reordering) { current_ts += ts_increment; } EXPECT_TRUE(buffer.Empty()); - - EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. -} - -// The test first inserts a packet with narrow-band CNG, then a packet with -// wide-band speech. The expected behavior of the packet buffer is to detect a -// change in sample rate, even though no speech packet has been inserted before, -// and flush out the CNG packet. -TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) { - TickTimer tick_timer; - PacketBuffer buffer(10, &tick_timer); // 10 packets. - const uint8_t kCngPt = 13; - const int kPayloadLen = 10; - const uint8_t kSpeechPt = 100; - - MockDecoderDatabase decoder_database; - auto factory = CreateBuiltinAudioDecoderFactory(); - const DecoderDatabase::DecoderInfo info_cng(SdpAudioFormat("cn", 8000, 1), - absl::nullopt, factory.get()); - EXPECT_CALL(decoder_database, GetDecoderInfo(kCngPt)) - .WillRepeatedly(Return(&info_cng)); - const DecoderDatabase::DecoderInfo info_speech( - SdpAudioFormat("l16", 16000, 1), absl::nullopt, factory.get()); - EXPECT_CALL(decoder_database, GetDecoderInfo(kSpeechPt)) - .WillRepeatedly(Return(&info_speech)); - - // Insert first packet, which is narrow-band CNG. - PacketGenerator gen(0, 0, kCngPt, 10); - PacketList list; - list.push_back(gen.NextPacket(kPayloadLen, nullptr)); - absl::optional<uint8_t> current_pt; - absl::optional<uint8_t> current_cng_pt; - - StrictMock<MockStatisticsCalculator> mock_stats; - - EXPECT_EQ( - PacketBuffer::kOK, - buffer.InsertPacketList(/*packet_list=*/&list, - /*decoder_database=*/decoder_database, - /*current_rtp_payload_type=*/¤t_pt, - /*current_cng_rtp_payload_type=*/¤t_cng_pt, - /*stats=*/&mock_stats, - /*last_decoded_length=*/kPayloadLen, - /*sample_rate=*/1000, - /*target_level_ms=*/30)); - EXPECT_TRUE(list.empty()); - EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); - ASSERT_TRUE(buffer.PeekNextPacket()); - EXPECT_EQ(kCngPt, buffer.PeekNextPacket()->payload_type); - EXPECT_EQ(current_pt, absl::nullopt); // Current payload type not set. - EXPECT_EQ(kCngPt, current_cng_pt); // CNG payload type set. - - // Insert second packet, which is wide-band speech. - { - Packet packet = gen.NextPacket(kPayloadLen, nullptr); - packet.payload_type = kSpeechPt; - list.push_back(std::move(packet)); - } - // Expect the buffer to flush out the CNG packet, since it does not match the - // new speech sample rate. - EXPECT_CALL(mock_stats, PacketsDiscarded(1)); - EXPECT_EQ( - PacketBuffer::kFlushed, - buffer.InsertPacketList(/*packet_list=*/&list, - /*decoder_database=*/decoder_database, - /*current_rtp_payload_type=*/¤t_pt, - /*current_cng_rtp_payload_type=*/¤t_cng_pt, - /*stats=*/&mock_stats, - /*last_decoded_length=*/kPayloadLen, - /*sample_rate=*/1000, - /*target_level_ms=*/30)); - EXPECT_TRUE(list.empty()); - EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); - ASSERT_TRUE(buffer.PeekNextPacket()); - EXPECT_EQ(kSpeechPt, buffer.PeekNextPacket()->payload_type); - - EXPECT_EQ(kSpeechPt, current_pt); // Current payload type set. - EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type reset. - - EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. } TEST(PacketBuffer, Failures) { @@ -681,80 +372,26 @@ TEST(PacketBuffer, Failures) { PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment); TickTimer tick_timer; StrictMock<MockStatisticsCalculator> mock_stats; - MockDecoderDatabase decoder_database; - PacketBuffer* buffer = new PacketBuffer(100, &tick_timer); // 100 packets. + PacketBuffer buffer(100, &tick_timer, &mock_stats); // 100 packets. { Packet packet = gen.NextPacket(payload_len, nullptr); packet.payload.Clear(); EXPECT_EQ(PacketBuffer::kInvalidPacket, - buffer->InsertPacket(/*packet=*/std::move(packet), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + buffer.InsertPacket(/*packet=*/std::move(packet))); } // Buffer should still be empty. Test all empty-checks. uint32_t temp_ts; - EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer->NextTimestamp(&temp_ts)); + EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer.NextTimestamp(&temp_ts)); EXPECT_EQ(PacketBuffer::kBufferEmpty, - buffer->NextHigherTimestamp(0, &temp_ts)); - EXPECT_EQ(NULL, buffer->PeekNextPacket()); - EXPECT_FALSE(buffer->GetNextPacket()); + buffer.NextHigherTimestamp(0, &temp_ts)); + EXPECT_EQ(NULL, buffer.PeekNextPacket()); + EXPECT_FALSE(buffer.GetNextPacket()); // Discarding packets will not invoke mock_stats.PacketDiscarded() because the // packet buffer is empty. - EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer->DiscardNextPacket(&mock_stats)); - buffer->DiscardAllOldPackets(0, &mock_stats); - - // Insert one packet to make the buffer non-empty. - EXPECT_EQ( - PacketBuffer::kOK, - buffer->InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr), - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); - EXPECT_EQ(PacketBuffer::kInvalidPointer, buffer->NextTimestamp(NULL)); - EXPECT_EQ(PacketBuffer::kInvalidPointer, - buffer->NextHigherTimestamp(0, NULL)); - delete buffer; - - // Insert packet list of three packets, where the second packet has an invalid - // payload. Expect first packet to be inserted, and the remaining two to be - // discarded. - buffer = new PacketBuffer(100, &tick_timer); // 100 packets. - PacketList list; - list.push_back(gen.NextPacket(payload_len, nullptr)); // Valid packet. - { - Packet packet = gen.NextPacket(payload_len, nullptr); - packet.payload.Clear(); // Invalid. - list.push_back(std::move(packet)); - } - list.push_back(gen.NextPacket(payload_len, nullptr)); // Valid packet. - auto factory = CreateBuiltinAudioDecoderFactory(); - const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1), - absl::nullopt, factory.get()); - EXPECT_CALL(decoder_database, GetDecoderInfo(0)) - .WillRepeatedly(Return(&info)); - absl::optional<uint8_t> current_pt; - absl::optional<uint8_t> current_cng_pt; - EXPECT_EQ( - PacketBuffer::kInvalidPacket, - buffer->InsertPacketList(/*packet_list=*/&list, - /*decoder_database=*/decoder_database, - /*current_rtp_payload_type=*/¤t_pt, - /*current_cng_rtp_payload_type=*/¤t_cng_pt, - /*stats=*/&mock_stats, - /*last_decoded_length=*/payload_len, - /*sample_rate=*/1000, - /*target_level_ms=*/30)); - EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list. - EXPECT_EQ(1u, buffer->NumPacketsInBuffer()); - delete buffer; - EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. + EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer.DiscardNextPacket()); + buffer.DiscardAllOldPackets(0); } // Test packet comparison function. @@ -873,9 +510,9 @@ TEST(PacketBuffer, GetSpanSamples) { constexpr int kSampleRateHz = 48000; constexpr bool kCountWaitingTime = false; TickTimer tick_timer; - PacketBuffer buffer(3, &tick_timer); - PacketGenerator gen(0, kStartTimeStamp, 0, kFrameSizeSamples); StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(3, &tick_timer, &mock_stats); + PacketGenerator gen(0, kStartTimeStamp, 0, kFrameSizeSamples); MockDecoderDatabase decoder_database; Packet packet_1 = gen.NextPacket(kPayloadSizeBytes, nullptr); @@ -891,12 +528,7 @@ TEST(PacketBuffer, GetSpanSamples) { packet_2.timestamp); // Tmestamp wrapped around. EXPECT_EQ(PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/std::move(packet_1), - /*stats=*/&mock_stats, - /*last_decoded_length=*/kFrameSizeSamples, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + buffer.InsertPacket(/*packet=*/std::move(packet_1))); constexpr size_t kLastDecodedSizeSamples = 2; // packet_1 has no access to duration, and relies last decoded duration as @@ -906,12 +538,7 @@ TEST(PacketBuffer, GetSpanSamples) { kCountWaitingTime)); EXPECT_EQ(PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/std::move(packet_2), - /*stats=*/&mock_stats, - /*last_decoded_length=*/kFrameSizeSamples, - /*sample_rate=*/1000, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + buffer.InsertPacket(/*packet=*/std::move(packet_2))); EXPECT_EQ(kFrameSizeSamples * 2, buffer.GetSpanSamples(0, kSampleRateHz, kCountWaitingTime)); @@ -931,20 +558,15 @@ TEST(PacketBuffer, GetSpanSamplesCountWaitingTime) { constexpr bool kCountWaitingTime = true; constexpr size_t kLastDecodedSizeSamples = 0; TickTimer tick_timer; - PacketBuffer buffer(3, &tick_timer); - PacketGenerator gen(0, kStartTimeStamp, 0, kFrameSizeSamples); StrictMock<MockStatisticsCalculator> mock_stats; + PacketBuffer buffer(3, &tick_timer, &mock_stats); + PacketGenerator gen(0, kStartTimeStamp, 0, kFrameSizeSamples); MockDecoderDatabase decoder_database; Packet packet = gen.NextPacket(kPayloadSizeBytes, nullptr); EXPECT_EQ(PacketBuffer::kOK, - buffer.InsertPacket(/*packet=*/std::move(packet), - /*stats=*/&mock_stats, - /*last_decoded_length=*/kFrameSizeSamples, - /*sample_rate=*/kSampleRateHz, - /*target_level_ms=*/60, - /*decoder_database=*/decoder_database)); + buffer.InsertPacket(/*packet=*/std::move(packet))); EXPECT_EQ(0u, buffer.GetSpanSamples(kLastDecodedSizeSamples, kSampleRateHz, kCountWaitingTime)); diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc index e6c1809fb6..e626d09c99 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc @@ -19,13 +19,13 @@ #include "test/testsupport/file_utils.h" #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT -RTC_PUSH_IGNORING_WUNDEF() + #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h" #else #include "modules/audio_coding/neteq/neteq_unittest.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() + #endif namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc index f5d50dc859..fee7b49eb3 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc @@ -13,19 +13,18 @@ #include <string> #include "absl/strings/string_view.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/message_digest.h" #include "rtc_base/string_encode.h" #include "test/gtest.h" #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT -RTC_PUSH_IGNORING_WUNDEF() + #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h" #else #include "modules/audio_coding/neteq/neteq_unittest.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() + #endif namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build index 04dbb03279..834a8d1265 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build @@ -234,7 +234,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -244,10 +243,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/pcm16b_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/pcm16b_c_gn/moz.build index 41f722069c..ef0c150cb8 100644 --- a/third_party/libwebrtc/modules/audio_coding/pcm16b_c_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/pcm16b_c_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/pcm16b_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/pcm16b_gn/moz.build index ed96e7c0f8..a1d9c8009d 100644 --- a/third_party/libwebrtc/modules/audio_coding/pcm16b_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/pcm16b_gn/moz.build @@ -197,7 +197,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -207,10 +206,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/red_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/red_gn/moz.build index 479cf67a2a..ab0d8129bb 100644 --- a/third_party/libwebrtc/modules/audio_coding/red_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/red_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_cng_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_cng_gn/moz.build index a8a6c576e2..d077aaa930 100644 --- a/third_party/libwebrtc/modules/audio_coding/webrtc_cng_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/webrtc_cng_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_multiopus_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_multiopus_gn/moz.build index 491f0cc543..d48fd68174 100644 --- a/third_party/libwebrtc/modules/audio_coding/webrtc_multiopus_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/webrtc_multiopus_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_opus_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_gn/moz.build index e2c57b99af..02986beaa4 100644 --- a/third_party/libwebrtc/modules/audio_coding/webrtc_opus_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_gn/moz.build @@ -204,7 +204,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -214,10 +213,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_opus_wrapper_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_wrapper_gn/moz.build index 268854264f..e6c31b48b5 100644 --- a/third_party/libwebrtc/modules/audio_coding/webrtc_opus_wrapper_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_wrapper_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_device/BUILD.gn b/third_party/libwebrtc/modules/audio_device/BUILD.gn index 4726f93279..a135f042db 100644 --- a/third_party/libwebrtc/modules/audio_device/BUILD.gn +++ b/third_party/libwebrtc/modules/audio_device/BUILD.gn @@ -50,6 +50,7 @@ rtc_source_set("audio_device_api") { "include/audio_device_defines.h", ] deps = [ + "../../api:ref_count", "../../api:scoped_refptr", "../../api/task_queue", "../../rtc_base:checks", @@ -490,7 +491,6 @@ if (rtc_include_tests && !build_with_chromium && !build_with_mozilla) { "../../common_audio", "../../rtc_base:buffer", "../../rtc_base:checks", - "../../rtc_base:ignore_wundef", "../../rtc_base:logging", "../../rtc_base:macromagic", "../../rtc_base:race_checker", diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_gn/moz.build b/third_party/libwebrtc/modules/audio_device/audio_device_gn/moz.build index df00e056c6..4128efbbf8 100644 --- a/third_party/libwebrtc/modules/audio_device/audio_device_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_device/audio_device_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device.h b/third_party/libwebrtc/modules/audio_device/include/audio_device.h index 936ee6cb04..47d2aecfa7 100644 --- a/third_party/libwebrtc/modules/audio_device/include/audio_device.h +++ b/third_party/libwebrtc/modules/audio_device/include/audio_device.h @@ -12,16 +12,16 @@ #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_ #include "absl/types/optional.h" +#include "api/ref_count.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" #include "modules/audio_device/include/audio_device_defines.h" -#include "rtc_base/ref_count.h" namespace webrtc { class AudioDeviceModuleForTest; -class AudioDeviceModule : public rtc::RefCountInterface { +class AudioDeviceModule : public webrtc::RefCountInterface { public: enum AudioLayer { kPlatformDefaultAudio = 0, diff --git a/third_party/libwebrtc/modules/audio_device/include/fake_audio_device.h b/third_party/libwebrtc/modules/audio_device/include/fake_audio_device.h index 2322ce0263..2a303173e9 100644 --- a/third_party/libwebrtc/modules/audio_device/include/fake_audio_device.h +++ b/third_party/libwebrtc/modules/audio_device/include/fake_audio_device.h @@ -23,8 +23,8 @@ class FakeAudioDeviceModule // references using scoped_refptr. Current code doesn't always use refcounting // for this class. void AddRef() const override {} - rtc::RefCountReleaseStatus Release() const override { - return rtc::RefCountReleaseStatus::kDroppedLastRef; + webrtc::RefCountReleaseStatus Release() const override { + return webrtc::RefCountReleaseStatus::kDroppedLastRef; } }; diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_gn/moz.build b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_gn/moz.build index edfac56a3a..cc60512cda 100644 --- a/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_gn/moz.build b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_gn/moz.build index 7108d9fbe1..6595939941 100644 --- a/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_gn/moz.build @@ -202,7 +202,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -212,10 +211,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/BUILD.gn index 2b81427da9..6aca7dee46 100644 --- a/third_party/libwebrtc/modules/audio_processing/BUILD.gn +++ b/third_party/libwebrtc/modules/audio_processing/BUILD.gn @@ -29,6 +29,7 @@ rtc_library("api") { ":audio_frame_view", ":audio_processing_statistics", "../../api:array_view", + "../../api:ref_count", "../../api:scoped_refptr", "../../api/audio:aec3_config", "../../api/audio:audio_frame_api", @@ -190,7 +191,6 @@ rtc_library("audio_processing") { "../../rtc_base:checks", "../../rtc_base:event_tracer", "../../rtc_base:gtest_prod", - "../../rtc_base:ignore_wundef", "../../rtc_base:logging", "../../rtc_base:macromagic", "../../rtc_base:safe_minmax", @@ -397,7 +397,6 @@ if (rtc_include_tests) { "../../common_audio:common_audio_c", "../../rtc_base:checks", "../../rtc_base:gtest_prod", - "../../rtc_base:ignore_wundef", "../../rtc_base:macromagic", "../../rtc_base:platform_thread", "../../rtc_base:protobuf_utils", @@ -573,7 +572,6 @@ if (rtc_include_tests) { "../../api/audio:echo_detector_creator", "../../common_audio", "../../rtc_base:checks", - "../../rtc_base:ignore_wundef", "../../rtc_base:logging", "../../rtc_base:protobuf_utils", "../../rtc_base:rtc_json", @@ -613,7 +611,6 @@ if (rtc_include_tests) { deps = [ ":audioproc_debug_proto", "../../rtc_base:checks", - "../../rtc_base:ignore_wundef", "../../rtc_base:protobuf_utils", "../../rtc_base/system:arch", ] diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_gn/moz.build index f21e65fb4a..7435b6a457 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_gn/moz.build index b9c819893f..0d2471073d 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_gn/moz.build @@ -191,16 +191,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_avx2_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_avx2_gn/moz.build index 097e67bbe5..f9844b6521 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_avx2_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_avx2_gn/moz.build @@ -181,10 +181,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86": CXXFLAGS += [ diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common_gn/moz.build index 955fe2022f..8d9c4e6bd7 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_gn/moz.build index 154d9f4406..d403ae8b96 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_gn/moz.build @@ -191,16 +191,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_gn/moz.build index 7ad4cffedf..85e5654231 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_gn/moz.build @@ -256,7 +256,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -266,10 +265,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_gn/moz.build index 0084077435..aa1aaf15d9 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_gn/moz.build index be2c3bbf56..0ebdb0798f 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_gn/moz.build index 2bd3ae0c01..6444c3137f 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_gn/moz.build index e40fdb1cf1..9cf3a7842a 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn index 38d8776258..78bae56835 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn @@ -66,7 +66,6 @@ if (rtc_enable_protobuf) { "../../../api/audio:audio_frame_api", "../../../api/task_queue", "../../../rtc_base:checks", - "../../../rtc_base:ignore_wundef", "../../../rtc_base:logging", "../../../rtc_base:macromagic", "../../../rtc_base:protobuf_utils", diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_gn/moz.build index f1280fed0d..13420467de 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_gn/moz.build @@ -187,16 +187,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h index fac3712b7a..429808f9af 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h @@ -17,20 +17,17 @@ #include "modules/audio_processing/aec_dump/capture_stream_info.h" #include "modules/audio_processing/include/aec_dump.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/race_checker.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/task_queue.h" #include "rtc_base/thread_annotations.h" // Files generated at build-time by the protobuf compiler. -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" #else #include "modules/audio_processing/debug.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h index 0819bbcb23..572990c150 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h @@ -15,16 +15,13 @@ #include <utility> #include "modules/audio_processing/include/aec_dump.h" -#include "rtc_base/ignore_wundef.h" // Files generated at build-time by the protobuf compiler. -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" #else #include "modules/audio_processing/debug.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory_gn/moz.build index 2966151ab6..4e8b16442d 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump_interface_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec_dump_interface_gn/moz.build index 02b847ed76..89ae508073 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump_interface_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump_interface_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_gn/moz.build index 9ff36991fe..ca3ffeb81e 100644 --- a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_gn/moz.build @@ -238,7 +238,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -248,10 +247,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/agc_gn/moz.build index f6f4442cfc..f26489f413 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc/agc_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc/agc_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc/gain_control_interface_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/gain_control_interface_gn/moz.build index be6b4f9b27..ebf241f7a5 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc/gain_control_interface_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc/gain_control_interface_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy_agc_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/legacy_agc_gn/moz.build index 4e6e295d34..aa5c6835cc 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc/legacy_agc_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy_agc_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc/level_estimation_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/level_estimation_gn/moz.build index 64ffa75960..a272555662 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc/level_estimation_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc/level_estimation_gn/moz.build @@ -201,7 +201,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -211,10 +210,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_gn/moz.build index 9473ac62f5..5e3b5801ad 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_gn/moz.build index c7a2f6d215..d9520efe2f 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_gn/moz.build index e70e3f68e9..dbf53e8e8e 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/common_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/common_gn/moz.build index 8690613542..44307f4147 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/common_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/common_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features_gn/moz.build index 4b0431db1a..e842cac9c3 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_gn/moz.build index 1b8da82f58..60614d4cc1 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_gn/moz.build @@ -202,7 +202,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -212,10 +211,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_gn/moz.build index bea71dcee3..691900e356 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_gn/moz.build index 03eb1fb3a1..ee04e973fb 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_gn/moz.build index f1a841d5ae..0bde4db9d4 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_gn/moz.build @@ -201,7 +201,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -211,10 +210,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_gn/moz.build index 40448f68a9..b7d0a9ba88 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_gn/moz.build index 9d4629e9ab..210539ab46 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build index 134ffac5fd..7965a026ef 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build index cf3de48a57..bdfe90cf16 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build @@ -191,16 +191,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build index dbb926c5fc..6a73ce96e4 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build index 92da260f90..27b40f13a5 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build index cedb17bc22..d66ed412b2 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build index d45bc78ff6..0ddc85f5ac 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build index 20da5f3615..25e813a226 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build index b0ba79562e..f54dd88a23 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build index 2d8396fa2a..d8b88047d7 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build index 143ba6960c..4a3c5bf28b 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build index d4dd169f15..01313fa460 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build @@ -176,10 +176,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86": CXXFLAGS += [ diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build index 09fe0c3d24..3f88913309 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build @@ -191,16 +191,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_gn/moz.build index 6b8def8650..6562d840b7 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_gn/moz.build @@ -201,7 +201,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -211,10 +210,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_gn/moz.build index 8f2996fa26..3afaa88450 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_gn/moz.build index 55cfbb60e7..3aa09832b2 100644 --- a/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/api_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/api_gn/moz.build index 7a02b7e10c..37e50af014 100644 --- a/third_party/libwebrtc/modules/audio_processing/api_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/api_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/apm_logging_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/apm_logging_gn/moz.build index 992376cd8a..53fd9d9f94 100644 --- a/third_party/libwebrtc/modules/audio_processing/apm_logging_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/apm_logging_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/audio_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_buffer_gn/moz.build index 88031a747d..2087aeb909 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_buffer_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/audio_buffer_gn/moz.build @@ -202,7 +202,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -212,10 +211,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/audio_frame_proxies_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_frame_proxies_gn/moz.build index 7e73b70483..737ca5e834 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_frame_proxies_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/audio_frame_proxies_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/audio_frame_view_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_gn/moz.build index 0f81755091..b7391a78b1 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_frame_view_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_processing_gn/moz.build index 7dc22bcf2b..5b4f4d5d54 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_gn/moz.build @@ -206,7 +206,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -216,10 +215,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h index fe80e0d912..1e058b5a32 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h @@ -43,7 +43,6 @@ #include "modules/audio_processing/rms_level.h" #include "modules/audio_processing/transient/transient_suppressor.h" #include "rtc_base/gtest_prod_util.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/swap_queue.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_unittest.cc index 9e50f994b1..e03f966b06 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_unittest.cc +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_unittest.cc @@ -48,7 +48,7 @@ class MockInitialize : public AudioProcessingImpl { } MOCK_METHOD(void, AddRef, (), (const, override)); - MOCK_METHOD(rtc::RefCountReleaseStatus, Release, (), (const, override)); + MOCK_METHOD(RefCountReleaseStatus, Release, (), (const, override)); }; // Creates MockEchoControl instances and provides a raw pointer access to diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_statistics_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_processing_statistics_gn/moz.build index 6d174505ed..6b3e54c3f7 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_statistics_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_statistics_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc index e320e71405..c2bedb2da4 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc @@ -38,7 +38,6 @@ #include "rtc_base/checks.h" #include "rtc_base/fake_clock.h" #include "rtc_base/gtest_prod_util.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/numerics/safe_minmax.h" #include "rtc_base/protobuf_utils.h" @@ -51,14 +50,13 @@ #include "test/gtest.h" #include "test/testsupport/file_utils.h" -RTC_PUSH_IGNORING_WUNDEF() -#include "modules/audio_processing/debug.pb.h" #ifdef WEBRTC_ANDROID_PLATFORM_BUILD +#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" #include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h" #else +#include "modules/audio_processing/debug.pb.h" #include "modules/audio_processing/test/unittest.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() ABSL_FLAG(bool, write_apm_ref_data, diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_gn/moz.build index d80a3bb1c6..ad198344e2 100644 --- a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/gain_controller2_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/gain_controller2_gn/moz.build index d6d9d3658b..ab31e68564 100644 --- a/third_party/libwebrtc/modules/audio_processing/gain_controller2_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/gain_controller2_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/high_pass_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/high_pass_filter_gn/moz.build index 8769a3a318..af06d4142f 100644 --- a/third_party/libwebrtc/modules/audio_processing/high_pass_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/high_pass_filter_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h index f613a38de1..e3223513af 100644 --- a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h +++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h @@ -28,10 +28,10 @@ #include "api/array_view.h" #include "api/audio/echo_canceller3_config.h" #include "api/audio/echo_control.h" +#include "api/ref_count.h" #include "api/scoped_refptr.h" #include "modules/audio_processing/include/audio_processing_statistics.h" #include "rtc_base/arraysize.h" -#include "rtc_base/ref_count.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/system/rtc_export.h" @@ -127,7 +127,7 @@ class CustomProcessing; // // Close the application... // apm.reset(); // -class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface { +class RTC_EXPORT AudioProcessing : public RefCountInterface { public: // The struct below constitutes the new parameter scheme for the audio // processing. It is being introduced gradually and until it is fully @@ -912,7 +912,7 @@ class CustomProcessing { }; // Interface for an echo detector submodule. -class EchoDetector : public rtc::RefCountInterface { +class EchoDetector : public RefCountInterface { public: // (Re-)Initializes the submodule. virtual void Initialize(int capture_sample_rate_hz, diff --git a/third_party/libwebrtc/modules/audio_processing/ns/ns_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/ns/ns_gn/moz.build index 14595abaf9..ac1c19134a 100644 --- a/third_party/libwebrtc/modules/audio_processing/ns/ns_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/ns/ns_gn/moz.build @@ -212,7 +212,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -222,10 +221,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators_gn/moz.build index af79a781f6..da2a1b7ae3 100644 --- a/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/rms_level_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/rms_level_gn/moz.build index 23f52652ae..d0c4b2bd8e 100644 --- a/third_party/libwebrtc/modules/audio_processing/rms_level_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/rms_level_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h index e2c1f3e4ba..4713c800ec 100644 --- a/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h +++ b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h @@ -15,15 +15,12 @@ #include <string> #include "modules/audio_processing/test/audio_processing_simulator.h" -#include "rtc_base/ignore_wundef.h" -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" #else #include "modules/audio_processing/debug.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() namespace webrtc { namespace test { diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h index be21c68663..077147eb68 100644 --- a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h +++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h @@ -16,11 +16,9 @@ #include "absl/strings/string_view.h" #include "common_audio/channel_buffer.h" #include "modules/audio_processing/include/audio_processing.h" -#include "rtc_base/ignore_wundef.h" -RTC_PUSH_IGNORING_WUNDEF() +// Generated at build-time by the protobuf compiler. #include "modules/audio_processing/debug.pb.h" -RTC_POP_IGNORING_WUNDEF() namespace webrtc { namespace test { diff --git a/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.h b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.h index b9c2e819f9..eb93383f5a 100644 --- a/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.h +++ b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.h @@ -14,12 +14,10 @@ #include <memory> #include <sstream> // no-presubmit-check TODO(webrtc:8982) -#include "rtc_base/ignore_wundef.h" #include "rtc_base/protobuf_utils.h" -RTC_PUSH_IGNORING_WUNDEF() +// Generated at build-time by the protobuf compiler. #include "modules/audio_processing/debug.pb.h" -RTC_POP_IGNORING_WUNDEF() namespace webrtc { diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_api_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_api_gn/moz.build index 6310b948ac..44571715b8 100644 --- a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_api_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_api_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl_gn/moz.build index 31e0736f30..d700fc1a32 100644 --- a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl_gn/moz.build @@ -203,7 +203,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -213,10 +212,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_gn/moz.build index 52587c0890..c67675f431 100644 --- a/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_gn/moz.build index 02813d2513..b6566a8950 100644 --- a/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/utility/legacy_delay_estimator_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/utility/legacy_delay_estimator_gn/moz.build index 67c6a218f6..c20d5b6189 100644 --- a/third_party/libwebrtc/modules/audio_processing/utility/legacy_delay_estimator_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/utility/legacy_delay_estimator_gn/moz.build @@ -189,7 +189,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -199,10 +198,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_gn/moz.build index 3213706005..936decab70 100644 --- a/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/vad/vad_gn/moz.build index 3739bbef8a..0ae31f5a2e 100644 --- a/third_party/libwebrtc/modules/audio_processing/vad/vad_gn/moz.build +++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_gn/moz.build @@ -206,7 +206,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -216,10 +215,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build index b5bcafa45f..1190193b94 100644 --- a/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build @@ -202,7 +202,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -212,10 +211,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build index b48fc38c39..40fd1189aa 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build index 31d8c420f6..e2087c6126 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build @@ -202,7 +202,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -212,10 +211,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build index e233806b43..7b77d3dc86 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build @@ -205,7 +205,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -215,10 +214,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build index 147a08113b..0e5182a469 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build @@ -201,7 +201,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -211,10 +210,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build index 0ee8a34df8..04b78b5988 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build index 5931292efe..d290fbe9ec 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build @@ -196,7 +196,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -206,10 +205,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc index ef200869a6..8e1a3c4698 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc @@ -132,7 +132,7 @@ LossBasedBweV2::LossBasedBweV2(const FieldTrialsView* key_value_config) instant_upper_bound_temporal_weights_.resize( config_->observation_window_size); CalculateTemporalWeights(); - hold_duration_ = kInitHoldDuration; + last_hold_info_.duration = kInitHoldDuration; } bool LossBasedBweV2::IsEnabled() const { @@ -149,6 +149,10 @@ bool LossBasedBweV2::ReadyToUseInStartPhase() const { return IsReady() && config_->use_in_start_phase; } +bool LossBasedBweV2::UseInStartPhase() const { + return config_->use_in_start_phase; +} + LossBasedBweV2::Result LossBasedBweV2::GetLossBasedResult() const { if (!IsReady()) { if (!IsEnabled()) { @@ -289,50 +293,69 @@ void LossBasedBweV2::UpdateBandwidthEstimate( /*new_estimate=*/best_candidate.loss_limited_bandwidth); // Bound the best candidate by the acked bitrate. if (increasing_when_loss_limited && IsValid(acknowledged_bitrate_)) { + double rampup_factor = config_->bandwidth_rampup_upper_bound_factor; + if (IsValid(last_hold_info_.rate) && + acknowledged_bitrate_ < + config_->bandwidth_rampup_hold_threshold * last_hold_info_.rate) { + rampup_factor = config_->bandwidth_rampup_upper_bound_factor_in_hold; + } + best_candidate.loss_limited_bandwidth = std::max(current_best_estimate_.loss_limited_bandwidth, std::min(best_candidate.loss_limited_bandwidth, - config_->bandwidth_rampup_upper_bound_factor * - (*acknowledged_bitrate_))); + rampup_factor * (*acknowledged_bitrate_))); + // Increase current estimate by at least 1kbps to make sure that the state + // will be switched to kIncreasing, thus padding is triggered. + if (loss_based_result_.state == LossBasedState::kDecreasing && + best_candidate.loss_limited_bandwidth == + current_best_estimate_.loss_limited_bandwidth) { + best_candidate.loss_limited_bandwidth = + current_best_estimate_.loss_limited_bandwidth + + DataRate::BitsPerSec(1); + } } } - current_best_estimate_ = best_candidate; - UpdateResult(); - - if (IsInLossLimitedState() && - (recovering_after_loss_timestamp_.IsInfinite() || - recovering_after_loss_timestamp_ + config_->delayed_increase_window < - last_send_time_most_recent_observation_)) { - bandwidth_limit_in_current_window_ = - std::max(kCongestionControllerMinBitrate, - current_best_estimate_.loss_limited_bandwidth * - config_->max_increase_factor); - recovering_after_loss_timestamp_ = last_send_time_most_recent_observation_; - } -} - -void LossBasedBweV2::UpdateResult() { DataRate bounded_bandwidth_estimate = DataRate::PlusInfinity(); if (IsValid(delay_based_estimate_)) { bounded_bandwidth_estimate = std::max(GetInstantLowerBound(), - std::min({current_best_estimate_.loss_limited_bandwidth, + std::min({best_candidate.loss_limited_bandwidth, GetInstantUpperBound(), delay_based_estimate_})); } else { - bounded_bandwidth_estimate = - std::max(GetInstantLowerBound(), - std::min(current_best_estimate_.loss_limited_bandwidth, - GetInstantUpperBound())); + bounded_bandwidth_estimate = std::max( + GetInstantLowerBound(), std::min(best_candidate.loss_limited_bandwidth, + GetInstantUpperBound())); + } + if (config_->bound_best_candidate && + bounded_bandwidth_estimate < best_candidate.loss_limited_bandwidth) { + RTC_LOG(LS_INFO) << "Resetting loss based BWE to " + << bounded_bandwidth_estimate.kbps() + << "due to loss. Avg loss rate: " + << GetAverageReportedLossRatio(); + current_best_estimate_.loss_limited_bandwidth = bounded_bandwidth_estimate; + current_best_estimate_.inherent_loss = 0; + } else { + current_best_estimate_ = best_candidate; + if (config_->lower_bound_by_acked_rate_factor > 0.0) { + current_best_estimate_.loss_limited_bandwidth = + std::max(current_best_estimate_.loss_limited_bandwidth, + GetInstantLowerBound()); + } } if (loss_based_result_.state == LossBasedState::kDecreasing && - last_hold_timestamp_ > last_send_time_most_recent_observation_ && + last_hold_info_.timestamp > last_send_time_most_recent_observation_ && bounded_bandwidth_estimate < delay_based_estimate_) { - // BWE is not allowed to increase during the HOLD duration. The purpose of + // Ensure that acked rate is the lower bound of HOLD rate. + if (config_->lower_bound_by_acked_rate_factor > 0.0) { + last_hold_info_.rate = + std::max(GetInstantLowerBound(), last_hold_info_.rate); + } + // BWE is not allowed to increase above the HOLD rate. The purpose of // HOLD is to not immediately ramp up BWE to a rate that may cause loss. - loss_based_result_.bandwidth_estimate = std::min( - loss_based_result_.bandwidth_estimate, bounded_bandwidth_estimate); + loss_based_result_.bandwidth_estimate = + std::min(last_hold_info_.rate, bounded_bandwidth_estimate); return; } @@ -359,22 +382,38 @@ void LossBasedBweV2::UpdateResult() { RTC_LOG(LS_INFO) << this << " " << "Switch to HOLD. Bounded BWE: " << bounded_bandwidth_estimate.kbps() - << ", duration: " << hold_duration_.seconds(); - last_hold_timestamp_ = - last_send_time_most_recent_observation_ + hold_duration_; - hold_duration_ = std::min(kMaxHoldDuration, - hold_duration_ * config_->hold_duration_factor); + << ", duration: " << last_hold_info_.duration.ms(); + last_hold_info_ = { + .timestamp = last_send_time_most_recent_observation_ + + last_hold_info_.duration, + .duration = + std::min(kMaxHoldDuration, last_hold_info_.duration * + config_->hold_duration_factor), + .rate = bounded_bandwidth_estimate}; } last_padding_info_ = PaddingInfo(); loss_based_result_.state = LossBasedState::kDecreasing; } else { - // Reset the HOLD duration if delay based estimate works to avoid getting + // Reset the HOLD info if delay based estimate works to avoid getting // stuck in low bitrate. - hold_duration_ = kInitHoldDuration; + last_hold_info_ = {.timestamp = Timestamp::MinusInfinity(), + .duration = kInitHoldDuration, + .rate = DataRate::PlusInfinity()}; last_padding_info_ = PaddingInfo(); loss_based_result_.state = LossBasedState::kDelayBasedEstimate; } loss_based_result_.bandwidth_estimate = bounded_bandwidth_estimate; + + if (IsInLossLimitedState() && + (recovering_after_loss_timestamp_.IsInfinite() || + recovering_after_loss_timestamp_ + config_->delayed_increase_window < + last_send_time_most_recent_observation_)) { + bandwidth_limit_in_current_window_ = + std::max(kCongestionControllerMinBitrate, + current_best_estimate_.loss_limited_bandwidth * + config_->max_increase_factor); + recovering_after_loss_timestamp_ = last_send_time_most_recent_observation_; + } } bool LossBasedBweV2::IsEstimateIncreasingWhenLossLimited( @@ -394,6 +433,10 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig( FieldTrialParameter<bool> enabled("Enabled", true); FieldTrialParameter<double> bandwidth_rampup_upper_bound_factor( "BwRampupUpperBoundFactor", 1000000.0); + FieldTrialParameter<double> bandwidth_rampup_upper_bound_factor_in_hold( + "BwRampupUpperBoundInHoldFactor", 1000000.0); + FieldTrialParameter<double> bandwidth_rampup_hold_threshold( + "BwRampupUpperBoundHoldThreshold", 1.3); FieldTrialParameter<double> rampup_acceleration_max_factor( "BwRampupAccelMaxFactor", 0.0); FieldTrialParameter<TimeDelta> rampup_acceleration_maxout_time( @@ -445,12 +488,6 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig( FieldTrialParameter<bool> not_increase_if_inherent_loss_less_than_average_loss( "NotIncreaseIfInherentLossLessThanAverageLoss", true); - FieldTrialParameter<double> high_loss_rate_threshold("HighLossRateThreshold", - 1.0); - FieldTrialParameter<DataRate> bandwidth_cap_at_high_loss_rate( - "BandwidthCapAtHighLossRate", DataRate::KilobitsPerSec(500.0)); - FieldTrialParameter<double> slope_of_bwe_high_loss_func( - "SlopeOfBweHighLossFunc", 1000); FieldTrialParameter<bool> not_use_acked_rate_in_alr("NotUseAckedRateInAlr", true); FieldTrialParameter<bool> use_in_start_phase("UseInStartPhase", false); @@ -461,9 +498,12 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig( FieldTrialParameter<bool> use_byte_loss_rate("UseByteLossRate", false); FieldTrialParameter<TimeDelta> padding_duration("PaddingDuration", TimeDelta::Zero()); + FieldTrialParameter<bool> bound_best_candidate("BoundBestCandidate", false); if (key_value_config) { ParseFieldTrial({&enabled, &bandwidth_rampup_upper_bound_factor, + &bandwidth_rampup_upper_bound_factor_in_hold, + &bandwidth_rampup_hold_threshold, &rampup_acceleration_max_factor, &rampup_acceleration_maxout_time, &candidate_factors, @@ -491,16 +531,14 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig( &max_increase_factor, &delayed_increase_window, ¬_increase_if_inherent_loss_less_than_average_loss, - &high_loss_rate_threshold, - &bandwidth_cap_at_high_loss_rate, - &slope_of_bwe_high_loss_func, ¬_use_acked_rate_in_alr, &use_in_start_phase, &min_num_observations, &lower_bound_by_acked_rate_factor, &hold_duration_factor, &use_byte_loss_rate, - &padding_duration}, + &padding_duration, + &bound_best_candidate}, key_value_config->Lookup("WebRTC-Bwe-LossBasedBweV2")); } @@ -511,6 +549,10 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig( config.emplace(Config()); config->bandwidth_rampup_upper_bound_factor = bandwidth_rampup_upper_bound_factor.Get(); + config->bandwidth_rampup_upper_bound_factor_in_hold = + bandwidth_rampup_upper_bound_factor_in_hold.Get(); + config->bandwidth_rampup_hold_threshold = + bandwidth_rampup_hold_threshold.Get(); config->rampup_acceleration_max_factor = rampup_acceleration_max_factor.Get(); config->rampup_acceleration_maxout_time = rampup_acceleration_maxout_time.Get(); @@ -553,10 +595,6 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig( config->delayed_increase_window = delayed_increase_window.Get(); config->not_increase_if_inherent_loss_less_than_average_loss = not_increase_if_inherent_loss_less_than_average_loss.Get(); - config->high_loss_rate_threshold = high_loss_rate_threshold.Get(); - config->bandwidth_cap_at_high_loss_rate = - bandwidth_cap_at_high_loss_rate.Get(); - config->slope_of_bwe_high_loss_func = slope_of_bwe_high_loss_func.Get(); config->not_use_acked_rate_in_alr = not_use_acked_rate_in_alr.Get(); config->use_in_start_phase = use_in_start_phase.Get(); config->min_num_observations = min_num_observations.Get(); @@ -565,7 +603,7 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig( config->hold_duration_factor = hold_duration_factor.Get(); config->use_byte_loss_rate = use_byte_loss_rate.Get(); config->padding_duration = padding_duration.Get(); - + config->bound_best_candidate = bound_best_candidate.Get(); return config; } @@ -582,6 +620,18 @@ bool LossBasedBweV2::IsConfigValid() const { << config_->bandwidth_rampup_upper_bound_factor; valid = false; } + if (config_->bandwidth_rampup_upper_bound_factor_in_hold <= 1.0) { + RTC_LOG(LS_WARNING) << "The bandwidth rampup upper bound factor in hold " + "must be greater than 1: " + << config_->bandwidth_rampup_upper_bound_factor_in_hold; + valid = false; + } + if (config_->bandwidth_rampup_hold_threshold < 0.0) { + RTC_LOG(LS_WARNING) << "The bandwidth rampup hold threshold must" + "must be non-negative.: " + << config_->bandwidth_rampup_hold_threshold; + valid = false; + } if (config_->rampup_acceleration_max_factor < 0.0) { RTC_LOG(LS_WARNING) << "The rampup acceleration max factor must be non-negative.: " @@ -739,12 +789,6 @@ bool LossBasedBweV2::IsConfigValid() const { << config_->delayed_increase_window.ms(); valid = false; } - if (config_->high_loss_rate_threshold <= 0.0 || - config_->high_loss_rate_threshold > 1.0) { - RTC_LOG(LS_WARNING) << "The high loss rate threshold must be in (0, 1]: " - << config_->high_loss_rate_threshold; - valid = false; - } if (config_->min_num_observations <= 0) { RTC_LOG(LS_WARNING) << "The min number of observations must be positive: " << config_->min_num_observations; @@ -834,15 +878,19 @@ DataRate LossBasedBweV2::GetCandidateBandwidthUpperBound() const { std::vector<LossBasedBweV2::ChannelParameters> LossBasedBweV2::GetCandidates( bool in_alr) const { + ChannelParameters best_estimate = current_best_estimate_; std::vector<DataRate> bandwidths; for (double candidate_factor : config_->candidate_factors) { bandwidths.push_back(candidate_factor * - current_best_estimate_.loss_limited_bandwidth); + best_estimate.loss_limited_bandwidth); } if (acknowledged_bitrate_.has_value() && config_->append_acknowledged_rate_candidate) { - if (!(config_->not_use_acked_rate_in_alr && in_alr)) { + if (!(config_->not_use_acked_rate_in_alr && in_alr) || + (config_->padding_duration > TimeDelta::Zero() && + last_padding_info_.padding_timestamp + config_->padding_duration >= + last_send_time_most_recent_observation_)) { bandwidths.push_back(*acknowledged_bitrate_ * config_->bandwidth_backoff_lower_bound_factor); } @@ -850,13 +898,13 @@ std::vector<LossBasedBweV2::ChannelParameters> LossBasedBweV2::GetCandidates( if (IsValid(delay_based_estimate_) && config_->append_delay_based_estimate_candidate) { - if (delay_based_estimate_ > current_best_estimate_.loss_limited_bandwidth) { + if (delay_based_estimate_ > best_estimate.loss_limited_bandwidth) { bandwidths.push_back(delay_based_estimate_); } } if (in_alr && config_->append_upper_bound_candidate_in_alr && - current_best_estimate_.loss_limited_bandwidth > GetInstantUpperBound()) { + best_estimate.loss_limited_bandwidth > GetInstantUpperBound()) { bandwidths.push_back(GetInstantUpperBound()); } @@ -866,10 +914,10 @@ std::vector<LossBasedBweV2::ChannelParameters> LossBasedBweV2::GetCandidates( std::vector<ChannelParameters> candidates; candidates.resize(bandwidths.size()); for (size_t i = 0; i < bandwidths.size(); ++i) { - ChannelParameters candidate = current_best_estimate_; - candidate.loss_limited_bandwidth = std::min( - bandwidths[i], std::max(current_best_estimate_.loss_limited_bandwidth, - candidate_bandwidth_upper_bound)); + ChannelParameters candidate = best_estimate; + candidate.loss_limited_bandwidth = + std::min(bandwidths[i], std::max(best_estimate.loss_limited_bandwidth, + candidate_bandwidth_upper_bound)); candidate.inherent_loss = GetFeasibleInherentLoss(candidate); candidates[i] = candidate; } @@ -1037,14 +1085,6 @@ void LossBasedBweV2::CalculateInstantUpperBound() { instant_limit = config_->instant_upper_bound_bandwidth_balance / (average_reported_loss_ratio - config_->instant_upper_bound_loss_offset); - if (average_reported_loss_ratio > config_->high_loss_rate_threshold) { - instant_limit = std::min( - instant_limit, DataRate::KilobitsPerSec(std::max( - static_cast<double>(min_bitrate_.kbps()), - config_->bandwidth_cap_at_high_loss_rate.kbps() - - config_->slope_of_bwe_high_loss_func * - average_reported_loss_ratio))); - } } cached_instant_upper_bound_ = instant_limit; diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h index 425ca2a0c8..9afbb11f1f 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h @@ -62,6 +62,9 @@ class LossBasedBweV2 { // Returns true if loss based BWE is ready to be used in the start phase. bool ReadyToUseInStartPhase() const; + // Returns true if loss based BWE can be used in the start phase. + bool UseInStartPhase() const; + // Returns `DataRate::PlusInfinity` if no BWE can be calculated. Result GetLossBasedResult() const; @@ -83,6 +86,8 @@ class LossBasedBweV2 { struct Config { double bandwidth_rampup_upper_bound_factor = 0.0; + double bandwidth_rampup_upper_bound_factor_in_hold = 0; + double bandwidth_rampup_hold_threshold = 0; double rampup_acceleration_max_factor = 0.0; TimeDelta rampup_acceleration_maxout_time = TimeDelta::Zero(); std::vector<double> candidate_factors; @@ -111,9 +116,6 @@ class LossBasedBweV2 { double max_increase_factor = 0.0; TimeDelta delayed_increase_window = TimeDelta::Zero(); bool not_increase_if_inherent_loss_less_than_average_loss = false; - double high_loss_rate_threshold = 1.0; - DataRate bandwidth_cap_at_high_loss_rate = DataRate::MinusInfinity(); - double slope_of_bwe_high_loss_func = 1000.0; bool not_use_acked_rate_in_alr = false; bool use_in_start_phase = false; int min_num_observations = 0; @@ -121,6 +123,7 @@ class LossBasedBweV2 { double hold_duration_factor = 0.0; bool use_byte_loss_rate = false; TimeDelta padding_duration = TimeDelta::Zero(); + bool bound_best_candidate = false; }; struct Derivatives { @@ -152,6 +155,12 @@ class LossBasedBweV2 { Timestamp padding_timestamp = Timestamp::MinusInfinity(); }; + struct HoldInfo { + Timestamp timestamp = Timestamp::MinusInfinity(); + TimeDelta duration = TimeDelta::Zero(); + DataRate rate = DataRate::PlusInfinity(); + }; + static absl::optional<Config> CreateConfig( const FieldTrialsView* key_value_config); bool IsConfigValid() const; @@ -180,7 +189,6 @@ class LossBasedBweV2 { // Returns false if no observation was created. bool PushBackObservation(rtc::ArrayView<const PacketResult> packet_results); - void UpdateResult(); bool IsEstimateIncreasingWhenLossLimited(DataRate old_estimate, DataRate new_estimate); bool IsInLossLimitedState() const; @@ -204,8 +212,7 @@ class LossBasedBweV2 { DataRate max_bitrate_ = DataRate::PlusInfinity(); DataRate delay_based_estimate_ = DataRate::PlusInfinity(); LossBasedBweV2::Result loss_based_result_ = LossBasedBweV2::Result(); - Timestamp last_hold_timestamp_ = Timestamp::MinusInfinity(); - TimeDelta hold_duration_ = TimeDelta::Zero(); + HoldInfo last_hold_info_ = HoldInfo(); PaddingInfo last_padding_info_ = PaddingInfo(); }; diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build index ca9f20ab87..709bcdb937 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc index 347e2a86d1..9b7ad03148 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc @@ -795,7 +795,7 @@ TEST_F(LossBasedBweV2Test, // The estimate is capped by acked_bitrate * BwRampupUpperBoundFactor. EXPECT_EQ(result.bandwidth_estimate, estimate_1 * 0.9 * 1.2); - // But if acked bitrate decrease, BWE does not decrease when there is no + // But if acked bitrate decreases, BWE does not decrease when there is no // loss. loss_based_bandwidth_estimator.SetAcknowledgedBitrate(estimate_1 * 0.9); loss_based_bandwidth_estimator.UpdateBandwidthEstimate( @@ -809,6 +809,53 @@ TEST_F(LossBasedBweV2Test, result.bandwidth_estimate); } +// Ensure that the state can switch to kIncrease even when the bandwidth is +// bounded by acked bitrate. +TEST_F(LossBasedBweV2Test, EnsureIncreaseEvenIfAckedBitrateBound) { + ExplicitKeyValueConfig key_value_config(ShortObservationConfig( + "LossThresholdOfHighBandwidthPreference:0.99," + "BwRampupUpperBoundFactor:1.2," + // Set InstantUpperBoundBwBalance high to disable InstantUpperBound cap. + "InstantUpperBoundBwBalance:10000kbps,")); + std::vector<PacketResult> enough_feedback_1 = + CreatePacketResultsWith100pLossRate( + /*first_packet_timestamp=*/Timestamp::Zero()); + LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); + DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000); + + loss_based_bandwidth_estimator.SetBandwidthEstimate( + DataRate::KilobitsPerSec(600)); + loss_based_bandwidth_estimator.SetAcknowledgedBitrate( + DataRate::KilobitsPerSec(300)); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate(enough_feedback_1, + delay_based_estimate, + /*in_alr=*/false); + ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, + LossBasedState::kDecreasing); + LossBasedBweV2::Result result = + loss_based_bandwidth_estimator.GetLossBasedResult(); + DataRate estimate_1 = result.bandwidth_estimate; + ASSERT_LT(estimate_1.kbps(), 600); + + // Set a low acked bitrate. + loss_based_bandwidth_estimator.SetAcknowledgedBitrate(estimate_1 / 2); + + int feedback_count = 1; + while (feedback_count < 5 && result.state != LossBasedState::kIncreasing) { + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + feedback_count++ * kObservationDurationLowerBound), + delay_based_estimate, + /*in_alr=*/false); + result = loss_based_bandwidth_estimator.GetLossBasedResult(); + } + + ASSERT_EQ(result.state, LossBasedState::kIncreasing); + // The estimate increases by 1kbps. + EXPECT_EQ(result.bandwidth_estimate, estimate_1 + DataRate::BitsPerSec(1)); +} + // After loss based bwe backs off, the estimate is bounded during the delayed // window. TEST_F(LossBasedBweV2Test, @@ -1007,164 +1054,6 @@ TEST_F(LossBasedBweV2Test, DataRate::KilobitsPerSec(600)); } -TEST_F(LossBasedBweV2Test, - StricterBoundUsingHighLossRateThresholdAt10pLossRate) { - ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HighLossRateThreshold:0.09")); - LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); - loss_based_bandwidth_estimator.SetMinMaxBitrate( - /*min_bitrate=*/DataRate::KilobitsPerSec(10), - /*max_bitrate=*/DataRate::KilobitsPerSec(1000000)); - DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000); - loss_based_bandwidth_estimator.SetBandwidthEstimate( - DataRate::KilobitsPerSec(600)); - - std::vector<PacketResult> enough_feedback_10p_loss_1 = - CreatePacketResultsWith10pPacketLossRate( - /*first_packet_timestamp=*/Timestamp::Zero()); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_10p_loss_1, delay_based_estimate, - - /*in_alr=*/false); - - std::vector<PacketResult> enough_feedback_10p_loss_2 = - CreatePacketResultsWith10pPacketLossRate( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_10p_loss_2, delay_based_estimate, - - /*in_alr=*/false); - - // At 10% loss rate and high loss rate threshold to be 10%, cap the estimate - // to be 500 * 1000-0.1 = 400kbps. - EXPECT_EQ( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - DataRate::KilobitsPerSec(400)); -} - -TEST_F(LossBasedBweV2Test, - StricterBoundUsingHighLossRateThresholdAt50pLossRate) { - ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HighLossRateThreshold:0.3")); - LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); - loss_based_bandwidth_estimator.SetMinMaxBitrate( - /*min_bitrate=*/DataRate::KilobitsPerSec(10), - /*max_bitrate=*/DataRate::KilobitsPerSec(1000000)); - DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000); - loss_based_bandwidth_estimator.SetBandwidthEstimate( - DataRate::KilobitsPerSec(600)); - - std::vector<PacketResult> enough_feedback_50p_loss_1 = - CreatePacketResultsWith50pPacketLossRate( - /*first_packet_timestamp=*/Timestamp::Zero()); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_50p_loss_1, delay_based_estimate, - - /*in_alr=*/false); - - std::vector<PacketResult> enough_feedback_50p_loss_2 = - CreatePacketResultsWith50pPacketLossRate( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_50p_loss_2, delay_based_estimate, - - /*in_alr=*/false); - - // At 50% loss rate and high loss rate threshold to be 30%, cap the estimate - // to be the min bitrate. - EXPECT_EQ( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - DataRate::KilobitsPerSec(10)); -} - -TEST_F(LossBasedBweV2Test, - StricterBoundUsingHighLossRateThresholdAt100pLossRate) { - ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HighLossRateThreshold:0.3")); - LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); - loss_based_bandwidth_estimator.SetMinMaxBitrate( - /*min_bitrate=*/DataRate::KilobitsPerSec(10), - /*max_bitrate=*/DataRate::KilobitsPerSec(1000000)); - DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000); - loss_based_bandwidth_estimator.SetBandwidthEstimate( - DataRate::KilobitsPerSec(600)); - - std::vector<PacketResult> enough_feedback_100p_loss_1 = - CreatePacketResultsWith100pLossRate( - /*first_packet_timestamp=*/Timestamp::Zero()); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_100p_loss_1, delay_based_estimate, - - /*in_alr=*/false); - - std::vector<PacketResult> enough_feedback_100p_loss_2 = - CreatePacketResultsWith100pLossRate( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_100p_loss_2, delay_based_estimate, - - /*in_alr=*/false); - - // At 100% loss rate and high loss rate threshold to be 30%, cap the estimate - // to be the min bitrate. - EXPECT_EQ( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - DataRate::KilobitsPerSec(10)); -} - -TEST_F(LossBasedBweV2Test, EstimateRecoversAfterHighLoss) { - ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HighLossRateThreshold:0.3")); - LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); - loss_based_bandwidth_estimator.SetMinMaxBitrate( - /*min_bitrate=*/DataRate::KilobitsPerSec(10), - /*max_bitrate=*/DataRate::KilobitsPerSec(1000000)); - DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000); - loss_based_bandwidth_estimator.SetBandwidthEstimate( - DataRate::KilobitsPerSec(600)); - - std::vector<PacketResult> enough_feedback_100p_loss_1 = - CreatePacketResultsWith100pLossRate( - /*first_packet_timestamp=*/Timestamp::Zero()); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_100p_loss_1, delay_based_estimate, - - /*in_alr=*/false); - - // Make sure that the estimate is set to min bitrate because of 100% loss - // rate. - EXPECT_EQ( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - DataRate::KilobitsPerSec(10)); - - // Create some feedbacks with 0 loss rate to simulate network recovering. - std::vector<PacketResult> enough_feedback_0p_loss_1 = - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_0p_loss_1, delay_based_estimate, - - /*in_alr=*/false); - - std::vector<PacketResult> enough_feedback_0p_loss_2 = - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound * 2); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - enough_feedback_0p_loss_2, delay_based_estimate, - - /*in_alr=*/false); - - // The estimate increases as network recovers. - EXPECT_GT( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - DataRate::KilobitsPerSec(10)); -} - TEST_F(LossBasedBweV2Test, EstimateIsNotHigherThanMaxBitrate) { ExplicitKeyValueConfig key_value_config( Config(/*enabled=*/true, /*valid=*/true)); @@ -1494,6 +1383,92 @@ TEST_F(LossBasedBweV2Test, IncreaseUsingPaddingStateIfFieldTrial) { LossBasedState::kIncreaseUsingPadding); } +TEST_F(LossBasedBweV2Test, BestCandidateResetsToUpperBoundInFieldTrial) { + ExplicitKeyValueConfig key_value_config( + ShortObservationConfig("PaddingDuration:1000ms,BoundBestCandidate:true")); + LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); + loss_based_bandwidth_estimator.SetBandwidthEstimate( + DataRate::KilobitsPerSec(2500)); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWith50pPacketLossRate( + /*first_packet_timestamp=*/Timestamp::Zero()), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/true); + LossBasedBweV2::Result result_after_loss = + loss_based_bandwidth_estimator.GetLossBasedResult(); + ASSERT_EQ(result_after_loss.state, LossBasedState::kDecreasing); + + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/true); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + 2 * kObservationDurationLowerBound), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/true); + // After a BWE decrease due to large loss, BWE is expected to ramp up slowly + // and follow the acked bitrate. + EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, + LossBasedState::kIncreaseUsingPadding); + EXPECT_NEAR(loss_based_bandwidth_estimator.GetLossBasedResult() + .bandwidth_estimate.kbps(), + result_after_loss.bandwidth_estimate.kbps(), 100); +} + +TEST_F(LossBasedBweV2Test, DecreaseToAckedCandidateIfPaddingInAlr) { + ExplicitKeyValueConfig key_value_config(ShortObservationConfig( + "PaddingDuration:1000ms," + // Set InstantUpperBoundBwBalance high to disable InstantUpperBound cap. + "InstantUpperBoundBwBalance:10000kbps")); + LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); + loss_based_bandwidth_estimator.SetBandwidthEstimate( + DataRate::KilobitsPerSec(1000)); + int feedback_id = 0; + while (loss_based_bandwidth_estimator.GetLossBasedResult().state != + LossBasedState::kDecreasing) { + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWith100pLossRate( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * feedback_id), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/true); + feedback_id++; + } + + while (loss_based_bandwidth_estimator.GetLossBasedResult().state != + LossBasedState::kIncreaseUsingPadding) { + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * feedback_id), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/true); + feedback_id++; + } + ASSERT_GT( + loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, + DataRate::KilobitsPerSec(900)); + + loss_based_bandwidth_estimator.SetAcknowledgedBitrate( + DataRate::KilobitsPerSec(100)); + // Padding is sent now, create some lost packets. + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWith100pLossRate( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * feedback_id), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/true); + EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, + LossBasedState::kDecreasing); + EXPECT_EQ( + loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, + DataRate::KilobitsPerSec(100)); +} + TEST_F(LossBasedBweV2Test, DecreaseAfterPadding) { ExplicitKeyValueConfig key_value_config(ShortObservationConfig( "PaddingDuration:1000ms,BwRampupUpperBoundFactor:2.0")); @@ -1580,7 +1555,7 @@ TEST_F(LossBasedBweV2Test, IncreaseEstimateIfNotHold) { TEST_F(LossBasedBweV2Test, IncreaseEstimateAfterHoldDuration) { ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HoldDurationFactor:3")); + ShortObservationConfig("HoldDurationFactor:10")); LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); loss_based_bandwidth_estimator.SetBandwidthEstimate( DataRate::KilobitsPerSec(2500)); @@ -1629,36 +1604,126 @@ TEST_F(LossBasedBweV2Test, IncreaseEstimateAfterHoldDuration) { /*in_alr=*/false); EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, LossBasedState::kDecreasing); - estimate = + DataRate estimate_at_hold = loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate; - // During the hold duration, e.g. next 900ms, the estimate cannot increase. + // In the hold duration, e.g. next 3s, the estimate cannot increase above the + // hold rate. Get some lost packets to get lower estimate than the HOLD rate. for (int i = 4; i <= 6; ++i) { loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( + CreatePacketResultsWith100pLossRate( /*first_packet_timestamp=*/Timestamp::Zero() + kObservationDurationLowerBound * i), /*delay_based_estimate=*/DataRate::PlusInfinity(), /*in_alr=*/false); EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, LossBasedState::kDecreasing); - EXPECT_EQ( + EXPECT_LT( loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - estimate); + estimate_at_hold); + } + + int feedback_id = 7; + while (loss_based_bandwidth_estimator.GetLossBasedResult().state != + LossBasedState::kIncreasing) { + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * feedback_id), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/false); + if (loss_based_bandwidth_estimator.GetLossBasedResult().state == + LossBasedState::kDecreasing) { + // In the hold duration, the estimate can not go higher than estimate at + // hold. + EXPECT_LE(loss_based_bandwidth_estimator.GetLossBasedResult() + .bandwidth_estimate, + estimate_at_hold); + } else if (loss_based_bandwidth_estimator.GetLossBasedResult().state == + LossBasedState::kIncreasing) { + // After the hold duration, the estimate can increase again. + EXPECT_GT(loss_based_bandwidth_estimator.GetLossBasedResult() + .bandwidth_estimate, + estimate_at_hold); + } + feedback_id++; } +} - // After the hold duration, the estimate can increase again. +TEST_F(LossBasedBweV2Test, HoldRateNotLowerThanAckedRate) { + ExplicitKeyValueConfig key_value_config(ShortObservationConfig( + "HoldDurationFactor:10,LowerBoundByAckedRateFactor:1.0")); + LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); + loss_based_bandwidth_estimator.SetBandwidthEstimate( + DataRate::KilobitsPerSec(2500)); loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( + CreatePacketResultsWith50pPacketLossRate( + /*first_packet_timestamp=*/Timestamp::Zero()), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/false); + ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, + LossBasedState::kDecreasing); + + // During the hold duration, hold rate is not lower than the acked rate. + loss_based_bandwidth_estimator.SetAcknowledgedBitrate( + DataRate::KilobitsPerSec(1000)); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWith50pPacketLossRate( /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound * 7), + kObservationDurationLowerBound), /*delay_based_estimate=*/DataRate::PlusInfinity(), /*in_alr=*/false); EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kIncreasing); - EXPECT_GE( + LossBasedState::kDecreasing); + EXPECT_EQ( loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - estimate); + DataRate::KilobitsPerSec(1000)); +} + +TEST_F(LossBasedBweV2Test, EstimateNotLowerThanAckedRate) { + ExplicitKeyValueConfig key_value_config( + ShortObservationConfig("LowerBoundByAckedRateFactor:1.0")); + LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); + loss_based_bandwidth_estimator.SetBandwidthEstimate( + DataRate::KilobitsPerSec(2500)); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWith100pLossRate( + /*first_packet_timestamp=*/Timestamp::Zero()), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/false); + ASSERT_LT( + loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, + DataRate::KilobitsPerSec(1000)); + + loss_based_bandwidth_estimator.SetAcknowledgedBitrate( + DataRate::KilobitsPerSec(1000)); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWith100pLossRate( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/false); + EXPECT_EQ( + loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, + DataRate::KilobitsPerSec(1000)); + + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * 2), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/false); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * 3), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/false); + + // Verify that the estimate recovers from the acked rate. + EXPECT_GT( + loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, + DataRate::KilobitsPerSec(1000)); } TEST_F(LossBasedBweV2Test, EndHoldDurationIfDelayBasedEstimateWorks) { diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build index 703c22a590..049ac6f477 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build index 291502c95a..6e1d0acff5 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc index b09cb22f49..22693d67e9 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc @@ -203,9 +203,9 @@ TimeDelta RttBasedBackoff::CorrectedRtt() const { RttBasedBackoff::~RttBasedBackoff() = default; SendSideBandwidthEstimation::SendSideBandwidthEstimation( - const FieldTrialsView* key_value_config, - RtcEventLog* event_log) - : rtt_backoff_(key_value_config), + const FieldTrialsView* key_value_config, RtcEventLog* event_log) + : key_value_config_(key_value_config), + rtt_backoff_(key_value_config), lost_packets_since_last_loss_update_(0), expected_packets_since_last_loss_update_(0), current_target_(DataRate::Zero()), @@ -234,7 +234,7 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation( high_loss_threshold_(kDefaultHighLossThreshold), bitrate_threshold_(kDefaultBitrateThreshold), loss_based_bandwidth_estimator_v1_(key_value_config), - loss_based_bandwidth_estimator_v2_(key_value_config), + loss_based_bandwidth_estimator_v2_(new LossBasedBweV2(key_value_config)), loss_based_state_(LossBasedState::kDelayBasedEstimate), disable_receiver_limit_caps_only_("Disabled") { RTC_DCHECK(event_log); @@ -252,7 +252,7 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation( ParseFieldTrial({&disable_receiver_limit_caps_only_}, key_value_config->Lookup("WebRTC-Bwe-ReceiverLimitCapsOnly")); if (LossBasedBandwidthEstimatorV2Enabled()) { - loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate( + loss_based_bandwidth_estimator_v2_->SetMinMaxBitrate( min_bitrate_configured_, max_bitrate_configured_); } } @@ -281,6 +281,10 @@ void SendSideBandwidthEstimation::OnRouteChange() { uma_update_state_ = kNoUpdate; uma_rtt_state_ = kNoUpdate; last_rtc_event_log_ = Timestamp::MinusInfinity(); + if (loss_based_bandwidth_estimator_v2_->UseInStartPhase()) { + loss_based_bandwidth_estimator_v2_.reset( + new LossBasedBweV2(key_value_config_)); + } } void SendSideBandwidthEstimation::SetBitrates( @@ -315,8 +319,8 @@ void SendSideBandwidthEstimation::SetMinMaxBitrate(DataRate min_bitrate, } else { max_bitrate_configured_ = kDefaultMaxBitrate; } - loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate(min_bitrate_configured_, - max_bitrate_configured_); + loss_based_bandwidth_estimator_v2_->SetMinMaxBitrate(min_bitrate_configured_, + max_bitrate_configured_); } int SendSideBandwidthEstimation::GetMinBitrate() const { @@ -371,7 +375,7 @@ void SendSideBandwidthEstimation::SetAcknowledgedRate( *acknowledged_rate, at_time); } if (LossBasedBandwidthEstimatorV2Enabled()) { - loss_based_bandwidth_estimator_v2_.SetAcknowledgedBitrate( + loss_based_bandwidth_estimator_v2_->SetAcknowledgedBitrate( *acknowledged_rate); } } @@ -386,7 +390,7 @@ void SendSideBandwidthEstimation::UpdateLossBasedEstimator( report.packet_feedbacks, report.feedback_time); } if (LossBasedBandwidthEstimatorV2Enabled()) { - loss_based_bandwidth_estimator_v2_.UpdateBandwidthEstimate( + loss_based_bandwidth_estimator_v2_->UpdateBandwidthEstimate( report.packet_feedbacks, delay_based_limit_, in_alr); UpdateEstimate(report.feedback_time); } @@ -492,7 +496,7 @@ void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) { // We trust the REMB and/or delay-based estimate during the first 2 seconds if // we haven't had any packet loss reported, to allow startup bitrate probing. if (last_fraction_loss_ == 0 && IsInStartPhase(at_time) && - !loss_based_bandwidth_estimator_v2_.ReadyToUseInStartPhase()) { + !loss_based_bandwidth_estimator_v2_->ReadyToUseInStartPhase()) { DataRate new_bitrate = current_target_; // TODO(srte): We should not allow the new_bitrate to be larger than the // receiver limit here. @@ -534,7 +538,7 @@ void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) { if (LossBasedBandwidthEstimatorV2ReadyForUse()) { LossBasedBweV2::Result result = - loss_based_bandwidth_estimator_v2_.GetLossBasedResult(); + loss_based_bandwidth_estimator_v2_->GetLossBasedResult(); loss_based_state_ = result.state; UpdateTargetBitrate(result.bandwidth_estimate, at_time); return; @@ -690,13 +694,13 @@ bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1ReadyForUse() } bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2Enabled() const { - return loss_based_bandwidth_estimator_v2_.IsEnabled(); + return loss_based_bandwidth_estimator_v2_->IsEnabled(); } bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2ReadyForUse() const { return LossBasedBandwidthEstimatorV2Enabled() && - loss_based_bandwidth_estimator_v2_.IsReady(); + loss_based_bandwidth_estimator_v2_->IsReady(); } } // namespace webrtc diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h index 3a4efc47c7..dd4d25a236 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h @@ -16,6 +16,7 @@ #include <stdint.h> #include <deque> +#include <memory> #include <utility> #include <vector> @@ -167,6 +168,7 @@ class SendSideBandwidthEstimation { bool LossBasedBandwidthEstimatorV1ReadyForUse() const; bool LossBasedBandwidthEstimatorV2ReadyForUse() const; + const FieldTrialsView* key_value_config_; RttBasedBackoff rtt_backoff_; LinkCapacityTracker link_capacity_; @@ -208,7 +210,7 @@ class SendSideBandwidthEstimation { float high_loss_threshold_; DataRate bitrate_threshold_; LossBasedBandwidthEstimation loss_based_bandwidth_estimator_v1_; - LossBasedBweV2 loss_based_bandwidth_estimator_v2_; + std::unique_ptr<LossBasedBweV2> loss_based_bandwidth_estimator_v2_; LossBasedState loss_based_state_; FieldTrialFlag disable_receiver_limit_caps_only_; }; diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build index d83d51f985..08cfdec69b 100644 --- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build index 7e8cb87820..62800e263d 100644 --- a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build index 40ead5619c..41f64326b2 100644 --- a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build +++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc index 40764de7ae..81caa9bd2d 100644 --- a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc +++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc @@ -112,6 +112,7 @@ void BaseCapturerPipeWire::OnScreenCastSessionClosed() { if (!capturer_failed_) { options_.screencast_stream()->StopScreenCastStream(); } + capturer_failed_ = true; } void BaseCapturerPipeWire::UpdateResolution(uint32_t width, uint32_t height) { diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc index 61c6957d27..473f913466 100644 --- a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc +++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc @@ -14,7 +14,6 @@ #include <libdrm/drm_fourcc.h> #include <pipewire/pipewire.h> #include <spa/param/video/format-utils.h> -#include <sys/mman.h> #include <vector> @@ -49,33 +48,6 @@ constexpr int CursorMetaSize(int w, int h) { constexpr PipeWireVersion kDmaBufModifierMinVersion = {0, 3, 33}; constexpr PipeWireVersion kDropSingleModifierMinVersion = {0, 3, 40}; -class ScopedBuf { - public: - ScopedBuf() {} - ScopedBuf(uint8_t* map, int map_size, int fd) - : map_(map), map_size_(map_size), fd_(fd) {} - ~ScopedBuf() { - if (map_ != MAP_FAILED) { - munmap(map_, map_size_); - } - } - - explicit operator bool() { return map_ != MAP_FAILED; } - - void initialize(uint8_t* map, int map_size, int fd) { - map_ = map; - map_size_ = map_size; - fd_ = fd; - } - - uint8_t* get() { return map_; } - - protected: - uint8_t* map_ = static_cast<uint8_t*>(MAP_FAILED); - int map_size_; - int fd_; -}; - class SharedScreenCastStreamPrivate { public: SharedScreenCastStreamPrivate(); diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.h b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.h index aad28d2f30..64ef5750ec 100644 --- a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.h +++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.h @@ -46,6 +46,8 @@ class DesktopFrameProvider { // Expected to be called before stopping the CGDisplayStreamRef streams. void Release(); + bool allow_iosurface() const { return allow_iosurface_; } + private: SequenceChecker thread_checker_; const bool allow_iosurface_; diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm b/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm index 1f4a62f7cd..785a15dfa4 100644 --- a/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm +++ b/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm @@ -442,6 +442,10 @@ void ScreenCapturerMac::ScreenConfigurationChanged() { bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() { RTC_DCHECK(thread_checker_.IsCurrent()); + if (!desktop_frame_provider_.allow_iosurface()) { + return true; + } + desktop_config_ = desktop_config_monitor_->desktop_configuration(); for (const auto& config : desktop_config_.displays) { size_t pixel_width = config.pixel_bounds.width(); diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.h index 2b1e0ab041..815986f680 100644 --- a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.h +++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.h @@ -132,7 +132,7 @@ class RTC_EXPORT DxgiDuplicatorController { // scoped_refptr<DxgiDuplicatorController> accesses private AddRef() and // Release() functions. - friend class rtc::scoped_refptr<DxgiDuplicatorController>; + friend class webrtc::scoped_refptr<DxgiDuplicatorController>; // A private constructor to ensure consumers to use // DxgiDuplicatorController::Instance(). diff --git a/third_party/libwebrtc/modules/module_api_gn/moz.build b/third_party/libwebrtc/modules/module_api_gn/moz.build index d61cca4a48..7613736af2 100644 --- a/third_party/libwebrtc/modules/module_api_gn/moz.build +++ b/third_party/libwebrtc/modules/module_api_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/module_api_public_gn/moz.build b/third_party/libwebrtc/modules/module_api_public_gn/moz.build index 45518d1a8a..c40e3cf5e9 100644 --- a/third_party/libwebrtc/modules/module_api_public_gn/moz.build +++ b/third_party/libwebrtc/modules/module_api_public_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/module_fec_api_gn/moz.build b/third_party/libwebrtc/modules/module_fec_api_gn/moz.build index 7b4274f1b8..86a280e5cc 100644 --- a/third_party/libwebrtc/modules/module_fec_api_gn/moz.build +++ b/third_party/libwebrtc/modules/module_fec_api_gn/moz.build @@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/pacing/interval_budget_gn/moz.build b/third_party/libwebrtc/modules/pacing/interval_budget_gn/moz.build index a528123ae0..8bb44ecf62 100644 --- a/third_party/libwebrtc/modules/pacing/interval_budget_gn/moz.build +++ b/third_party/libwebrtc/modules/pacing/interval_budget_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.cc b/third_party/libwebrtc/modules/pacing/pacing_controller.cc index 13ff9a2a95..5b81207d56 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_controller.cc +++ b/third_party/libwebrtc/modules/pacing/pacing_controller.cc @@ -73,7 +73,7 @@ PacingController::PacingController(Clock* clock, keyframe_flushing_( IsEnabled(field_trials_, "WebRTC-Pacer-KeyframeFlushing")), transport_overhead_per_packet_(DataSize::Zero()), - send_burst_interval_(TimeDelta::Zero()), + send_burst_interval_(kDefaultBurstInterval), last_timestamp_(clock_->CurrentTime()), paused_(false), media_debt_(DataSize::Zero()), diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.h b/third_party/libwebrtc/modules/pacing/pacing_controller.h index dd5636ccef..04e0a820f9 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_controller.h +++ b/third_party/libwebrtc/modules/pacing/pacing_controller.h @@ -25,6 +25,7 @@ #include "api/transport/field_trial_based_config.h" #include "api/transport/network_types.h" #include "api/units/data_size.h" +#include "api/units/time_delta.h" #include "modules/pacing/bitrate_prober.h" #include "modules/pacing/interval_budget.h" #include "modules/pacing/prioritized_packet_queue.h" @@ -92,6 +93,10 @@ class PacingController { // the send burst interval. // Ex: max send burst interval = 63Kb / 10Mbit/s = 50ms. static constexpr DataSize kMaxBurstSize = DataSize::Bytes(63 * 1000); + // The pacer is allowed to send enqued packets in bursts and can build up a + // packet "debt" that correspond to approximately the send rate during + // the burst interval. + static constexpr TimeDelta kDefaultBurstInterval = TimeDelta::Millis(40); PacingController(Clock* clock, PacketSender* packet_sender, diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc index ba93d05bb7..9e6ede6dc0 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc +++ b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc @@ -427,6 +427,7 @@ TEST_F(PacingControllerTest, BudgetAffectsAudioInTrial) { DataRate pacing_rate = DataRate::BitsPerSec(kPacketSize / 3 * 8 * kProcessIntervalsPerSecond); pacer.SetPacingRates(pacing_rate, DataRate::Zero()); + pacer.SetSendBurstInterval(TimeDelta::Zero()); // Video fills budget for following process periods. pacer.EnqueuePacket(video_.BuildNextPacket(kPacketSize)); EXPECT_CALL(callback_, SendPacket).Times(1); @@ -484,7 +485,7 @@ TEST_F(PacingControllerTest, FirstSentPacketTimeIsSet) { EXPECT_EQ(kStartTime, pacer->FirstSentPacketTime()); } -TEST_F(PacingControllerTest, QueueAndPacePackets) { +TEST_F(PacingControllerTest, QueueAndPacePacketsWithZeroBurstPeriod) { const uint32_t kSsrc = 12345; uint16_t sequence_number = 1234; const DataSize kPackeSize = DataSize::Bytes(250); @@ -495,6 +496,7 @@ TEST_F(PacingControllerTest, QueueAndPacePackets) { const size_t kPacketsToSend = (kSendInterval * kTargetRate).bytes() * kPaceMultiplier / kPackeSize.bytes(); auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_); + pacer->SetSendBurstInterval(TimeDelta::Zero()); pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero()); for (size_t i = 0; i < kPacketsToSend; ++i) { @@ -536,30 +538,30 @@ TEST_F(PacingControllerTest, PaceQueuedPackets) { auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_); pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero()); - // Due to the multiplicative factor we can send 5 packets during a send - // interval. (network capacity * multiplier / (8 bits per byte * - // (packet size * #send intervals per second) - const size_t packets_to_send_per_interval = - kTargetRate.bps() * kPaceMultiplier / (8 * kPacketSize * 200); - for (size_t i = 0; i < packets_to_send_per_interval; ++i) { + const size_t packets_to_send_per_burst_interval = + (kTargetRate * kPaceMultiplier * PacingController::kDefaultBurstInterval) + .bytes() / + kPacketSize; + for (size_t i = 0; i < packets_to_send_per_burst_interval; ++i) { SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc, sequence_number++, clock_.TimeInMilliseconds(), kPacketSize); } - for (size_t j = 0; j < packets_to_send_per_interval * 10; ++j) { + for (size_t j = 0; j < packets_to_send_per_burst_interval * 10; ++j) { pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc, sequence_number++, clock_.TimeInMilliseconds(), kPacketSize)); } - EXPECT_EQ(packets_to_send_per_interval + packets_to_send_per_interval * 10, + EXPECT_EQ(packets_to_send_per_burst_interval + + packets_to_send_per_burst_interval * 10, pacer->QueueSizePackets()); - while (pacer->QueueSizePackets() > packets_to_send_per_interval * 10) { + while (pacer->QueueSizePackets() > packets_to_send_per_burst_interval * 10) { AdvanceTimeUntil(pacer->NextSendTime()); pacer->ProcessPackets(); } - EXPECT_EQ(pacer->QueueSizePackets(), packets_to_send_per_interval * 10); + EXPECT_EQ(pacer->QueueSizePackets(), packets_to_send_per_burst_interval * 10); EXPECT_CALL(callback_, SendPadding).Times(0); EXPECT_CALL(callback_, SendPacket(ssrc, _, _, false, false)) @@ -582,12 +584,12 @@ TEST_F(PacingControllerTest, PaceQueuedPackets) { pacer->ProcessPackets(); // Send some more packet, just show that we can..? - for (size_t i = 0; i < packets_to_send_per_interval; ++i) { + for (size_t i = 0; i < packets_to_send_per_burst_interval; ++i) { SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc, sequence_number++, clock_.TimeInMilliseconds(), 250); } - EXPECT_EQ(packets_to_send_per_interval, pacer->QueueSizePackets()); - for (size_t i = 0; i < packets_to_send_per_interval; ++i) { + EXPECT_EQ(packets_to_send_per_burst_interval, pacer->QueueSizePackets()); + for (size_t i = 0; i < packets_to_send_per_burst_interval; ++i) { AdvanceTimeUntil(pacer->NextSendTime()); pacer->ProcessPackets(); } @@ -641,19 +643,23 @@ TEST_F(PacingControllerTest, TEST_F(PacingControllerTest, Padding) { uint32_t ssrc = 12345; uint16_t sequence_number = 1234; - const size_t kPacketSize = 250; + const size_t kPacketSize = 1000; auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_); pacer->SetPacingRates(kTargetRate * kPaceMultiplier, kTargetRate); - const size_t kPacketsToSend = 20; + const size_t kPacketsToSend = 30; for (size_t i = 0; i < kPacketsToSend; ++i) { SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc, sequence_number++, clock_.TimeInMilliseconds(), kPacketSize); } + + int expected_bursts = + floor(DataSize::Bytes(pacer->QueueSizePackets() * kPacketSize) / + (kPaceMultiplier * kTargetRate) / + PacingController::kDefaultBurstInterval); const TimeDelta expected_pace_time = - DataSize::Bytes(pacer->QueueSizePackets() * kPacketSize) / - (kPaceMultiplier * kTargetRate); + (expected_bursts - 1) * PacingController::kDefaultBurstInterval; EXPECT_CALL(callback_, SendPadding).Times(0); // Only the media packets should be sent. Timestamp start_time = clock_.CurrentTime(); @@ -663,7 +669,7 @@ TEST_F(PacingControllerTest, Padding) { } const TimeDelta actual_pace_time = clock_.CurrentTime() - start_time; EXPECT_LE((actual_pace_time - expected_pace_time).Abs(), - PacingController::kMinSleepTime); + PacingController::kDefaultBurstInterval); // Pacing media happens at 2.5x, but padding was configured with 1.0x // factor. We have to wait until the padding debt is gone before we start @@ -766,8 +772,8 @@ TEST_F(PacingControllerTest, VerifyAverageBitrateVaryingMediaPayload) { media_payload)); media_bytes += media_payload; } - - AdvanceTimeUntil(pacer->NextSendTime()); + AdvanceTimeUntil(std::min(clock_.CurrentTime() + TimeDelta::Millis(20), + pacer->NextSendTime())); pacer->ProcessPackets(); } @@ -805,20 +811,18 @@ TEST_F(PacingControllerTest, Priority) { // Expect all high and normal priority to be sent out first. EXPECT_CALL(callback_, SendPadding).Times(0); + testing::Sequence s; EXPECT_CALL(callback_, SendPacket(ssrc, _, capture_time_ms, _, _)) - .Times(packets_to_send_per_interval + 1); + .Times(packets_to_send_per_interval + 1) + .InSequence(s); + EXPECT_CALL(callback_, SendPacket(ssrc_low_priority, _, + capture_time_ms_low_priority, _, _)) + .InSequence(s); - while (pacer->QueueSizePackets() > 1) { + while (pacer->QueueSizePackets() > 0) { AdvanceTimeUntil(pacer->NextSendTime()); pacer->ProcessPackets(); } - - EXPECT_EQ(1u, pacer->QueueSizePackets()); - - EXPECT_CALL(callback_, SendPacket(ssrc_low_priority, _, - capture_time_ms_low_priority, _, _)); - AdvanceTimeUntil(pacer->NextSendTime()); - pacer->ProcessPackets(); } TEST_F(PacingControllerTest, RetransmissionPriority) { @@ -829,23 +833,22 @@ TEST_F(PacingControllerTest, RetransmissionPriority) { auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_); pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero()); - // Due to the multiplicative factor we can send 5 packets during a send - // interval. (network capacity * multiplier / (8 bits per byte * - // (packet size * #send intervals per second) - const size_t packets_to_send_per_interval = - kTargetRate.bps() * kPaceMultiplier / (8 * 250 * 200); + const size_t packets_to_send_per_burst_interval = + (kTargetRate * kPaceMultiplier * PacingController::kDefaultBurstInterval) + .bytes() / + 250; pacer->ProcessPackets(); EXPECT_EQ(0u, pacer->QueueSizePackets()); // Alternate retransmissions and normal packets. - for (size_t i = 0; i < packets_to_send_per_interval; ++i) { + for (size_t i = 0; i < packets_to_send_per_burst_interval; ++i) { pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc, sequence_number++, capture_time_ms, 250)); pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission, ssrc, sequence_number++, capture_time_ms_retransmission, 250)); } - EXPECT_EQ(2 * packets_to_send_per_interval, pacer->QueueSizePackets()); + EXPECT_EQ(2 * packets_to_send_per_burst_interval, pacer->QueueSizePackets()); // Expect all retransmissions to be sent out first despite having a later // capture time. @@ -853,19 +856,19 @@ TEST_F(PacingControllerTest, RetransmissionPriority) { EXPECT_CALL(callback_, SendPacket(_, _, _, false, _)).Times(0); EXPECT_CALL(callback_, SendPacket(ssrc, _, capture_time_ms_retransmission, true, _)) - .Times(packets_to_send_per_interval); + .Times(packets_to_send_per_burst_interval); - while (pacer->QueueSizePackets() > packets_to_send_per_interval) { + while (pacer->QueueSizePackets() > packets_to_send_per_burst_interval) { AdvanceTimeUntil(pacer->NextSendTime()); pacer->ProcessPackets(); } - EXPECT_EQ(packets_to_send_per_interval, pacer->QueueSizePackets()); + EXPECT_EQ(packets_to_send_per_burst_interval, pacer->QueueSizePackets()); // Expect the remaining (non-retransmission) packets to be sent. EXPECT_CALL(callback_, SendPadding).Times(0); EXPECT_CALL(callback_, SendPacket(_, _, _, true, _)).Times(0); EXPECT_CALL(callback_, SendPacket(ssrc, _, capture_time_ms, false, _)) - .Times(packets_to_send_per_interval); + .Times(packets_to_send_per_burst_interval); while (pacer->QueueSizePackets() > 0) { AdvanceTimeUntil(pacer->NextSendTime()); @@ -890,13 +893,13 @@ TEST_F(PacingControllerTest, HighPrioDoesntAffectBudget) { sequence_number++, capture_time_ms, kPacketSize); } pacer->ProcessPackets(); + EXPECT_EQ(pacer->QueueSizePackets(), 0u); // Low prio packets does affect the budget. - // Due to the multiplicative factor we can send 5 packets during a send - // interval. (network capacity * multiplier / (8 bits per byte * - // (packet size * #send intervals per second) - const size_t kPacketsToSendPerInterval = - kTargetRate.bps() * kPaceMultiplier / (8 * kPacketSize * 200); - for (size_t i = 0; i < kPacketsToSendPerInterval; ++i) { + const size_t kPacketsToSendPerBurstInterval = + (kTargetRate * kPaceMultiplier * PacingController::kDefaultBurstInterval) + .bytes() / + kPacketSize; + for (size_t i = 0; i < kPacketsToSendPerBurstInterval; ++i) { SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc, sequence_number++, clock_.TimeInMilliseconds(), kPacketSize); @@ -904,16 +907,16 @@ TEST_F(PacingControllerTest, HighPrioDoesntAffectBudget) { // Send all packets and measure pace time. Timestamp start_time = clock_.CurrentTime(); + EXPECT_EQ(pacer->NextSendTime(), clock_.CurrentTime()); while (pacer->QueueSizePackets() > 0) { AdvanceTimeUntil(pacer->NextSendTime()); pacer->ProcessPackets(); } - // Measure pacing time. Expect only low-prio packets to affect this. + // Measure pacing time. TimeDelta pacing_time = clock_.CurrentTime() - start_time; - TimeDelta expected_pacing_time = - DataSize::Bytes(kPacketsToSendPerInterval * kPacketSize) / - (kTargetRate * kPaceMultiplier); + // All packets sent in one burst since audio packets are not accounted for. + TimeDelta expected_pacing_time = TimeDelta::Zero(); EXPECT_NEAR(pacing_time.us<double>(), expected_pacing_time.us<double>(), PacingController::kMinSleepTime.us<double>()); } @@ -965,6 +968,7 @@ TEST_F(PacingControllerTest, DoesNotAllowOveruseAfterCongestion) { auto now_ms = [this] { return clock_.TimeInMilliseconds(); }; auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_); pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero()); + pacer->SetSendBurstInterval(TimeDelta::Zero()); EXPECT_CALL(callback_, SendPadding).Times(0); // The pacing rate is low enough that the budget should not allow two packets // to be sent in a row. @@ -1853,6 +1857,7 @@ TEST_F(PacingControllerTest, AccountsForAudioEnqueueTime) { // Audio not paced, but still accounted for in budget. pacer->SetAccountForAudioPackets(true); pacer->SetPacingRates(kPacingDataRate, kPaddingDataRate); + pacer->SetSendBurstInterval(TimeDelta::Zero()); // Enqueue two audio packets, advance clock to where one packet // should have drained the buffer already, has they been sent @@ -1898,13 +1903,12 @@ TEST_F(PacingControllerTest, NextSendTimeAccountsForPadding) { EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), PacingController::kPausedProcessInterval); - // Enqueue a new packet, that can't be sent until previous buffer has - // drained. + // Enqueue a new packet, that can be sent immediately due to default burst + // rate is 40ms. SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc, sequnce_number++, clock_.TimeInMilliseconds(), kPacketSize.bytes()); - EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), kPacketPacingTime); - clock_.AdvanceTime(kPacketPacingTime); + EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), TimeDelta::Zero()); pacer->ProcessPackets(); ::testing::Mock::VerifyAndClearExpectations(&callback_); @@ -1916,11 +1920,13 @@ TEST_F(PacingControllerTest, NextSendTimeAccountsForPadding) { // previous debt has cleared. Since padding was disabled before, there // currently is no padding debt. pacer->SetPacingRates(kPacingDataRate, kPacingDataRate / 2); - EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), kPacketPacingTime); + EXPECT_EQ(pacer->QueueSizePackets(), 0u); + EXPECT_LT(pacer->NextSendTime() - clock_.CurrentTime(), + PacingController::kDefaultBurstInterval); // Advance time, expect padding. EXPECT_CALL(callback_, SendPadding).WillOnce(Return(kPacketSize.bytes())); - clock_.AdvanceTime(kPacketPacingTime); + clock_.AdvanceTime(pacer->NextSendTime() - clock_.CurrentTime()); pacer->ProcessPackets(); ::testing::Mock::VerifyAndClearExpectations(&callback_); @@ -1933,7 +1939,7 @@ TEST_F(PacingControllerTest, NextSendTimeAccountsForPadding) { pacer->EnqueuePacket( BuildPacket(RtpPacketMediaType::kVideo, kSsrc, sequnce_number++, clock_.TimeInMilliseconds(), kPacketSize.bytes())); - EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), kPacketPacingTime); + EXPECT_EQ(pacer->NextSendTime(), clock_.CurrentTime()); } TEST_F(PacingControllerTest, PaddingTargetAccountsForPaddingRate) { @@ -2011,8 +2017,8 @@ TEST_F(PacingControllerTest, SendsFecPackets) { TEST_F(PacingControllerTest, GapInPacingDoesntAccumulateBudget) { const uint32_t kSsrc = 12345; uint16_t sequence_number = 1234; - const DataSize kPackeSize = DataSize::Bytes(250); - const TimeDelta kPacketSendTime = TimeDelta::Millis(15); + const DataSize kPackeSize = DataSize::Bytes(1000); + const TimeDelta kPacketSendTime = TimeDelta::Millis(25); auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_); pacer->SetPacingRates(kPackeSize / kPacketSendTime, @@ -2028,15 +2034,20 @@ TEST_F(PacingControllerTest, GapInPacingDoesntAccumulateBudget) { // Advance time kPacketSendTime past where the media debt should be 0. clock_.AdvanceTime(2 * kPacketSendTime); - // Enqueue two new packets. Expect only one to be sent one ProcessPackets(). + // Enqueue three new packets. Expect only two to be sent one ProcessPackets() + // since the default burst interval is 40ms. + SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc, + sequence_number++, clock_.TimeInMilliseconds(), + kPackeSize.bytes()); + SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc, + sequence_number++, clock_.TimeInMilliseconds(), + kPackeSize.bytes()); + EXPECT_CALL(callback_, SendPacket(kSsrc, sequence_number + 1, _, _, _)) + .Times(0); pacer->EnqueuePacket( BuildPacket(RtpPacketMediaType::kVideo, kSsrc, sequence_number + 1, clock_.TimeInMilliseconds(), kPackeSize.bytes())); - pacer->EnqueuePacket( - BuildPacket(RtpPacketMediaType::kVideo, kSsrc, sequence_number + 2, - clock_.TimeInMilliseconds(), kPackeSize.bytes())); - EXPECT_CALL(callback_, SendPacket(kSsrc, sequence_number + 1, - clock_.TimeInMilliseconds(), false, false)); + pacer->ProcessPackets(); } @@ -2044,6 +2055,7 @@ TEST_F(PacingControllerTest, HandlesSubMicrosecondSendIntervals) { static constexpr DataSize kPacketSize = DataSize::Bytes(1); static constexpr TimeDelta kPacketSendTime = TimeDelta::Micros(1); auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_); + pacer->SetSendBurstInterval(TimeDelta::Zero()); // Set pacing rate such that a packet is sent in 0.5us. pacer->SetPacingRates(/*pacing_rate=*/2 * kPacketSize / kPacketSendTime, diff --git a/third_party/libwebrtc/modules/pacing/pacing_gn/moz.build b/third_party/libwebrtc/modules/pacing/pacing_gn/moz.build index 6b7f69865f..353f876c55 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_gn/moz.build +++ b/third_party/libwebrtc/modules/pacing/pacing_gn/moz.build @@ -207,7 +207,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -217,10 +216,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc index afa36ea88d..f7218e48a1 100644 --- a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc +++ b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc @@ -17,35 +17,19 @@ #include "api/task_queue/pending_task_safety_flag.h" #include "api/transport/network_types.h" #include "rtc_base/checks.h" -#include "rtc_base/experiments/field_trial_parser.h" -#include "rtc_base/experiments/field_trial_units.h" #include "rtc_base/trace_event.h" namespace webrtc { -namespace { - -constexpr const char* kBurstyPacerFieldTrial = "WebRTC-BurstyPacer"; - -} // namespace - const int TaskQueuePacedSender::kNoPacketHoldback = -1; -TaskQueuePacedSender::BurstyPacerFlags::BurstyPacerFlags( - const FieldTrialsView& field_trials) - : burst("burst") { - ParseFieldTrial({&burst}, field_trials.Lookup(kBurstyPacerFieldTrial)); -} - TaskQueuePacedSender::TaskQueuePacedSender( Clock* clock, PacingController::PacketSender* packet_sender, const FieldTrialsView& field_trials, TimeDelta max_hold_back_window, - int max_hold_back_window_in_packets, - absl::optional<TimeDelta> burst_interval) + int max_hold_back_window_in_packets) : clock_(clock), - bursty_pacer_flags_(field_trials), max_hold_back_window_(max_hold_back_window), max_hold_back_window_in_packets_(max_hold_back_window_in_packets), pacing_controller_(clock, packet_sender, field_trials), @@ -56,17 +40,6 @@ TaskQueuePacedSender::TaskQueuePacedSender( include_overhead_(false), task_queue_(TaskQueueBase::Current()) { RTC_DCHECK_GE(max_hold_back_window_, PacingController::kMinSleepTime); - // There are multiple field trials that can affect burst. If multiple bursts - // are specified we pick the largest of the values. - absl::optional<TimeDelta> burst = bursty_pacer_flags_.burst.GetOptional(); - // If not overriden by an experiment, the burst is specified by the - // `burst_interval` argument. - if (!burst.has_value()) { - burst = burst_interval; - } - if (burst.has_value()) { - pacing_controller_.SetSendBurstInterval(burst.value()); - } } TaskQueuePacedSender::~TaskQueuePacedSender() { @@ -74,6 +47,11 @@ TaskQueuePacedSender::~TaskQueuePacedSender() { is_shutdown_ = true; } +void TaskQueuePacedSender::SetSendBurstInterval(TimeDelta burst_interval) { + RTC_DCHECK_RUN_ON(task_queue_); + pacing_controller_.SetSendBurstInterval(burst_interval); +} + void TaskQueuePacedSender::EnsureStarted() { RTC_DCHECK_RUN_ON(task_queue_); is_started_ = true; diff --git a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.h b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.h index fd71be1654..e29acdf878 100644 --- a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.h +++ b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.h @@ -45,23 +45,21 @@ class TaskQueuePacedSender : public RtpPacketPacer, public RtpPacketSender { // processed. Increasing this reduces thread wakeups at the expense of higher // latency. // - // If the `burst_interval` parameter is set, the pacer is allowed to build up - // a packet "debt" that correspond to approximately the send rate during the - // specified interval. This greatly reduced wake ups by not pacing packets - // within the allowed burst budget. - // // The taskqueue used when constructing a TaskQueuePacedSender will also be // used for pacing. - TaskQueuePacedSender( - Clock* clock, - PacingController::PacketSender* packet_sender, - const FieldTrialsView& field_trials, - TimeDelta max_hold_back_window, - int max_hold_back_window_in_packets, - absl::optional<TimeDelta> burst_interval = absl::nullopt); + TaskQueuePacedSender(Clock* clock, + PacingController::PacketSender* packet_sender, + const FieldTrialsView& field_trials, + TimeDelta max_hold_back_window, + int max_hold_back_window_in_packets); ~TaskQueuePacedSender() override; + // The pacer is allowed to send enqued packets in bursts and can build up a + // packet "debt" that correspond to approximately the send rate during + // 'burst_interval'. + void SetSendBurstInterval(TimeDelta burst_interval); + // Ensure that necessary delayed tasks are scheduled. void EnsureStarted(); @@ -145,15 +143,6 @@ class TaskQueuePacedSender : public RtpPacketPacer, public RtpPacketSender { Stats GetStats() const; Clock* const clock_; - struct BurstyPacerFlags { - // Parses `kBurstyPacerFieldTrial`. Example: - // --force-fieldtrials=WebRTC-BurstyPacer/burst:20ms/ - explicit BurstyPacerFlags(const FieldTrialsView& field_trials); - // If set, the pacer is allowed to build up a packet "debt" that correspond - // to approximately the send rate during the specified interval. - FieldTrialOptional<TimeDelta> burst; - }; - const BurstyPacerFlags bursty_pacer_flags_; // The holdback window prevents too frequent delayed MaybeProcessPackets() // calls. These are only applicable if `allow_low_precision` is false. diff --git a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender_unittest.cc b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender_unittest.cc index 54347493e7..f0a9ad78c2 100644 --- a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender_unittest.cc +++ b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender_unittest.cc @@ -11,6 +11,7 @@ #include "modules/pacing/task_queue_paced_sender.h" #include <algorithm> +#include <any> #include <atomic> #include <list> #include <memory> @@ -24,6 +25,7 @@ #include "api/units/data_rate.h" #include "api/units/data_size.h" #include "api/units/time_delta.h" +#include "modules/pacing/pacing_controller.h" #include "modules/pacing/packet_router.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "test/gmock.h" @@ -33,6 +35,9 @@ using ::testing::_; using ::testing::AtLeast; +using ::testing::AtMost; +using ::testing::Lt; +using ::testing::NiceMock; using ::testing::Return; using ::testing::SaveArg; @@ -167,9 +172,10 @@ TEST(TaskQueuePacedSenderTest, PacesPacketsWithBurst) { TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials, PacingController::kMinSleepTime, - TaskQueuePacedSender::kNoPacketHoldback, - // Half a second of bursting. - TimeDelta::Seconds(0.5)); + TaskQueuePacedSender::kNoPacketHoldback); + pacer.SetSendBurstInterval( + // Half a second of bursting. + TimeDelta::Seconds(0.5)); // Insert a number of packets, covering one second. static constexpr size_t kPacketsToSend = 42; @@ -262,7 +268,7 @@ TEST(TaskQueuePacedSenderTest, ReschedulesProcessOnRateChange) { TEST(TaskQueuePacedSenderTest, SendsAudioImmediately) { GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234)); - MockPacketRouter packet_router; + NiceMock<MockPacketRouter> packet_router; ScopedKeyValueConfig trials; TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials, @@ -270,21 +276,16 @@ TEST(TaskQueuePacedSenderTest, SendsAudioImmediately) { TaskQueuePacedSender::kNoPacketHoldback); const DataRate kPacingDataRate = DataRate::KilobitsPerSec(125); - const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize); - const TimeDelta kPacketPacingTime = kPacketSize / kPacingDataRate; pacer.SetPacingRates(kPacingDataRate, DataRate::Zero()); pacer.EnsureStarted(); - // Add some initial video packets, only one should be sent. - EXPECT_CALL(packet_router, SendPacket); + // Add some initial video packets. Not all should be sent immediately. + EXPECT_CALL(packet_router, SendPacket).Times(AtMost(9)); pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 10)); time_controller.AdvanceTime(TimeDelta::Zero()); ::testing::Mock::VerifyAndClearExpectations(&packet_router); - // Advance time, but still before next packet should be sent. - time_controller.AdvanceTime(kPacketPacingTime / 2); - // Insert an audio packet, it should be sent immediately. EXPECT_CALL(packet_router, SendPacket); pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kAudio, 1)); @@ -295,12 +296,13 @@ TEST(TaskQueuePacedSenderTest, SendsAudioImmediately) { TEST(TaskQueuePacedSenderTest, SleepsDuringCoalscingWindow) { const TimeDelta kCoalescingWindow = TimeDelta::Millis(5); GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234)); - MockPacketRouter packet_router; + NiceMock<MockPacketRouter> packet_router; ScopedKeyValueConfig trials; TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials, kCoalescingWindow, TaskQueuePacedSender::kNoPacketHoldback); + pacer.SetSendBurstInterval(TimeDelta::Zero()); // Set rates so one packet adds one ms of buffer level. const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize); @@ -310,9 +312,9 @@ TEST(TaskQueuePacedSenderTest, SleepsDuringCoalscingWindow) { pacer.SetPacingRates(kPacingDataRate, DataRate::Zero()); pacer.EnsureStarted(); - // Add 10 packets. The first should be sent immediately since the buffers - // are clear. - EXPECT_CALL(packet_router, SendPacket); + // Add 10 packets. The first burst should be sent immediately since the + // buffers are clear. + EXPECT_CALL(packet_router, SendPacket).Times(AtMost(9)); pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 10)); time_controller.AdvanceTime(TimeDelta::Zero()); ::testing::Mock::VerifyAndClearExpectations(&packet_router); @@ -370,11 +372,12 @@ TEST(TaskQueuePacedSenderTest, SchedulesProbeAtSentTime) { ScopedKeyValueConfig trials( "WebRTC-Bwe-ProbingBehavior/min_probe_delta:1ms/"); GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234)); - MockPacketRouter packet_router; + NiceMock<MockPacketRouter> packet_router; TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials, PacingController::kMinSleepTime, TaskQueuePacedSender::kNoPacketHoldback); + pacer.SetSendBurstInterval(TimeDelta::Zero()); // Set rates so one packet adds 4ms of buffer level. const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize); @@ -504,11 +507,12 @@ TEST(TaskQueuePacedSenderTest, PacketBasedCoalescing) { const int kPacketBasedHoldback = 5; GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234)); - MockPacketRouter packet_router; + NiceMock<MockPacketRouter> packet_router; ScopedKeyValueConfig trials; TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials, kFixedCoalescingWindow, kPacketBasedHoldback); + pacer.SetSendBurstInterval(TimeDelta::Zero()); // Set rates so one packet adds one ms of buffer level. const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize); @@ -559,6 +563,7 @@ TEST(TaskQueuePacedSenderTest, FixedHoldBackHasPriorityOverPackets) { TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials, kFixedCoalescingWindow, kPacketBasedHoldback); + pacer.SetSendBurstInterval(TimeDelta::Zero()); // Set rates so one packet adds one ms of buffer level. const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize); @@ -691,7 +696,7 @@ TEST(TaskQueuePacedSenderTest, PostedPacketsNotSendFromRemovePacketsForSsrc) { TEST(TaskQueuePacedSenderTest, Stats) { static constexpr Timestamp kStartTime = Timestamp::Millis(1234); GlobalSimulatedTimeController time_controller(kStartTime); - MockPacketRouter packet_router; + NiceMock<MockPacketRouter> packet_router; ScopedKeyValueConfig trials; TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials, @@ -708,7 +713,8 @@ TEST(TaskQueuePacedSenderTest, Stats) { // Allowed `QueueSizeData` and `ExpectedQueueTime` deviation. static constexpr size_t kAllowedPacketsDeviation = 1; static constexpr DataSize kAllowedQueueSizeDeviation = - DataSize::Bytes(kDefaultPacketSize * kAllowedPacketsDeviation); + DataSize::Bytes(kDefaultPacketSize * kAllowedPacketsDeviation) + + kPacingRate * PacingController::kDefaultBurstInterval; static constexpr TimeDelta kAllowedQueueTimeDeviation = kAllowedQueueSizeDeviation / kPacingRate; diff --git a/third_party/libwebrtc/modules/portal/pipewire_utils.h b/third_party/libwebrtc/modules/portal/pipewire_utils.h index 8344a8cefb..c1327b85c9 100644 --- a/third_party/libwebrtc/modules/portal/pipewire_utils.h +++ b/third_party/libwebrtc/modules/portal/pipewire_utils.h @@ -11,6 +11,21 @@ #ifndef MODULES_PORTAL_PIPEWIRE_UTILS_H_ #define MODULES_PORTAL_PIPEWIRE_UTILS_H_ +#include <errno.h> +#include <stdint.h> +#include <sys/ioctl.h> +#include <sys/mman.h> + +// static +struct dma_buf_sync { + uint64_t flags; +}; +#define DMA_BUF_SYNC_READ (1 << 0) +#define DMA_BUF_SYNC_START (0 << 2) +#define DMA_BUF_SYNC_END (1 << 2) +#define DMA_BUF_BASE 'b' +#define DMA_BUF_IOCTL_SYNC _IOW(DMA_BUF_BASE, 0, struct dma_buf_sync) + struct pw_thread_loop; namespace webrtc { @@ -32,6 +47,66 @@ class PipeWireThreadLoopLock { pw_thread_loop* const loop_; }; +// We should synchronize DMA Buffer object access from CPU to avoid potential +// cache incoherency and data loss. +// See +// https://01.org/linuxgraphics/gfx-docs/drm/driver-api/dma-buf.html#cpu-access-to-dma-buffer-objects +static bool SyncDmaBuf(int fd, uint64_t start_or_end) { + struct dma_buf_sync sync = {0}; + + sync.flags = start_or_end | DMA_BUF_SYNC_READ; + + while (true) { + int ret; + ret = ioctl(fd, DMA_BUF_IOCTL_SYNC, &sync); + if (ret == -1 && errno == EINTR) { + continue; + } else if (ret == -1) { + return false; + } else { + break; + } + } + + return true; +} + +class ScopedBuf { + public: + ScopedBuf() {} + ScopedBuf(uint8_t* map, int map_size, int fd, bool is_dma_buf = false) + : map_(map), map_size_(map_size), fd_(fd), is_dma_buf_(is_dma_buf) {} + ~ScopedBuf() { + if (map_ != MAP_FAILED) { + if (is_dma_buf_) { + SyncDmaBuf(fd_, DMA_BUF_SYNC_END); + } + munmap(map_, map_size_); + } + } + + explicit operator bool() { return map_ != MAP_FAILED; } + + void initialize(uint8_t* map, int map_size, int fd, bool is_dma_buf = false) { + map_ = map; + map_size_ = map_size; + is_dma_buf_ = is_dma_buf; + fd_ = fd; + + if (is_dma_buf_) { + SyncDmaBuf(fd_, DMA_BUF_SYNC_START); + } + } + + uint8_t* get() { return map_; } + + protected: + uint8_t* map_ = static_cast<uint8_t*>(MAP_FAILED); + int map_size_; + int fd_; + bool is_dma_buf_; +}; + } // namespace webrtc #endif // MODULES_PORTAL_PIPEWIRE_UTILS_H_ diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.h index 97fa490adf..c9edc4f551 100644 --- a/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.h +++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.h @@ -108,7 +108,7 @@ class AimdRateControl { // If "Disabled", estimated link capacity is not used as upper bound. FieldTrialFlag disable_estimate_bounded_increase_{"Disabled"}; FieldTrialParameter<bool> use_current_estimate_as_min_upper_bound_{"c_upper", - false}; + true}; absl::optional<DataRate> last_decrease_; }; } // namespace webrtc diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc index f26afe995c..401e87e310 100644 --- a/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc +++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc @@ -208,6 +208,7 @@ TEST(AimdRateControlTest, SetEstimateIncreaseBweInAlr) { TEST(AimdRateControlTest, SetEstimateUpperLimitedByNetworkEstimate) { AimdRateControl aimd_rate_control(ExplicitKeyValueConfig(""), /*send_side=*/true); + aimd_rate_control.SetEstimate(DataRate::BitsPerSec(300'000), kInitialTime); NetworkStateEstimate network_estimate; network_estimate.link_capacity_upper = DataRate::BitsPerSec(400'000); aimd_rate_control.SetNetworkStateEstimate(network_estimate); @@ -217,11 +218,9 @@ TEST(AimdRateControlTest, SetEstimateUpperLimitedByNetworkEstimate) { } TEST(AimdRateControlTest, - SetEstimateUpperLimitedByCurrentBitrateIfNetworkEstimateIsLow) { - AimdRateControl aimd_rate_control( - ExplicitKeyValueConfig( - "WebRTC-Bwe-EstimateBoundedIncrease/c_upper:true/"), - /*send_side=*/true); + SetEstimateDefaultUpperLimitedByCurrentBitrateIfNetworkEstimateIsLow) { + AimdRateControl aimd_rate_control(ExplicitKeyValueConfig(""), + /*send_side=*/true); aimd_rate_control.SetEstimate(DataRate::BitsPerSec(500'000), kInitialTime); ASSERT_EQ(aimd_rate_control.LatestEstimate(), DataRate::BitsPerSec(500'000)); @@ -233,9 +232,12 @@ TEST(AimdRateControlTest, } TEST(AimdRateControlTest, - SetEstimateDefaultNotUpperLimitedByCurrentBitrateIfNetworkEstimateIsLow) { - AimdRateControl aimd_rate_control(ExplicitKeyValueConfig(""), - /*send_side=*/true); + SetEstimateNotUpperLimitedByCurrentBitrateIfNetworkEstimateIsLowIf) { + AimdRateControl aimd_rate_control( + ExplicitKeyValueConfig( + "WebRTC-Bwe-EstimateBoundedIncrease/c_upper:false/"), + /*send_side=*/true); + aimd_rate_control.SetEstimate(DataRate::BitsPerSec(500'000), kInitialTime); ASSERT_EQ(aimd_rate_control.LatestEstimate(), DataRate::BitsPerSec(500'000)); diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_gn/moz.build b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_gn/moz.build index 2876755e91..45104d15ca 100644 --- a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_gn/moz.build +++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_gn/moz.build @@ -211,7 +211,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -221,10 +220,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn b/third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn index 0fc9931f39..b471c2fa76 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn +++ b/third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn @@ -258,6 +258,13 @@ rtc_library("rtp_rtcp") { "source/video_rtp_depacketizer_vp9.h", ] + if (rtc_use_h265) { + sources += [ + "source/rtp_packetizer_h265.cc", + "source/rtp_packetizer_h265.h", + ] + } + if (rtc_enable_bwe_test_logging) { defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ] } else { @@ -624,6 +631,10 @@ if (rtc_include_tests) { "source/video_rtp_depacketizer_vp8_unittest.cc", "source/video_rtp_depacketizer_vp9_unittest.cc", ] + if (rtc_use_h265) { + sources += [ "source/rtp_packetizer_h265_unittest.cc" ] + } + deps = [ ":fec_test_helper", ":frame_transformer_factory_unittest", diff --git a/third_party/libwebrtc/modules/rtp_rtcp/leb128_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/leb128_gn/moz.build index 88f2cb22e0..e42ea18507 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/leb128_gn/moz.build +++ b/third_party/libwebrtc/modules/rtp_rtcp/leb128_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_format_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_format_gn/moz.build index da304ae5a4..33d8799fb2 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_format_gn/moz.build +++ b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_format_gn/moz.build @@ -241,7 +241,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -251,10 +250,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_gn/moz.build index 382194837b..8c49736436 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_gn/moz.build +++ b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_gn/moz.build @@ -255,7 +255,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -265,10 +264,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/rtp_rtcp/rtp_video_header_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/rtp_video_header_gn/moz.build index 2c8b5e2321..d2a102cfe3 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/rtp_video_header_gn/moz.build +++ b/third_party/libwebrtc/modules/rtp_rtcp/rtp_video_header_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc index cfca7cb066..3e6d04d59c 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc @@ -138,9 +138,9 @@ bool FlexfecHeaderReader::ReadFecHeader( mask_part0 <<= 1; ByteWriter<uint16_t>::WriteBigEndian(&data[byte_index], mask_part0); byte_index += kFlexfecPacketMaskSizes[0]; - if (k_bit0) { - // The first K-bit is set, and the packet mask is thus only 2 bytes long. - // We have finished reading the properties for current ssrc. + if (!k_bit0) { + // The first K-bit is clear, and the packet mask is thus only 2 bytes + // long. We have finished reading the properties for current ssrc. fec_packet->protected_streams[i].packet_mask_size = kFlexfecPacketMaskSizes[0]; } else { @@ -162,8 +162,8 @@ bool FlexfecHeaderReader::ReadFecHeader( mask_part1 <<= 2; ByteWriter<uint32_t>::WriteBigEndian(&data[byte_index], mask_part1); byte_index += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0]; - if (k_bit1) { - // The first K-bit is clear, but the second K-bit is set. The packet + if (!k_bit1) { + // The first K-bit is set, but the second K-bit is clear. The packet // mask is thus 6 bytes long. We have finished reading the properties // for current ssrc. fec_packet->protected_streams[i].packet_mask_size = @@ -273,8 +273,9 @@ void FlexfecHeaderWriter::FinalizeFecHeader( tmp_mask_part0 >>= 1; // Shift, thus clearing K-bit 0. ByteWriter<uint16_t>::WriteBigEndian(write_at, tmp_mask_part0); + *write_at |= 0x80; // Set K-bit 0. write_at += kFlexfecPacketMaskSizes[0]; - tmp_mask_part1 >>= 2; // Shift, thus clearing K-bit 1 and bit 15. + tmp_mask_part1 >>= 2; // Shift twice, thus clearing K-bit 1 and bit 15. ByteWriter<uint32_t>::WriteBigEndian(write_at, tmp_mask_part1); bool bit15 = (protected_stream.packet_mask[1] & 0x01) != 0; @@ -284,9 +285,9 @@ void FlexfecHeaderWriter::FinalizeFecHeader( bool bit46 = (protected_stream.packet_mask[5] & 0x02) != 0; bool bit47 = (protected_stream.packet_mask[5] & 0x01) != 0; if (!bit46 && !bit47) { - *write_at |= 0x80; // Set K-bit 1. write_at += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0]; } else { + *write_at |= 0x80; // Set K-bit 1. write_at += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0]; // Clear all trailing bits. memset(write_at, 0, @@ -307,14 +308,13 @@ void FlexfecHeaderWriter::FinalizeFecHeader( ByteWriter<uint16_t>::WriteBigEndian(write_at, tmp_mask_part0); bool bit15 = (protected_stream.packet_mask[1] & 0x01) != 0; if (!bit15) { - *write_at |= 0x80; // Set K-bit 0. write_at += kFlexfecPacketMaskSizes[0]; } else { + *write_at |= 0x80; // Set K-bit 0. write_at += kFlexfecPacketMaskSizes[0]; // Clear all trailing bits. memset(write_at, 0U, kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0]); - *write_at |= 0x80; // Set K-bit 1. *write_at |= 0x40; // Set bit 15. write_at += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0]; } diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc index 6995ba3871..f25e0d8d2a 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc @@ -36,11 +36,12 @@ using ReceivedFecPacket = ForwardErrorCorrection::ReceivedFecPacket; using ::testing::Each; using ::testing::ElementsAreArray; -constexpr uint8_t kMask0[] = {0xAB, 0xCD}; // First K bit is set. -constexpr uint8_t kMask1[] = {0x12, 0x34, // First K bit cleared. - 0xF6, 0x78, 0x9A, 0xBC}; // Second K bit set. -constexpr uint8_t kMask2[] = {0x12, 0x34, // First K bit cleared. - 0x56, 0x78, 0x9A, 0xBC, // Second K bit cleared. +constexpr uint8_t kKBit = 1 << 7; +constexpr uint8_t kMask0[] = {0x2B, 0xCD}; // First K bit is cleared. +constexpr uint8_t kMask1[] = {0x92, 0x34, // First K bit set. + 0x76, 0x78, 0x9A, 0xBC}; // Second K bit cleared. +constexpr uint8_t kMask2[] = {0x92, 0x34, // First K bit set. + 0xD6, 0x78, 0x9A, 0xBC, // Second K bit set. 0xDE, 0xF0, 0x12, 0x34, 0x56, 0x78, 0x9A, 0xBC}; constexpr size_t kMediaPacketLength = 1234; @@ -186,11 +187,10 @@ void VerifyWrittenAndReadHeaders( } // namespace -TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit0SetSingleStream) { - constexpr uint8_t kKBit0 = 1 << 7; +TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit0ClearSingleStream) { constexpr size_t kExpectedFecHeaderSize = 12; constexpr uint16_t kSnBase = 0x0102; - constexpr uint8_t kFlexfecPktMask[] = {kKBit0 | 0x08, 0x81}; + constexpr uint8_t kFlexfecPktMask[] = {0x08, 0x81}; constexpr uint8_t kUlpfecPacketMask[] = {0x11, 0x02}; constexpr uint8_t kPacketData[] = { kFlexible, kPtRecovery, kLengthRecovery[0], kLengthRecovery[1], @@ -215,13 +215,11 @@ TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit0SetSingleStream) { VerifyReadHeaders(kExpectedFecHeaderSize, read_packet, expected); } -TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit1SetSingleStream) { - constexpr uint8_t kKBit0 = 0 << 7; - constexpr uint8_t kKBit1 = 1 << 7; +TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit1ClearSingleStream) { constexpr size_t kExpectedFecHeaderSize = 16; constexpr uint16_t kSnBase = 0x0102; - constexpr uint8_t kFlexfecPktMask[] = {kKBit0 | 0x48, 0x81, // - kKBit1 | 0x02, 0x11, 0x00, 0x21}; + constexpr uint8_t kFlexfecPktMask[] = {kKBit | 0x48, 0x81, // + 0x02, 0x11, 0x00, 0x21}; constexpr uint8_t kUlpfecPacketMask[] = {0x91, 0x02, // 0x08, 0x44, 0x00, 0x84}; constexpr uint8_t kPacketData[] = { @@ -250,15 +248,13 @@ TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit1SetSingleStream) { VerifyReadHeaders(kExpectedFecHeaderSize, read_packet, expected); } -TEST(FlexfecHeaderReaderTest, ReadsHeaderWithNoKBitsSetSingleStream) { - constexpr uint8_t kKBit0 = 0 << 7; - constexpr uint8_t kKBit1 = 0 << 7; +TEST(FlexfecHeaderReaderTest, ReadsHeaderWithBothKBitsSetSingleStream) { constexpr size_t kExpectedFecHeaderSize = 24; constexpr uint16_t kSnBase = 0x0102; - constexpr uint8_t kFlexfecPacketMask[] = {kKBit0 | 0x48, 0x81, // - kKBit1 | 0x02, 0x11, 0x00, 0x21, // - 0x01, 0x11, 0x11, 0x11, - 0x11, 0x11, 0x11, 0x11}; + constexpr uint8_t kFlexfecPacketMask[] = {kKBit | 0x48, 0x81, // + kKBit | 0x02, 0x11, 0x00, 0x21, // + 0x01, 0x11, 0x11, 0x11, + 0x11, 0x11, 0x11, 0x11}; constexpr uint8_t kUlpfecPacketMask[] = {0x91, 0x02, // 0x08, 0x44, 0x00, 0x84, // 0x04, 0x44, 0x44, 0x44, @@ -309,14 +305,13 @@ TEST(FlexfecHeaderReaderTest, ReadsHeaderWithNoKBitsSetSingleStream) { VerifyReadHeaders(kExpectedFecHeaderSize, read_packet, expected); } -TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit0Set2Streams) { - constexpr uint8_t kKBit0 = 1 << 7; +TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit0Clear2Streams) { constexpr size_t kExpectedFecHeaderSize = 16; constexpr uint16_t kSnBase0 = 0x0102; constexpr uint16_t kSnBase1 = 0x0304; - constexpr uint8_t kFlexfecPktMask1[] = {kKBit0 | 0x08, 0x81}; + constexpr uint8_t kFlexfecPktMask1[] = {0x08, 0x81}; constexpr uint8_t kUlpfecPacketMask1[] = {0x11, 0x02}; - constexpr uint8_t kFlexfecPktMask2[] = {kKBit0 | 0x04, 0x41}; + constexpr uint8_t kFlexfecPktMask2[] = {0x04, 0x41}; constexpr uint8_t kUlpfecPacketMask2[] = {0x08, 0x82}; constexpr uint8_t kPacketData[] = { @@ -349,18 +344,16 @@ TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit0Set2Streams) { VerifyReadHeaders(kExpectedFecHeaderSize, read_packet, expected); } -TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit1Set2Streams) { - constexpr uint8_t kKBit0 = 0 << 7; - constexpr uint8_t kKBit1 = 1 << 7; +TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit1Clear2Streams) { constexpr size_t kExpectedFecHeaderSize = 24; constexpr uint16_t kSnBase0 = 0x0102; constexpr uint16_t kSnBase1 = 0x0304; - constexpr uint8_t kFlexfecPktMask1[] = {kKBit0 | 0x48, 0x81, // - kKBit1 | 0x02, 0x11, 0x00, 0x21}; + constexpr uint8_t kFlexfecPktMask1[] = {kKBit | 0x48, 0x81, // + 0x02, 0x11, 0x00, 0x21}; constexpr uint8_t kUlpfecPacketMask1[] = {0x91, 0x02, // 0x08, 0x44, 0x00, 0x84}; - constexpr uint8_t kFlexfecPktMask2[] = {kKBit0 | 0x57, 0x82, // - kKBit1 | 0x04, 0x33, 0x00, 0x51}; + constexpr uint8_t kFlexfecPktMask2[] = {kKBit | 0x57, 0x82, // + 0x04, 0x33, 0x00, 0x51}; constexpr uint8_t kUlpfecPacketMask2[] = {0xAF, 0x04, // 0x10, 0xCC, 0x01, 0x44}; constexpr uint8_t kPacketData[] = { @@ -398,24 +391,22 @@ TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit1Set2Streams) { VerifyReadHeaders(kExpectedFecHeaderSize, read_packet, expected); } -TEST(FlexfecHeaderReaderTest, ReadsHeaderWithNoKBitsSet2Streams) { - constexpr uint8_t kKBit0 = 0 << 7; - constexpr uint8_t kKBit1 = 0 << 7; +TEST(FlexfecHeaderReaderTest, ReadsHeaderWithBothKBitsSet2Streams) { constexpr size_t kExpectedFecHeaderSize = 40; constexpr uint16_t kSnBase0 = 0x0102; constexpr uint16_t kSnBase1 = 0x0304; - constexpr uint8_t kFlexfecPktMask1[] = {kKBit0 | 0x48, 0x81, // - kKBit1 | 0x02, 0x11, 0x00, 0x21, // - 0x01, 0x11, 0x11, 0x11, - 0x11, 0x11, 0x11, 0x11}; + constexpr uint8_t kFlexfecPktMask1[] = {kKBit | 0x48, 0x81, // + kKBit | 0x02, 0x11, 0x00, 0x21, // + 0x01, 0x11, 0x11, 0x11, + 0x11, 0x11, 0x11, 0x11}; constexpr uint8_t kUlpfecPacketMask1[] = {0x91, 0x02, // 0x08, 0x44, 0x00, 0x84, // 0x04, 0x44, 0x44, 0x44, 0x44, 0x44, 0x44, 0x44}; - constexpr uint8_t kFlexfecPktMask2[] = {kKBit0 | 0x32, 0x84, // - kKBit1 | 0x05, 0x23, 0x00, 0x55, // - 0xA3, 0x22, 0x22, 0x22, - 0x22, 0x22, 0x22, 0x35}; + constexpr uint8_t kFlexfecPktMask2[] = {kKBit | 0x32, 0x84, // + kKBit | 0x05, 0x23, 0x00, 0x55, // + 0xA3, 0x22, 0x22, 0x22, + 0x22, 0x22, 0x22, 0x35}; constexpr uint8_t kUlpfecPacketMask2[] = {0x65, 0x08, // 0x14, 0x8C, 0x01, 0x56, // 0x8C, 0x88, 0x88, 0x88, @@ -490,29 +481,27 @@ TEST(FlexfecHeaderReaderTest, ReadsHeaderWithNoKBitsSet2Streams) { } TEST(FlexfecHeaderReaderTest, ReadsHeaderWithMultipleStreamsMultipleMasks) { - constexpr uint8_t kBit0 = 0 << 7; - constexpr uint8_t kBit1 = 1 << 7; constexpr size_t kExpectedFecHeaderSize = 44; constexpr uint16_t kSnBase0 = 0x0102; constexpr uint16_t kSnBase1 = 0x0304; constexpr uint16_t kSnBase2 = 0x0506; constexpr uint16_t kSnBase3 = 0x0708; - constexpr uint8_t kFlexfecPacketMask1[] = {kBit1 | 0x29, 0x91}; + constexpr uint8_t kFlexfecPacketMask1[] = {0x29, 0x91}; constexpr uint8_t kUlpfecPacketMask1[] = {0x53, 0x22}; - constexpr uint8_t kFlexfecPacketMask2[] = {kBit0 | 0x32, 0xA1, // - kBit1 | 0x02, 0x11, 0x00, 0x21}; + constexpr uint8_t kFlexfecPacketMask2[] = {kKBit | 0x32, 0xA1, // + 0x02, 0x11, 0x00, 0x21}; constexpr uint8_t kUlpfecPacketMask2[] = {0x65, 0x42, // 0x08, 0x44, 0x00, 0x84}; - constexpr uint8_t kFlexfecPacketMask3[] = {kBit0 | 0x48, 0x81, // - kBit0 | 0x02, 0x11, 0x00, 0x21, // + constexpr uint8_t kFlexfecPacketMask3[] = {kKBit | 0x48, 0x81, // + kKBit | 0x02, 0x11, 0x00, 0x21, // 0x01, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11}; constexpr uint8_t kUlpfecPacketMask3[] = {0x91, 0x02, // 0x08, 0x44, 0x00, 0x84, // 0x04, 0x44, 0x44, 0x44, 0x44, 0x44, 0x44, 0x44}; - constexpr uint8_t kFlexfecPacketMask4[] = {kBit0 | 0x32, 0x84, // - kBit1 | 0x05, 0x23, 0x00, 0x55}; + constexpr uint8_t kFlexfecPacketMask4[] = {kKBit | 0x32, 0x84, // + 0x05, 0x23, 0x00, 0x55}; constexpr uint8_t kUlpfecPacketMask4[] = {0x65, 0x08, // 0x14, 0x8C, 0x01, 0x54}; constexpr uint8_t kPacketData[] = {kFlexible, @@ -642,7 +631,7 @@ TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit0SetShouldFail) { EXPECT_FALSE(reader.ReadFecHeader(&read_packet)); } -TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit1SetShouldFail) { +TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit1ClearShouldFail) { // Simulate short received packet. constexpr uint8_t kPacketData[] = { kFlexible, kPtRecovery, kLengthRecovery[0], kLengthRecovery[1], @@ -659,7 +648,7 @@ TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit1SetShouldFail) { EXPECT_FALSE(reader.ReadFecHeader(&read_packet)); } -TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit1ClearedShouldFail) { +TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit1SetShouldFail) { // Simulate short received packet. constexpr uint8_t kPacketData[] = { kFlexible, kPtRecovery, kLengthRecovery[0], kLengthRecovery[1], @@ -698,8 +687,8 @@ TEST(FlexfecHeaderReaderTest, ReadShortPacketMultipleStreamsShouldFail) { EXPECT_FALSE(reader.ReadFecHeader(&read_packet)); } -TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit0SetSingleStream) { - constexpr uint8_t kFlexfecPacketMask[] = {0x88, 0x81}; +TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit0ClearSingleStream) { + constexpr uint8_t kFlexfecPacketMask[] = {0x08, 0x81}; constexpr uint8_t kUlpfecPacketMask[] = {0x11, 0x02}; constexpr uint16_t kMediaStartSeqNum = 1234; Packet written_packet = WritePacket({{.ssrc = 0x01, @@ -714,8 +703,8 @@ TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit0SetSingleStream) { VerifyFinalizedHeaders(written_packet, expected); } -TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit1SetSingleStream) { - constexpr uint8_t kFlexfecPacketMask[] = {0x48, 0x81, 0x82, 0x11, 0x00, 0x21}; +TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit1ClearSingleStream) { + constexpr uint8_t kFlexfecPacketMask[] = {0xC8, 0x81, 0x02, 0x11, 0x00, 0x21}; constexpr uint8_t kUlpfecPacketMask[] = {0x91, 0x02, 0x08, 0x44, 0x00, 0x84}; constexpr uint16_t kMediaStartSeqNum = 1234; Packet written_packet = WritePacket({{.ssrc = 0x01, @@ -730,10 +719,10 @@ TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit1SetSingleStream) { VerifyFinalizedHeaders(written_packet, expected); } -TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithNoKBitsSetSingleStream) { +TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithBothKBitsSetSingleStream) { constexpr uint8_t kFlexfecPacketMask[] = { - 0x11, 0x11, // K-bit 0 clear. - 0x11, 0x11, 0x11, 0x10, // K-bit 1 clear. + 0x91, 0x11, // K-bit 0 set. + 0x91, 0x11, 0x11, 0x10, // K-bit 1 set. 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 // }; constexpr uint8_t kUlpfecPacketMask[] = {0x22, 0x22, 0x44, 0x44, 0x44, 0x41}; @@ -752,22 +741,22 @@ TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithNoKBitsSetSingleStream) { TEST(FlexfecHeaderWriterTest, FinalizesHeaderMultipleStreamsMultipleMasks) { constexpr uint8_t kFlexfecPacketMask1[] = { - 0x11, 0x11, // K-bit 0 clear. - 0x11, 0x11, 0x11, 0x10, // K-bit 1 clear. + 0x91, 0x11, // K-bit 0 set. + 0x91, 0x11, 0x11, 0x10, // K-bit 1 set. 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 // }; constexpr uint8_t kUlpfecPacketMask1[] = {0x22, 0x22, 0x44, 0x44, 0x44, 0x41}; constexpr uint16_t kMediaStartSeqNum1 = 1234; - constexpr uint8_t kFlexfecPacketMask2[] = {0x88, 0x81}; + constexpr uint8_t kFlexfecPacketMask2[] = {0x08, 0x81}; constexpr uint8_t kUlpfecPacketMask2[] = {0x11, 0x02}; constexpr uint16_t kMediaStartSeqNum2 = 2345; - constexpr uint8_t kFlexfecPacketMask3[] = {0x48, 0x81, 0x82, + constexpr uint8_t kFlexfecPacketMask3[] = {0xC8, 0x81, 0x02, 0x11, 0x00, 0x21}; constexpr uint8_t kUlpfecPacketMask3[] = {0x91, 0x02, 0x08, 0x44, 0x00, 0x84}; constexpr uint16_t kMediaStartSeqNum3 = 3456; constexpr uint8_t kFlexfecPacketMask4[] = { - 0x55, 0xAA, // K-bit 0 clear. - 0x22, 0xAB, 0xCD, 0xEF, // K-bit 1 clear. + 0xD5, 0xAA, // K-bit 0 set. + 0xA2, 0xAB, 0xCD, 0xEF, // K-bit 1 set. 0xC0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 // }; constexpr uint8_t kUlpfecPacketMask4[] = {0xAB, 0x54, 0x8A, 0xAF, 0x37, 0xBF}; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc index 2c11a29bfa..c7534dee40 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc @@ -22,6 +22,9 @@ #include "modules/video_coding/codecs/vp8/include/vp8_globals.h" #include "modules/video_coding/codecs/vp9/include/vp9_globals.h" #include "rtc_base/checks.h" +#ifdef RTC_ENABLE_H265 +#include "modules/rtp_rtcp/source/rtp_packetizer_h265.h" +#endif namespace webrtc { @@ -57,7 +60,11 @@ std::unique_ptr<RtpPacketizer> RtpPacketizer::Create( return std::make_unique<RtpPacketizerAv1>( payload, limits, rtp_video_header.frame_type, rtp_video_header.is_last_frame_in_picture); - // TODO(bugs.webrtc.org/13485): Implement RtpPacketizerH265. +#ifdef RTC_ENABLE_H265 + case kVideoCodecH265: { + return std::make_unique<RtpPacketizerH265>(payload, limits); + } +#endif default: { return std::make_unique<RtpPacketizerGeneric>(payload, limits, rtp_video_header); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc index 95dbaf364c..859b529a47 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc @@ -74,8 +74,7 @@ RtpPacketizerAv1::RtpPacketizerAv1(rtc::ArrayView<const uint8_t> payload, std::vector<RtpPacketizerAv1::Obu> RtpPacketizerAv1::ParseObus( rtc::ArrayView<const uint8_t> payload) { std::vector<Obu> result; - rtc::ByteBufferReader payload_reader( - reinterpret_cast<const char*>(payload.data()), payload.size()); + rtc::ByteBufferReader payload_reader(payload); while (payload_reader.Length() > 0) { Obu obu; payload_reader.ReadUInt8(&obu.header); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.cc new file mode 100644 index 0000000000..313680cc87 --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.cc @@ -0,0 +1,350 @@ +/* + * Copyright (c) 2023 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/rtp_rtcp/source/rtp_packetizer_h265.h" + +#include <vector> + +#include "absl/types/optional.h" +#include "common_video/h264/h264_common.h" +#include "common_video/h265/h265_common.h" +#include "modules/rtp_rtcp/source/byte_io.h" +#include "rtc_base/logging.h" + +namespace webrtc { +namespace { + +// The payload header consists of the same +// fields (F, Type, LayerId and TID) as the NAL unit header. Refer to +// section 4.2 in RFC 7798. +constexpr size_t kH265PayloadHeaderSize = 2; +// Unlike H.264, H265 NAL header is 2-bytes. +constexpr size_t kH265NalHeaderSize = 2; +// H265's FU is constructed of 2-byte payload header, 1-byte FU header and FU +// payload. +constexpr size_t kH265FuHeaderSize = 1; +// The NALU size for H265 RTP aggregated packet indicates the size of the NAL +// unit is 2-bytes. +constexpr size_t kH265LengthFieldSize = 2; + +enum H265NalHdrMasks { + kH265FBit = 0x80, + kH265TypeMask = 0x7E, + kH265LayerIDHMask = 0x1, + kH265LayerIDLMask = 0xF8, + kH265TIDMask = 0x7, + kH265TypeMaskN = 0x81, + kH265TypeMaskInFuHeader = 0x3F +}; + +// Bit masks for FU headers. +enum H265FuBitmasks { + kH265SBitMask = 0x80, + kH265EBitMask = 0x40, + kH265FuTypeBitMask = 0x3F +}; + +} // namespace + +RtpPacketizerH265::RtpPacketizerH265(rtc::ArrayView<const uint8_t> payload, + PayloadSizeLimits limits) + : limits_(limits), num_packets_left_(0) { + for (const auto& nalu : + H264::FindNaluIndices(payload.data(), payload.size())) { + input_fragments_.push_back( + payload.subview(nalu.payload_start_offset, nalu.payload_size)); + } + + if (!GeneratePackets()) { + // If failed to generate all the packets, discard already generated + // packets in case the caller would ignore return value and still try to + // call NextPacket(). + num_packets_left_ = 0; + while (!packets_.empty()) { + packets_.pop(); + } + } +} + +RtpPacketizerH265::~RtpPacketizerH265() = default; + +size_t RtpPacketizerH265::NumPackets() const { + return num_packets_left_; +} + +bool RtpPacketizerH265::GeneratePackets() { + for (size_t i = 0; i < input_fragments_.size();) { + int fragment_len = input_fragments_[i].size(); + int single_packet_capacity = limits_.max_payload_len; + if (input_fragments_.size() == 1) { + single_packet_capacity -= limits_.single_packet_reduction_len; + } else if (i == 0) { + single_packet_capacity -= limits_.first_packet_reduction_len; + } else if (i + 1 == input_fragments_.size()) { + // Pretend that last fragment is larger instead of making last packet + // smaller. + single_packet_capacity -= limits_.last_packet_reduction_len; + } + if (fragment_len > single_packet_capacity) { + if (!PacketizeFu(i)) { + return false; + } + ++i; + } else { + i = PacketizeAp(i); + } + } + return true; +} + +bool RtpPacketizerH265::PacketizeFu(size_t fragment_index) { + // Fragment payload into packets (FU). + // Strip out the original header and leave room for the FU header. + rtc::ArrayView<const uint8_t> fragment = input_fragments_[fragment_index]; + PayloadSizeLimits limits = limits_; + // Refer to section 4.4.3 in RFC7798, each FU fragment will have a 2-bytes + // payload header and a one-byte FU header. DONL is not supported so ignore + // its size when calculating max_payload_len. + limits.max_payload_len -= kH265FuHeaderSize + kH265PayloadHeaderSize; + + // Update single/first/last packet reductions unless it is single/first/last + // fragment. + if (input_fragments_.size() != 1) { + // if this fragment is put into a single packet, it might still be the + // first or the last packet in the whole sequence of packets. + if (fragment_index == input_fragments_.size() - 1) { + limits.single_packet_reduction_len = limits_.last_packet_reduction_len; + } else if (fragment_index == 0) { + limits.single_packet_reduction_len = limits_.first_packet_reduction_len; + } else { + limits.single_packet_reduction_len = 0; + } + } + if (fragment_index != 0) { + limits.first_packet_reduction_len = 0; + } + if (fragment_index != input_fragments_.size() - 1) { + limits.last_packet_reduction_len = 0; + } + + // Strip out the original header. + size_t payload_left = fragment.size() - kH265NalHeaderSize; + int offset = kH265NalHeaderSize; + + std::vector<int> payload_sizes = SplitAboutEqually(payload_left, limits); + if (payload_sizes.empty()) { + return false; + } + + for (size_t i = 0; i < payload_sizes.size(); ++i) { + int packet_length = payload_sizes[i]; + RTC_CHECK_GT(packet_length, 0); + uint16_t header = (fragment[0] << 8) | fragment[1]; + packets_.push({.source_fragment = fragment.subview(offset, packet_length), + .first_fragment = (i == 0), + .last_fragment = (i == payload_sizes.size() - 1), + .aggregated = false, + .header = header}); + offset += packet_length; + payload_left -= packet_length; + } + num_packets_left_ += payload_sizes.size(); + RTC_CHECK_EQ(payload_left, 0); + return true; +} + +int RtpPacketizerH265::PacketizeAp(size_t fragment_index) { + // Aggregate fragments into one packet. + size_t payload_size_left = limits_.max_payload_len; + if (input_fragments_.size() == 1) { + payload_size_left -= limits_.single_packet_reduction_len; + } else if (fragment_index == 0) { + payload_size_left -= limits_.first_packet_reduction_len; + } + int aggregated_fragments = 0; + size_t fragment_headers_length = 0; + rtc::ArrayView<const uint8_t> fragment = input_fragments_[fragment_index]; + RTC_CHECK_GE(payload_size_left, fragment.size()); + ++num_packets_left_; + + auto payload_size_needed = [&] { + size_t fragment_size = fragment.size() + fragment_headers_length; + if (input_fragments_.size() == 1) { + // Single fragment, single packet, payload_size_left already adjusted + // with limits_.single_packet_reduction_len. + return fragment_size; + } + if (fragment_index == input_fragments_.size() - 1) { + // Last fragment, so this might be the last packet. + return fragment_size + limits_.last_packet_reduction_len; + } + return fragment_size; + }; + + while (payload_size_left >= payload_size_needed()) { + RTC_CHECK_GT(fragment.size(), 0); + packets_.push({.source_fragment = fragment, + .first_fragment = (aggregated_fragments == 0), + .last_fragment = false, + .aggregated = true, + .header = fragment[0]}); + payload_size_left -= fragment.size(); + payload_size_left -= fragment_headers_length; + + fragment_headers_length = kH265LengthFieldSize; + // If we are going to try to aggregate more fragments into this packet + // we need to add the AP NALU header and a length field for the first + // NALU of this packet. + if (aggregated_fragments == 0) { + fragment_headers_length += kH265PayloadHeaderSize + kH265LengthFieldSize; + } + ++aggregated_fragments; + + // Next fragment. + ++fragment_index; + if (fragment_index == input_fragments_.size()) { + break; + } + fragment = input_fragments_[fragment_index]; + } + RTC_CHECK_GT(aggregated_fragments, 0); + packets_.back().last_fragment = true; + return fragment_index; +} + +bool RtpPacketizerH265::NextPacket(RtpPacketToSend* rtp_packet) { + RTC_DCHECK(rtp_packet); + + if (packets_.empty()) { + return false; + } + + PacketUnit packet = packets_.front(); + + if (packet.first_fragment && packet.last_fragment) { + // Single NAL unit packet. Do not support DONL for single NAL unit packets, + // DONL field is not present. + size_t bytes_to_send = packet.source_fragment.size(); + uint8_t* buffer = rtp_packet->AllocatePayload(bytes_to_send); + memcpy(buffer, packet.source_fragment.data(), bytes_to_send); + packets_.pop(); + input_fragments_.pop_front(); + } else if (packet.aggregated) { + NextAggregatePacket(rtp_packet); + } else { + NextFragmentPacket(rtp_packet); + } + rtp_packet->SetMarker(packets_.empty()); + --num_packets_left_; + return true; +} + +void RtpPacketizerH265::NextAggregatePacket(RtpPacketToSend* rtp_packet) { + size_t payload_capacity = rtp_packet->FreeCapacity(); + RTC_CHECK_GE(payload_capacity, kH265PayloadHeaderSize); + uint8_t* buffer = rtp_packet->AllocatePayload(payload_capacity); + RTC_CHECK(buffer); + PacketUnit* packet = &packets_.front(); + RTC_CHECK(packet->first_fragment); + + /* + +---------------+---------------+ + |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7| + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + |F| Type | LayerId | TID | + +-------------+-----------------+ + */ + // Refer to section section 4.4.2 for aggregation packets and modify type to + // 48 in PayloadHdr for aggregate packet. Do not support DONL for aggregation + // packets, DONL field is not present. + uint8_t payload_hdr_h = packet->header >> 8; + uint8_t payload_hdr_l = packet->header & 0xFF; + uint8_t layer_id_h = payload_hdr_h & kH265LayerIDHMask; + payload_hdr_h = (payload_hdr_h & kH265TypeMaskN) | + (H265::NaluType::kAp << 1) | layer_id_h; + buffer[0] = payload_hdr_h; + buffer[1] = payload_hdr_l; + + int index = kH265PayloadHeaderSize; + bool is_last_fragment = packet->last_fragment; + while (packet->aggregated) { + // Add NAL unit length field. + rtc::ArrayView<const uint8_t> fragment = packet->source_fragment; + ByteWriter<uint16_t>::WriteBigEndian(&buffer[index], fragment.size()); + index += kH265LengthFieldSize; + // Add NAL unit. + memcpy(&buffer[index], fragment.data(), fragment.size()); + index += fragment.size(); + packets_.pop(); + input_fragments_.pop_front(); + if (is_last_fragment) { + break; + } + packet = &packets_.front(); + is_last_fragment = packet->last_fragment; + } + RTC_CHECK(is_last_fragment); + rtp_packet->SetPayloadSize(index); +} + +void RtpPacketizerH265::NextFragmentPacket(RtpPacketToSend* rtp_packet) { + PacketUnit* packet = &packets_.front(); + // NAL unit fragmented over multiple packets (FU). + // We do not send original NALU header, so it will be replaced by the + // PayloadHdr of the first packet. + /* + +---------------+---------------+ + |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7| + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + |F| Type | LayerId | TID | + +-------------+-----------------+ + */ + // Refer to section section 4.4.3 for aggregation packets and modify type to + // 49 in PayloadHdr for aggregate packet. + uint8_t payload_hdr_h = + packet->header >> 8; // 1-bit F, 6-bit type, 1-bit layerID highest-bit + uint8_t payload_hdr_l = packet->header & 0xFF; + uint8_t layer_id_h = payload_hdr_h & kH265LayerIDHMask; + uint8_t fu_header = 0; + /* + +---------------+ + |0|1|2|3|4|5|6|7| + +-+-+-+-+-+-+-+-+ + |S|E| FuType | + +---------------+ + */ + // S bit indicates the start of a fragmented NAL unit. + // E bit indicates the end of a fragmented NAL unit. + // FuType must be equal to the field type value of the fragmented NAL unit. + fu_header |= (packet->first_fragment ? kH265SBitMask : 0); + fu_header |= (packet->last_fragment ? kH265EBitMask : 0); + uint8_t type = (payload_hdr_h & kH265TypeMask) >> 1; + fu_header |= type; + // Now update payload_hdr_h with FU type. + payload_hdr_h = (payload_hdr_h & kH265TypeMaskN) | + (H265::NaluType::kFu << 1) | layer_id_h; + rtc::ArrayView<const uint8_t> fragment = packet->source_fragment; + uint8_t* buffer = rtp_packet->AllocatePayload( + kH265FuHeaderSize + kH265PayloadHeaderSize + fragment.size()); + RTC_CHECK(buffer); + buffer[0] = payload_hdr_h; + buffer[1] = payload_hdr_l; + buffer[2] = fu_header; + + // Do not support DONL for fragmentation units, DONL field is not present. + memcpy(buffer + kH265FuHeaderSize + kH265PayloadHeaderSize, fragment.data(), + fragment.size()); + if (packet->last_fragment) { + input_fragments_.pop_front(); + } + packets_.pop(); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h new file mode 100644 index 0000000000..95442f795c --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2023 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_ +#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_ + +#include <deque> +#include <queue> +#include <string> + +#include "api/array_view.h" +#include "modules/rtp_rtcp/source/rtp_format.h" +#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" + +namespace webrtc { + +class RtpPacketizerH265 : public RtpPacketizer { + public: + // Initialize with payload from encoder. + // The payload_data must be exactly one encoded H.265 frame. + // For H265 we only support tx-mode SRST. + RtpPacketizerH265(rtc::ArrayView<const uint8_t> payload, + PayloadSizeLimits limits); + + RtpPacketizerH265(const RtpPacketizerH265&) = delete; + RtpPacketizerH265& operator=(const RtpPacketizerH265&) = delete; + + ~RtpPacketizerH265() override; + + size_t NumPackets() const override; + + // Get the next payload with H.265 payload header. + // Write payload and set marker bit of the `packet`. + // Returns true on success or false if there was no payload to packetize. + bool NextPacket(RtpPacketToSend* rtp_packet) override; + + private: + struct PacketUnit { + rtc::ArrayView<const uint8_t> source_fragment; + bool first_fragment = false; + bool last_fragment = false; + bool aggregated = false; + uint16_t header = 0; + }; + std::deque<rtc::ArrayView<const uint8_t>> input_fragments_; + std::queue<PacketUnit> packets_; + + bool GeneratePackets(); + bool PacketizeFu(size_t fragment_index); + int PacketizeAp(size_t fragment_index); + + void NextAggregatePacket(RtpPacketToSend* rtp_packet); + void NextFragmentPacket(RtpPacketToSend* rtp_packet); + + const PayloadSizeLimits limits_; + size_t num_packets_left_ = 0; +}; +} // namespace webrtc +#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_ diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265_unittest.cc new file mode 100644 index 0000000000..cb1de334c0 --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265_unittest.cc @@ -0,0 +1,525 @@ +/* + * Copyright (c) 2023 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/rtp_rtcp/source/rtp_packetizer_h265.h" + +#include <vector> + +#include "common_video/h265/h265_common.h" +#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" +#include "modules/rtp_rtcp/source/byte_io.h" +#include "test/gmock.h" +#include "test/gtest.h" + +namespace webrtc { +namespace { + +using ::testing::Each; +using ::testing::ElementsAre; +using ::testing::ElementsAreArray; +using ::testing::Eq; +using ::testing::IsEmpty; +using ::testing::SizeIs; + +constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr; +constexpr size_t kMaxPayloadSize = 1200; +constexpr size_t kLengthFieldLength = 2; +constexpr RtpPacketizer::PayloadSizeLimits kNoLimits; + +constexpr size_t kNalHeaderSize = 2; +constexpr size_t kFuHeaderSize = 3; + +constexpr uint8_t kNaluTypeMask = 0x7E; + +// Bit masks for FU headers. +constexpr uint8_t kH265SBit = 0x80; +constexpr uint8_t kH265EBit = 0x40; + +// Creates Buffer that looks like nal unit of given size. +rtc::Buffer GenerateNalUnit(size_t size) { + RTC_CHECK_GT(size, 0); + rtc::Buffer buffer(size); + // Set some valid header with type TRAIL_R and temporal id + buffer[0] = 2; + buffer[1] = 2; + for (size_t i = 2; i < size; ++i) { + buffer[i] = static_cast<uint8_t>(i); + } + // Last byte shouldn't be 0, or it may be counted as part of next 4-byte start + // sequence. + buffer[size - 1] |= 0x10; + return buffer; +} + +// Create frame consisting of nalus of given size. +rtc::Buffer CreateFrame(std::initializer_list<size_t> nalu_sizes) { + static constexpr int kStartCodeSize = 3; + rtc::Buffer frame(absl::c_accumulate(nalu_sizes, size_t{0}) + + kStartCodeSize * nalu_sizes.size()); + size_t offset = 0; + for (size_t nalu_size : nalu_sizes) { + EXPECT_GE(nalu_size, 1u); + // Insert nalu start code + frame[offset] = 0; + frame[offset + 1] = 0; + frame[offset + 2] = 1; + // Set some valid header. + frame[offset + 3] = 2; + // Fill payload avoiding accidental start codes + if (nalu_size > 1) { + memset(frame.data() + offset + 4, 0x3f, nalu_size - 1); + } + offset += (kStartCodeSize + nalu_size); + } + return frame; +} + +// Create frame consisting of given nalus. +rtc::Buffer CreateFrame(rtc::ArrayView<const rtc::Buffer> nalus) { + static constexpr int kStartCodeSize = 3; + int frame_size = 0; + for (const rtc::Buffer& nalu : nalus) { + frame_size += (kStartCodeSize + nalu.size()); + } + rtc::Buffer frame(frame_size); + size_t offset = 0; + for (const rtc::Buffer& nalu : nalus) { + // Insert nalu start code + frame[offset] = 0; + frame[offset + 1] = 0; + frame[offset + 2] = 1; + // Copy the nalu unit. + memcpy(frame.data() + offset + 3, nalu.data(), nalu.size()); + offset += (kStartCodeSize + nalu.size()); + } + return frame; +} + +std::vector<RtpPacketToSend> FetchAllPackets(RtpPacketizerH265* packetizer) { + std::vector<RtpPacketToSend> result; + size_t num_packets = packetizer->NumPackets(); + result.reserve(num_packets); + RtpPacketToSend packet(kNoExtensions); + while (packetizer->NextPacket(&packet)) { + result.push_back(packet); + } + EXPECT_THAT(result, SizeIs(num_packets)); + return result; +} + +// Single nalu tests. +TEST(RtpPacketizerH265Test, SingleNalu) { + const uint8_t frame[] = {0, 0, 1, H265::kIdrWRadl, 0xFF}; + + RtpPacketizerH265 packetizer(frame, kNoLimits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(1)); + EXPECT_THAT(packets[0].payload(), ElementsAre(H265::kIdrWRadl, 0xFF)); +} + +TEST(RtpPacketizerH265Test, SingleNaluTwoPackets) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = kMaxPayloadSize; + rtc::Buffer nalus[] = {GenerateNalUnit(kMaxPayloadSize), + GenerateNalUnit(100)}; + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(2)); + EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0])); + EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[1])); +} + +TEST(RtpPacketizerH265Test, + SingleNaluFirstPacketReductionAppliesOnlyToFirstFragment) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 200; + limits.first_packet_reduction_len = 5; + rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/195), + GenerateNalUnit(/*size=*/200), + GenerateNalUnit(/*size=*/200)}; + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(3)); + EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0])); + EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[1])); + EXPECT_THAT(packets[2].payload(), ElementsAreArray(nalus[2])); +} + +TEST(RtpPacketizerH265Test, + SingleNaluLastPacketReductionAppliesOnlyToLastFragment) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 200; + limits.last_packet_reduction_len = 5; + rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/200), + GenerateNalUnit(/*size=*/200), + GenerateNalUnit(/*size=*/195)}; + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(3)); + EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0])); + EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[1])); + EXPECT_THAT(packets[2].payload(), ElementsAreArray(nalus[2])); +} + +TEST(RtpPacketizerH265Test, + SingleNaluFirstAndLastPacketReductionSumsForSinglePacket) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 200; + limits.first_packet_reduction_len = 20; + limits.last_packet_reduction_len = 30; + rtc::Buffer frame = CreateFrame({150}); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + EXPECT_THAT(packets, SizeIs(1)); +} + +// Aggregation tests. +TEST(RtpPacketizerH265Test, ApRespectsNoPacketReduction) { + rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/2), + GenerateNalUnit(/*size=*/2), + GenerateNalUnit(/*size=*/0x123)}; + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, kNoLimits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(1)); + auto payload = packets[0].payload(); + int type = H265::ParseNaluType(payload[0]); + EXPECT_EQ(payload.size(), + kNalHeaderSize + 3 * kLengthFieldLength + 2 + 2 + 0x123); + + EXPECT_EQ(type, H265::NaluType::kAp); + payload = payload.subview(kNalHeaderSize); + // 1st fragment. + EXPECT_THAT(payload.subview(0, kLengthFieldLength), + ElementsAre(0, 2)); // Size. + EXPECT_THAT(payload.subview(kLengthFieldLength, 2), + ElementsAreArray(nalus[0])); + payload = payload.subview(kLengthFieldLength + 2); + // 2nd fragment. + EXPECT_THAT(payload.subview(0, kLengthFieldLength), + ElementsAre(0, 2)); // Size. + EXPECT_THAT(payload.subview(kLengthFieldLength, 2), + ElementsAreArray(nalus[1])); + payload = payload.subview(kLengthFieldLength + 2); + // 3rd fragment. + EXPECT_THAT(payload.subview(0, kLengthFieldLength), + ElementsAre(0x1, 0x23)); // Size. + EXPECT_THAT(payload.subview(kLengthFieldLength), ElementsAreArray(nalus[2])); +} + +TEST(RtpPacketizerH265Test, ApRespectsFirstPacketReduction) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1000; + limits.first_packet_reduction_len = 100; + const size_t kFirstFragmentSize = + limits.max_payload_len - limits.first_packet_reduction_len; + rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/kFirstFragmentSize), + GenerateNalUnit(/*size=*/2), + GenerateNalUnit(/*size=*/2)}; + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(2)); + // Expect 1st packet is single nalu. + EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0])); + // Expect 2nd packet is aggregate of last two fragments. + // The size of H265 nal_unit_header is 2 bytes, according to 7.3.1.2 + // in H265 spec. Aggregation packet type is 48, and nuh_temporal_id_plus1 + // is 2, so the nal_unit_header should be "01100000 00000010", + // which is 96 and 2. + EXPECT_THAT(packets[1].payload(), + ElementsAre(96, 2, // + 0, 2, nalus[1][0], nalus[1][1], // + 0, 2, nalus[2][0], nalus[2][1])); +} + +TEST(RtpPacketizerH265Test, ApRespectsLastPacketReduction) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1000; + limits.last_packet_reduction_len = 100; + const size_t kLastFragmentSize = + limits.max_payload_len - limits.last_packet_reduction_len; + rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/2), + GenerateNalUnit(/*size=*/2), + GenerateNalUnit(/*size=*/kLastFragmentSize)}; + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(2)); + // Expect 1st packet is aggregate of 1st two fragments. + EXPECT_THAT(packets[0].payload(), + ElementsAre(96, 2, // + 0, 2, nalus[0][0], nalus[0][1], // + 0, 2, nalus[1][0], nalus[1][1])); + // Expect 2nd packet is single nalu. + EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[2])); +} + +TEST(RtpPacketizerH265Test, TooSmallForApHeaders) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1000; + const size_t kLastFragmentSize = + limits.max_payload_len - 3 * kLengthFieldLength - 4; + rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/2), + GenerateNalUnit(/*size=*/2), + GenerateNalUnit(/*size=*/kLastFragmentSize)}; + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(2)); + // Expect 1st packet is aggregate of 1st two fragments. + EXPECT_THAT(packets[0].payload(), + ElementsAre(96, 2, // + 0, 2, nalus[0][0], nalus[0][1], // + 0, 2, nalus[1][0], nalus[1][1])); + // Expect 2nd packet is single nalu. + EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[2])); +} + +TEST(RtpPacketizerH265Test, LastFragmentFitsInSingleButNotLastPacket) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1178; + limits.first_packet_reduction_len = 0; + limits.last_packet_reduction_len = 20; + limits.single_packet_reduction_len = 20; + // Actual sizes, which triggered this bug. + rtc::Buffer frame = CreateFrame({20, 8, 18, 1161}); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + // Last packet has to be of correct size. + // Incorrect implementation might miss this constraint and not split the last + // fragment in two packets. + EXPECT_LE(static_cast<int>(packets.back().payload_size()), + limits.max_payload_len - limits.last_packet_reduction_len); +} + +// Splits frame with payload size `frame_payload_size` without fragmentation, +// Returns sizes of the payloads excluding FU headers. +std::vector<int> TestFu(size_t frame_payload_size, + const RtpPacketizer::PayloadSizeLimits& limits) { + rtc::Buffer nalu[] = {GenerateNalUnit(kNalHeaderSize + frame_payload_size)}; + rtc::Buffer frame = CreateFrame(nalu); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + EXPECT_GE(packets.size(), 2u); // Single packet indicates it is not FU. + std::vector<uint16_t> fu_header; + std::vector<int> payload_sizes; + + for (const RtpPacketToSend& packet : packets) { + auto payload = packet.payload(); + EXPECT_GT(payload.size(), kFuHeaderSize); + // FU header is after the 2-bytes size PayloadHdr according to 4.4.3 in spec + fu_header.push_back(payload[2]); + payload_sizes.push_back(payload.size() - kFuHeaderSize); + } + + EXPECT_TRUE(fu_header.front() & kH265SBit); + EXPECT_TRUE(fu_header.back() & kH265EBit); + // Clear S and E bits before testing all are duplicating same original header. + fu_header.front() &= ~kH265SBit; + fu_header.back() &= ~kH265EBit; + uint8_t nalu_type = (nalu[0][0] & kNaluTypeMask) >> 1; + EXPECT_THAT(fu_header, Each(Eq(nalu_type))); + + return payload_sizes; +} + +// Fragmentation tests. +TEST(RtpPacketizerH265Test, FuOddSize) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1200; + EXPECT_THAT(TestFu(1200, limits), ElementsAre(600, 600)); +} + +TEST(RtpPacketizerH265Test, FuWithFirstPacketReduction) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1200; + limits.first_packet_reduction_len = 4; + limits.single_packet_reduction_len = 4; + EXPECT_THAT(TestFu(1198, limits), ElementsAre(597, 601)); +} + +TEST(RtpPacketizerH265Test, FuWithLastPacketReduction) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1200; + limits.last_packet_reduction_len = 4; + limits.single_packet_reduction_len = 4; + EXPECT_THAT(TestFu(1198, limits), ElementsAre(601, 597)); +} + +TEST(RtpPacketizerH265Test, FuWithSinglePacketReduction) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1199; + limits.single_packet_reduction_len = 200; + EXPECT_THAT(TestFu(1000, limits), ElementsAre(500, 500)); +} + +TEST(RtpPacketizerH265Test, FuEvenSize) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1200; + EXPECT_THAT(TestFu(1201, limits), ElementsAre(600, 601)); +} + +TEST(RtpPacketizerH265Test, FuRounding) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1448; + EXPECT_THAT(TestFu(10123, limits), + ElementsAre(1265, 1265, 1265, 1265, 1265, 1266, 1266, 1266)); +} + +TEST(RtpPacketizerH265Test, FuBig) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 1200; + // Generate 10 full sized packets, leave room for FU headers. + EXPECT_THAT( + TestFu(10 * (1200 - kFuHeaderSize), limits), + ElementsAre(1197, 1197, 1197, 1197, 1197, 1197, 1197, 1197, 1197, 1197)); +} + +struct PacketInfo { + bool first_fragment = false; + bool last_fragment = false; + bool aggregated = false; + int nalu_index = 0; + int nalu_number = 0; + int payload_size = 0; + int start_offset = 0; +}; + +struct MixedApFuTestParams { + std::vector<int> nalus; + int expect_packetsSize = 0; + std::vector<PacketInfo> expected_packets; +}; + +class RtpPacketizerH265ParametrizedTest + : public ::testing::TestWithParam<MixedApFuTestParams> {}; + +// Fragmentation + aggregation mixed testing. +TEST_P(RtpPacketizerH265ParametrizedTest, MixedApFu) { + RtpPacketizer::PayloadSizeLimits limits; + const MixedApFuTestParams params = GetParam(); + limits.max_payload_len = 100; + std::vector<rtc::Buffer> nalus; + nalus.reserve(params.nalus.size()); + + // Generate nalus according to size specified in paramters + for (size_t index = 0; index < params.nalus.size(); index++) { + nalus.push_back(GenerateNalUnit(params.nalus[index])); + } + rtc::Buffer frame = CreateFrame(nalus); + + RtpPacketizerH265 packetizer(frame, limits); + std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer); + + ASSERT_THAT(packets, SizeIs(params.expect_packetsSize)); + for (int i = 0; i < params.expect_packetsSize; i++) { + PacketInfo expected_packet = params.expected_packets[i]; + if (expected_packet.aggregated) { + int type = H265::ParseNaluType(packets[i].payload()[0]); + EXPECT_THAT(type, H265::NaluType::kAp); + auto payload = packets[i].payload().subview(kNalHeaderSize); + int offset = 0; + // Generated AP packet header and payload align + for (int j = expected_packet.nalu_index; j < expected_packet.nalu_number; + j++) { + EXPECT_THAT(payload.subview(0, kLengthFieldLength), + ElementsAre(0, nalus[j].size())); + EXPECT_THAT( + payload.subview(offset + kLengthFieldLength, nalus[j].size()), + ElementsAreArray(nalus[j])); + offset += kLengthFieldLength + nalus[j].size(); + } + } else { + uint8_t fu_header = 0; + fu_header |= (expected_packet.first_fragment ? kH265SBit : 0); + fu_header |= (expected_packet.last_fragment ? kH265EBit : 0); + fu_header |= H265::NaluType::kTrailR; + EXPECT_THAT(packets[i].payload().subview(0, kFuHeaderSize), + ElementsAre(98, 2, fu_header)); + EXPECT_THAT( + packets[i].payload().subview(kFuHeaderSize), + ElementsAreArray(nalus[expected_packet.nalu_index].data() + + kNalHeaderSize + expected_packet.start_offset, + expected_packet.payload_size)); + } + } +} + +INSTANTIATE_TEST_SUITE_P( + RtpPacketizerH265Test, + RtpPacketizerH265ParametrizedTest, + testing::Values( + // FU + AP + FU. + // GenerateNalUnit will include 2 bytes nalu header, for FU packet split + // calculation, this 2-byte nalu header length should be excluded. + MixedApFuTestParams{.nalus = {140, 20, 20, 160}, + .expect_packetsSize = 5, + .expected_packets = {{.first_fragment = true, + .nalu_index = 0, + .payload_size = 69, + .start_offset = 0}, + {.last_fragment = true, + .nalu_index = 0, + .payload_size = 69, + .start_offset = 69}, + {.aggregated = true, + .nalu_index = 1, + .nalu_number = 2}, + {.first_fragment = true, + .nalu_index = 3, + .payload_size = 79, + .start_offset = 0}, + {.last_fragment = true, + .nalu_index = 3, + .payload_size = 79, + .start_offset = 79}}}, + // AP + FU + AP + MixedApFuTestParams{ + .nalus = {20, 20, 160, 30, 30}, + .expect_packetsSize = 4, + .expected_packets = { + {.aggregated = true, .nalu_index = 0, .nalu_number = 2}, + {.first_fragment = true, + .nalu_index = 2, + .payload_size = 79, + .start_offset = 0}, + {.last_fragment = true, + .nalu_index = 2, + .payload_size = 79, + .start_offset = 79}, + {.aggregated = true, .nalu_index = 3, .nalu_number = 2}}})); + +} // namespace +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc index ff482b39b6..31e8b71117 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc @@ -53,8 +53,8 @@ RTCPSender::Configuration AddRtcpSendEvaluationCallback( RtpPacketHistory::PaddingMode GetPaddingMode( const FieldTrialsView* field_trials) { - if (field_trials && - field_trials->IsEnabled("WebRTC-PaddingMode-RecentLargePacket")) { + if (!field_trials || + !field_trials->IsDisabled("WebRTC-PaddingMode-RecentLargePacket")) { return RtpPacketHistory::PaddingMode::kRecentLargePacket; } return RtpPacketHistory::PaddingMode::kPriority; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc index b826c30e07..9d2258dc66 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc @@ -254,7 +254,8 @@ bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) { return false; } - std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket(); + std::unique_ptr<RtpPacketToSend> packet = + rtp_sender_->AllocatePacket(frame.csrcs); packet->SetMarker(MarkerBit(frame.type, frame.payload_id)); packet->SetPayloadType(frame.payload_id); packet->SetTimestamp(frame.rtp_timestamp); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h index 662f908216..83a2cb211f 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h @@ -61,6 +61,9 @@ class RTPSenderAudio { // header-extension-for-audio-level-indication. // Valid range is [0,127]. Actual value is negative. absl::optional<int> audio_level_dbov; + + // Contributing sources list. + rtc::ArrayView<const uint32_t> csrcs; }; bool SendAudio(const RtpAudioFrame& frame); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc index 0db610c149..724cd3a5e0 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc @@ -222,4 +222,19 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { EXPECT_FALSE(transport_.last_sent_packet().Marker()); } +TEST_F(RtpSenderAudioTest, SendsCsrcs) { + const char payload_name[] = "audio"; + const uint8_t payload_type = 127; + ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload( + payload_name, payload_type, 48000, 0, 1500)); + uint8_t payload[] = {47, 11, 32, 93, 89}; + + std::vector<uint32_t> csrcs({123, 456, 789}); + + ASSERT_TRUE(rtp_sender_audio_->SendAudio( + {.payload = payload, .payload_id = payload_type, .csrcs = csrcs})); + + EXPECT_EQ(transport_.last_sent_packet().Csrcs(), csrcs); +} + } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc index 9d7c58d19a..ae9eb6b4bd 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc @@ -19,10 +19,17 @@ #include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h" #include "rtc_base/checks.h" #include "rtc_base/event.h" +#include "rtc_base/logging.h" namespace webrtc { namespace { +// Using a reasonable default of 10ms for the retransmission delay for frames +// not coming from this sender's encoder. This is usually taken from an +// estimate of the RTT of the link,so 10ms should be a reasonable estimate for +// frames being re-transmitted to a peer, probably on the same network. +const TimeDelta kDefaultRetransmissionsTime = TimeDelta::Millis(10); + class TransformableVideoSenderFrame : public TransformableVideoFrameInterface { public: TransformableVideoSenderFrame(const EncodedImage& encoded_image, @@ -155,6 +162,17 @@ bool RTPSenderVideoFrameTransformerDelegate::TransformFrame( const EncodedImage& encoded_image, RTPVideoHeader video_header, TimeDelta expected_retransmission_time) { + { + MutexLock lock(&sender_lock_); + if (short_circuit_) { + sender_->SendVideo(payload_type, codec_type, rtp_timestamp, + encoded_image.CaptureTime(), + *encoded_image.GetEncodedData(), encoded_image.size(), + video_header, expected_retransmission_time, + /*csrcs=*/{}); + return true; + } + } frame_transformer_->Transform(std::make_unique<TransformableVideoSenderFrame>( encoded_image, video_header, payload_type, codec_type, rtp_timestamp, expected_retransmission_time, ssrc_, @@ -177,6 +195,11 @@ void RTPSenderVideoFrameTransformerDelegate::OnTransformedFrame( }); } +void RTPSenderVideoFrameTransformerDelegate::StartShortCircuiting() { + MutexLock lock(&sender_lock_); + short_circuit_ = true; +} + void RTPSenderVideoFrameTransformerDelegate::SendVideo( std::unique_ptr<TransformableFrameInterface> transformed_frame) const { RTC_DCHECK_RUN_ON(transformation_queue_.get()); @@ -200,15 +223,17 @@ void RTPSenderVideoFrameTransformerDelegate::SendVideo( auto* transformed_video_frame = static_cast<TransformableVideoFrameInterface*>(transformed_frame.get()); VideoFrameMetadata metadata = transformed_video_frame->Metadata(); - sender_->SendVideo( - transformed_video_frame->GetPayloadType(), metadata.GetCodec(), - transformed_video_frame->GetTimestamp(), - /*capture_time=*/Timestamp::MinusInfinity(), - transformed_video_frame->GetData(), - transformed_video_frame->GetData().size(), - RTPVideoHeader::FromMetadata(metadata), - /*expected_retransmission_time=*/TimeDelta::PlusInfinity(), - metadata.GetCsrcs()); + // TODO(bugs.webrtc.org/14708): Use an actual RTT estimate for the + // retransmission time instead of a const default, in the same way as a + // locally encoded frame. + sender_->SendVideo(transformed_video_frame->GetPayloadType(), + metadata.GetCodec(), + transformed_video_frame->GetTimestamp(), + /*capture_time=*/Timestamp::MinusInfinity(), + transformed_video_frame->GetData(), + transformed_video_frame->GetData().size(), + RTPVideoHeader::FromMetadata(metadata), + kDefaultRetransmissionsTime, metadata.GetCsrcs()); } } @@ -253,13 +278,14 @@ std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame( ? VideoFrameType::kVideoFrameKey : VideoFrameType::kVideoFrameDelta; // TODO(bugs.webrtc.org/14708): Fill in other EncodedImage parameters - + // TODO(bugs.webrtc.org/14708): Use an actual RTT estimate for the + // retransmission time instead of a const default, in the same way as a + // locally encoded frame. VideoFrameMetadata metadata = original->Metadata(); RTPVideoHeader new_header = RTPVideoHeader::FromMetadata(metadata); return std::make_unique<TransformableVideoSenderFrame>( encoded_image, new_header, original->GetPayloadType(), new_header.codec, - original->GetTimestamp(), - /*expected_retransmission_time=*/TimeDelta::PlusInfinity(), + original->GetTimestamp(), kDefaultRetransmissionsTime, original->GetSsrc(), metadata.GetCsrcs(), original->GetRid()); } diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h index 3379ead364..1f70a23ccc 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h @@ -76,6 +76,8 @@ class RTPSenderVideoFrameTransformerDelegate : public TransformedFrameCallback { void OnTransformedFrame( std::unique_ptr<TransformableFrameInterface> frame) override; + void StartShortCircuiting() override; + // Delegates the call to RTPSendVideo::SendVideo on the `encoder_queue_`. void SendVideo(std::unique_ptr<TransformableFrameInterface> frame) const RTC_RUN_ON(transformation_queue_); @@ -109,6 +111,7 @@ class RTPSenderVideoFrameTransformerDelegate : public TransformedFrameCallback { // Used when the encoded frames arrives without a current task queue. This can // happen if a hardware encoder was used. std::unique_ptr<TaskQueueBase, TaskQueueDeleter> transformation_queue_; + bool short_circuit_ RTC_GUARDED_BY(sender_lock_) = false; }; // Method to support cloning a Sender frame from another frame diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc index a376be77b4..6790fc3a71 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc @@ -83,7 +83,7 @@ class RtpSenderVideoFrameTransformerDelegateTest : public ::testing::Test { delegate->TransformFrame( /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2, encoded_image, RTPVideoHeader::FromMetadata(metadata), - /*expected_retransmission_time=*/TimeDelta::PlusInfinity()); + /*expected_retransmission_time=*/TimeDelta::Millis(10)); return frame; } @@ -123,7 +123,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest, delegate->TransformFrame( /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2, encoded_image, RTPVideoHeader(), - /*expected_retransmission_time=*/TimeDelta::PlusInfinity()); + /*expected_retransmission_time=*/TimeDelta::Millis(10)); } TEST_F(RtpSenderVideoFrameTransformerDelegateTest, @@ -260,7 +260,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest, test_sender_, SendVideo(payload_type, absl::make_optional(kVideoCodecVP8), timestamp, /*capture_time=*/Timestamp::MinusInfinity(), buffer, _, _, - /*expected_retransmission_time=*/TimeDelta::PlusInfinity(), + /*expected_retransmission_time=*/TimeDelta::Millis(10), frame_csrcs)) .WillOnce(WithoutArgs([&] { event.Set(); @@ -289,5 +289,29 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest, SettingRTPTimestamp) { EXPECT_EQ(video_frame.GetTimestamp(), rtp_timestamp); } +TEST_F(RtpSenderVideoFrameTransformerDelegateTest, + ShortCircuitingSkipsTransform) { + auto delegate = rtc::make_ref_counted<RTPSenderVideoFrameTransformerDelegate>( + &test_sender_, frame_transformer_, + /*ssrc=*/1111, time_controller_.CreateTaskQueueFactory().get()); + EXPECT_CALL(*frame_transformer_, + RegisterTransformedFrameSinkCallback(_, 1111)); + delegate->Init(); + + delegate->StartShortCircuiting(); + + // Will not call the actual transformer. + EXPECT_CALL(*frame_transformer_, Transform).Times(0); + // Will pass the frame straight to the reciever. + EXPECT_CALL(test_sender_, SendVideo); + + EncodedImage encoded_image; + encoded_image.SetEncodedData(EncodedImageBuffer::Create(1)); + delegate->TransformFrame( + /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2, + encoded_image, RTPVideoHeader(), + /*expected_retransmission_time=*/TimeDelta::Millis(10)); +} + } // namespace } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc index 94c9249e16..7af945c623 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc @@ -17,6 +17,7 @@ #include "absl/memory/memory.h" #include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h" #include "rtc_base/checks.h" +#include "rtc_base/logging.h" #include "rtc_base/thread.h" namespace webrtc { @@ -123,9 +124,14 @@ void RtpVideoStreamReceiverFrameTransformerDelegate::Reset() { void RtpVideoStreamReceiverFrameTransformerDelegate::TransformFrame( std::unique_ptr<RtpFrameObject> frame) { RTC_DCHECK_RUN_ON(&network_sequence_checker_); - frame_transformer_->Transform( - std::make_unique<TransformableVideoReceiverFrame>(std::move(frame), ssrc_, - receiver_)); + if (short_circuit_) { + // Just pass the frame straight back. + receiver_->ManageFrame(std::move(frame)); + } else { + frame_transformer_->Transform( + std::make_unique<TransformableVideoReceiverFrame>(std::move(frame), + ssrc_, receiver_)); + } } void RtpVideoStreamReceiverFrameTransformerDelegate::OnTransformedFrame( @@ -138,6 +144,20 @@ void RtpVideoStreamReceiverFrameTransformerDelegate::OnTransformedFrame( }); } +void RtpVideoStreamReceiverFrameTransformerDelegate::StartShortCircuiting() { + rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate> delegate( + this); + network_thread_->PostTask([delegate = std::move(delegate)]() mutable { + delegate->StartShortCircuitingOnNetworkSequence(); + }); +} + +void RtpVideoStreamReceiverFrameTransformerDelegate:: + StartShortCircuitingOnNetworkSequence() { + RTC_DCHECK_RUN_ON(&network_sequence_checker_); + short_circuit_ = true; +} + void RtpVideoStreamReceiverFrameTransformerDelegate::ManageFrame( std::unique_ptr<TransformableFrameInterface> frame) { RTC_DCHECK_RUN_ON(&network_sequence_checker_); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h index 20f9a5caa9..02f2e53923 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h @@ -54,6 +54,8 @@ class RtpVideoStreamReceiverFrameTransformerDelegate void OnTransformedFrame( std::unique_ptr<TransformableFrameInterface> frame) override; + void StartShortCircuiting() override; + // Delegates the call to RtpVideoFrameReceiver::ManageFrame on the // `network_thread_`. void ManageFrame(std::unique_ptr<TransformableFrameInterface> frame); @@ -62,6 +64,8 @@ class RtpVideoStreamReceiverFrameTransformerDelegate ~RtpVideoStreamReceiverFrameTransformerDelegate() override = default; private: + void StartShortCircuitingOnNetworkSequence(); + RTC_NO_UNIQUE_ADDRESS SequenceChecker network_sequence_checker_; RtpVideoFrameReceiver* receiver_ RTC_GUARDED_BY(network_sequence_checker_); rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_ @@ -69,6 +73,7 @@ class RtpVideoStreamReceiverFrameTransformerDelegate TaskQueueBase* const network_thread_; const uint32_t ssrc_; Clock* const clock_; + bool short_circuit_ RTC_GUARDED_BY(network_sequence_checker_) = false; }; } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc index f403c91a74..cf3062610f 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc @@ -349,5 +349,28 @@ TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest, rtc::ThreadManager::ProcessAllMessageQueuesForTesting(); } +TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest, + ShortCircuitingSkipsTransform) { + rtc::AutoThread main_thread_; + TestRtpVideoFrameReceiver receiver; + auto mock_frame_transformer = + rtc::make_ref_counted<NiceMock<MockFrameTransformer>>(); + SimulatedClock clock(0); + auto delegate = + rtc::make_ref_counted<RtpVideoStreamReceiverFrameTransformerDelegate>( + &receiver, &clock, mock_frame_transformer, rtc::Thread::Current(), + 1111); + delegate->Init(); + + delegate->StartShortCircuiting(); + rtc::ThreadManager::ProcessAllMessageQueuesForTesting(); + + // Will not call the actual transformer. + EXPECT_CALL(*mock_frame_transformer, Transform).Times(0); + // Will pass the frame straight to the reciever. + EXPECT_CALL(receiver, ManageFrame); + delegate->TransformFrame(CreateRtpFrameObject()); +} + } // namespace } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc index 870f788538..30bbbc5000 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc @@ -188,8 +188,7 @@ VectorObuInfo ParseObus( VectorObuInfo obu_infos; bool expect_continues_obu = false; for (rtc::ArrayView<const uint8_t> rtp_payload : rtp_payloads) { - rtc::ByteBufferReader payload( - reinterpret_cast<const char*>(rtp_payload.data()), rtp_payload.size()); + rtc::ByteBufferReader payload(rtp_payload); uint8_t aggregation_header; if (!payload.ReadUInt8(&aggregation_header)) { RTC_DLOG(LS_WARNING) diff --git a/third_party/libwebrtc/modules/third_party/fft/fft_gn/moz.build b/third_party/libwebrtc/modules/third_party/fft/fft_gn/moz.build index d2e3ea0128..c260743e28 100644 --- a/third_party/libwebrtc/modules/third_party/fft/fft_gn/moz.build +++ b/third_party/libwebrtc/modules/third_party/fft/fft_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/third_party/g711/g711_3p_gn/moz.build b/third_party/libwebrtc/modules/third_party/g711/g711_3p_gn/moz.build index aa7a21a680..c2d2597a21 100644 --- a/third_party/libwebrtc/modules/third_party/g711/g711_3p_gn/moz.build +++ b/third_party/libwebrtc/modules/third_party/g711/g711_3p_gn/moz.build @@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/third_party/g722/g722_3p_gn/moz.build b/third_party/libwebrtc/modules/third_party/g722/g722_3p_gn/moz.build index 41a8c05bae..468cc88c65 100644 --- a/third_party/libwebrtc/modules/third_party/g722/g722_3p_gn/moz.build +++ b/third_party/libwebrtc/modules/third_party/g722/g722_3p_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/utility/utility_gn/moz.build b/third_party/libwebrtc/modules/utility/utility_gn/moz.build index b6921b7626..6c17ac236e 100644 --- a/third_party/libwebrtc/modules/utility/utility_gn/moz.build +++ b/third_party/libwebrtc/modules/utility/utility_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": ] OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_capture/linux/camera_portal.cc b/third_party/libwebrtc/modules/video_capture/linux/camera_portal.cc index 85b9f20228..106ca1682c 100644 --- a/third_party/libwebrtc/modules/video_capture/linux/camera_portal.cc +++ b/third_party/libwebrtc/modules/video_capture/linux/camera_portal.cc @@ -15,6 +15,7 @@ #include "modules/portal/pipewire_utils.h" #include "modules/portal/xdg_desktop_portal_utils.h" +#include "rtc_base/synchronization/mutex.h" namespace webrtc { @@ -54,7 +55,9 @@ class CameraPortalPrivate { GAsyncResult* result, gpointer user_data); - CameraPortal::PortalNotifier* notifier_ = nullptr; + webrtc::Mutex notifier_lock_; + CameraPortal::PortalNotifier* notifier_ RTC_GUARDED_BY(¬ifier_lock_) = + nullptr; GDBusConnection* connection_ = nullptr; GDBusProxy* proxy_ = nullptr; @@ -66,6 +69,11 @@ CameraPortalPrivate::CameraPortalPrivate(CameraPortal::PortalNotifier* notifier) : notifier_(notifier) {} CameraPortalPrivate::~CameraPortalPrivate() { + { + webrtc::MutexLock lock(¬ifier_lock_); + notifier_ = nullptr; + } + if (access_request_signal_id_) { g_dbus_connection_signal_unsubscribe(connection_, access_request_signal_id_); @@ -229,7 +237,11 @@ void CameraPortalPrivate::OnOpenResponse(GDBusProxy* proxy, } void CameraPortalPrivate::OnPortalDone(RequestResponse result, int fd) { - notifier_->OnCameraRequestResult(result, fd); + webrtc::MutexLock lock(¬ifier_lock_); + if (notifier_) { + notifier_->OnCameraRequestResult(result, fd); + notifier_ = nullptr; + } } CameraPortal::CameraPortal(PortalNotifier* notifier) diff --git a/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.cc b/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.cc index ad6cea57b8..31d922035b 100644 --- a/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.cc +++ b/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.cc @@ -20,10 +20,10 @@ #include <vector> +#include "modules/video_capture/linux/pipewire_session.h" #include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_defines.h" #include "modules/video_capture/video_capture_impl.h" -#include "modules/video_capture/video_capture_options.h" #include "rtc_base/logging.h" namespace webrtc { @@ -38,6 +38,8 @@ int32_t DeviceInfoPipeWire::Init() { DeviceInfoPipeWire::~DeviceInfoPipeWire() = default; uint32_t DeviceInfoPipeWire::NumberOfDevices() { + RTC_CHECK(pipewire_session_); + return pipewire_session_->nodes().size(); } @@ -50,6 +52,8 @@ int32_t DeviceInfoPipeWire::GetDeviceName(uint32_t deviceNumber, uint32_t productUniqueIdUTF8Length, pid_t* pid, bool* deviceIsPlaceholder) { + RTC_CHECK(pipewire_session_); + if (deviceNumber >= NumberOfDevices()) return -1; @@ -85,6 +89,8 @@ int32_t DeviceInfoPipeWire::GetDeviceName(uint32_t deviceNumber, int32_t DeviceInfoPipeWire::CreateCapabilityMap( const char* deviceUniqueIdUTF8) { + RTC_CHECK(pipewire_session_); + for (auto& node : pipewire_session_->nodes()) { if (node.unique_id().compare(deviceUniqueIdUTF8) != 0) continue; diff --git a/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.h b/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.h index 1a1324e92b..00715c94bc 100644 --- a/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.h +++ b/third_party/libwebrtc/modules/video_capture/linux/device_info_pipewire.h @@ -14,7 +14,7 @@ #include <stdint.h> #include "modules/video_capture/device_info_impl.h" -#include "modules/video_capture/linux/pipewire_session.h" +#include "modules/video_capture/video_capture_options.h" namespace webrtc { namespace videocapturemodule { diff --git a/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc b/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc index eaeed26b7c..401c38f9c5 100644 --- a/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc +++ b/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc @@ -57,24 +57,6 @@ #define BUF_LEN ( 1024 * ( EVENT_SIZE + 16 ) ) #endif -// These defines are here to support building on kernel 3.16 which some -// downstream projects, e.g. Firefox, use. -// TODO(apehrson): Remove them and their undefs when no longer needed. -#ifndef V4L2_PIX_FMT_ABGR32 -#define ABGR32_OVERRIDE 1 -#define V4L2_PIX_FMT_ABGR32 v4l2_fourcc('A', 'R', '2', '4') -#endif - -#ifndef V4L2_PIX_FMT_ARGB32 -#define ARGB32_OVERRIDE 1 -#define V4L2_PIX_FMT_ARGB32 v4l2_fourcc('B', 'A', '2', '4') -#endif - -#ifndef V4L2_PIX_FMT_RGBA32 -#define RGBA32_OVERRIDE 1 -#define V4L2_PIX_FMT_RGBA32 v4l2_fourcc('A', 'B', '2', '4') -#endif - namespace webrtc { namespace videocapturemodule { #ifdef WEBRTC_LINUX diff --git a/third_party/libwebrtc/modules/video_capture/linux/video_capture_pipewire.cc b/third_party/libwebrtc/modules/video_capture/linux/video_capture_pipewire.cc index 8af483636a..319824d3c5 100644 --- a/third_party/libwebrtc/modules/video_capture/linux/video_capture_pipewire.cc +++ b/third_party/libwebrtc/modules/video_capture/linux/video_capture_pipewire.cc @@ -178,8 +178,7 @@ int32_t VideoCaptureModulePipeWire::StartCapture( int res = pw_stream_connect( stream_, PW_DIRECTION_INPUT, node_id_, static_cast<enum pw_stream_flags>(PW_STREAM_FLAG_AUTOCONNECT | - PW_STREAM_FLAG_DONT_RECONNECT | - PW_STREAM_FLAG_MAP_BUFFERS), + PW_STREAM_FLAG_DONT_RECONNECT), params.data(), params.size()); if (res != 0) { RTC_LOG(LS_ERROR) << "Could not connect to camera stream: " @@ -312,11 +311,11 @@ void VideoCaptureModulePipeWire::OnFormatChanged(const struct spa_pod* format) { 0); } + const int buffer_types = + (1 << SPA_DATA_DmaBuf) | (1 << SPA_DATA_MemFd) | (1 << SPA_DATA_MemPtr); spa_pod_builder_add( &builder, SPA_PARAM_BUFFERS_buffers, SPA_POD_CHOICE_RANGE_Int(8, 1, 32), - SPA_PARAM_BUFFERS_dataType, - SPA_POD_CHOICE_FLAGS_Int((1 << SPA_DATA_MemFd) | (1 << SPA_DATA_MemPtr)), - 0); + SPA_PARAM_BUFFERS_dataType, SPA_POD_CHOICE_FLAGS_Int(buffer_types), 0); params.push_back( static_cast<spa_pod*>(spa_pod_builder_pop(&builder, &frame))); @@ -384,14 +383,15 @@ void VideoCaptureModulePipeWire::ProcessBuffers() { RTC_CHECK_RUNS_SERIALIZED(&capture_checker_); while (pw_buffer* buffer = pw_stream_dequeue_buffer(stream_)) { + spa_buffer* spaBuffer = buffer->buffer; struct spa_meta_header* h; h = static_cast<struct spa_meta_header*>( - spa_buffer_find_meta_data(buffer->buffer, SPA_META_Header, sizeof(*h))); + spa_buffer_find_meta_data(spaBuffer, SPA_META_Header, sizeof(*h))); struct spa_meta_videotransform* videotransform; videotransform = static_cast<struct spa_meta_videotransform*>(spa_buffer_find_meta_data( - buffer->buffer, SPA_META_VideoTransform, sizeof(*videotransform))); + spaBuffer, SPA_META_VideoTransform, sizeof(*videotransform))); if (videotransform) { VideoRotation rotation = VideorotationFromPipeWireTransform(videotransform->transform); @@ -401,11 +401,35 @@ void VideoCaptureModulePipeWire::ProcessBuffers() { if (h->flags & SPA_META_HEADER_FLAG_CORRUPTED) { RTC_LOG(LS_INFO) << "Dropping corruped frame."; - } else { - IncomingFrame(static_cast<unsigned char*>(buffer->buffer->datas[0].data), - buffer->buffer->datas[0].chunk->size, - configured_capability_); + pw_stream_queue_buffer(stream_, buffer); + continue; + } + + if (spaBuffer->datas[0].type == SPA_DATA_DmaBuf || + spaBuffer->datas[0].type == SPA_DATA_MemFd) { + ScopedBuf frame; + frame.initialize( + static_cast<uint8_t*>( + mmap(nullptr, + spaBuffer->datas[0].maxsize + spaBuffer->datas[0].mapoffset, + PROT_READ, MAP_PRIVATE, spaBuffer->datas[0].fd, 0)), + spaBuffer->datas[0].maxsize + spaBuffer->datas[0].mapoffset, + spaBuffer->datas[0].fd, spaBuffer->datas[0].type == SPA_DATA_DmaBuf); + + if (!frame) { + RTC_LOG(LS_ERROR) << "Failed to mmap the memory: " + << std::strerror(errno); + return; + } + + IncomingFrame( + SPA_MEMBER(frame.get(), spaBuffer->datas[0].mapoffset, uint8_t), + spaBuffer->datas[0].chunk->size, configured_capability_); + } else { // SPA_DATA_MemPtr + IncomingFrame(static_cast<uint8_t*>(spaBuffer->datas[0].data), + spaBuffer->datas[0].chunk->size, configured_capability_); } + pw_stream_queue_buffer(stream_, buffer); } } diff --git a/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc b/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc index c887683dc8..08d23f7f58 100644 --- a/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc +++ b/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc @@ -294,7 +294,7 @@ int32_t VideoCaptureModuleV4L2::StartCapture( if (_captureThread.empty()) { quit_ = false; _captureThread = rtc::PlatformThread::SpawnJoinable( - [self = rtc::scoped_refptr(this)] { + [self = scoped_refptr(this)] { while (self->CaptureProcess()) { } }, diff --git a/third_party/libwebrtc/modules/video_capture/video_capture.h b/third_party/libwebrtc/modules/video_capture/video_capture.h index 378a53b4d2..f59c34f8b2 100644 --- a/third_party/libwebrtc/modules/video_capture/video_capture.h +++ b/third_party/libwebrtc/modules/video_capture/video_capture.h @@ -34,7 +34,7 @@ protected: virtual ~VideoInputFeedBack(){} }; -class VideoCaptureModule : public rtc::RefCountInterface { +class VideoCaptureModule : public RefCountInterface { public: // Interface for receiving information about available camera devices. class DeviceInfo { diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_internal_impl_gn/moz.build b/third_party/libwebrtc/modules/video_capture/video_capture_internal_impl_gn/moz.build index 24988a1ffc..f58aa8e782 100644 --- a/third_party/libwebrtc/modules/video_capture/video_capture_internal_impl_gn/moz.build +++ b/third_party/libwebrtc/modules/video_capture/video_capture_internal_impl_gn/moz.build @@ -267,7 +267,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -277,10 +276,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["WEBRTC_USE_PIPEWIRE"] = True diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_module_gn/moz.build b/third_party/libwebrtc/modules/video_capture/video_capture_module_gn/moz.build index 49c62d5cf6..820d5655df 100644 --- a/third_party/libwebrtc/modules/video_capture/video_capture_module_gn/moz.build +++ b/third_party/libwebrtc/modules/video_capture/video_capture_module_gn/moz.build @@ -204,7 +204,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -214,10 +213,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/BUILD.gn b/third_party/libwebrtc/modules/video_coding/BUILD.gn index cca9d8883a..d9e614ff81 100644 --- a/third_party/libwebrtc/modules/video_coding/BUILD.gn +++ b/third_party/libwebrtc/modules/video_coding/BUILD.gn @@ -757,7 +757,6 @@ if (rtc_include_tests) { "../../api/video_codecs:video_codecs_api", "../../modules/utility:utility", "../../rtc_base:checks", - "../../rtc_base:ignore_wundef", "../../sdk/android:internal_jni", "../../sdk/android:native_api_base", "../../sdk/android:native_api_codecs", @@ -852,8 +851,6 @@ if (rtc_include_tests) { "../../api:frame_generator_api", "../../api:scoped_refptr", "../../api:sequence_checker", - "../../api:video_codec_stats_api", - "../../api:video_codec_tester_api", "../../api:videocodec_test_fixture_api", "../../api/numerics:numerics", "../../api/task_queue", @@ -995,46 +992,6 @@ if (rtc_include_tests) { ] } - rtc_library("video_codec_tester") { - testonly = true - sources = [ - "codecs/test/video_codec_analyzer.cc", - "codecs/test/video_codec_analyzer.h", - "codecs/test/video_codec_stats_impl.cc", - "codecs/test/video_codec_stats_impl.h", - "codecs/test/video_codec_tester_impl.cc", - "codecs/test/video_codec_tester_impl.h", - ] - - deps = [ - ":video_coding_utility", - "../../api:sequence_checker", - "../../api:video_codec_stats_api", - "../../api:video_codec_tester_api", - "../../api/numerics:numerics", - "../../api/task_queue:default_task_queue_factory", - "../../api/test/metrics:metrics_logger", - "../../api/units:data_rate", - "../../api/units:frequency", - "../../api/units:time_delta", - "../../api/units:timestamp", - "../../api/video:encoded_image", - "../../api/video:resolution", - "../../api/video:video_codec_constants", - "../../api/video:video_frame", - "../../rtc_base:checks", - "../../rtc_base:rtc_event", - "../../rtc_base:task_queue_for_test", - "../../rtc_base:timeutils", - "../../rtc_base/system:no_unique_address", - "../../system_wrappers", - "../../test:video_test_support", - "//third_party/libyuv", - ] - - absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] - } - rtc_test("video_codec_perf_tests") { testonly = true @@ -1042,28 +999,20 @@ if (rtc_include_tests) { deps = [ ":video_codec_interface", - ":video_codec_tester", - "../../api:create_video_codec_tester_api", - "../../api:video_codec_tester_api", - "../../api:videocodec_test_stats_api", "../../api/test/metrics:global_metrics_logger_and_exporter", "../../api/units:data_rate", "../../api/units:frequency", - "../../api/video:encoded_image", "../../api/video:resolution", - "../../api/video:video_frame", - "../../api/video_codecs:scalability_mode", - "../../api/video_codecs:video_codecs_api", - "../../media:rtc_internal_video_codecs", + "../../api/video_codecs:builtin_video_decoder_factory", + "../../api/video_codecs:builtin_video_encoder_factory", + "../../modules/video_coding/svc:scalability_mode_util", "../../rtc_base:logging", + "../../rtc_base:stringutils", "../../test:fileutils", "../../test:test_flags", "../../test:test_main", "../../test:test_support", - "../../test:video_test_support", - "../rtp_rtcp:rtp_rtcp_format", - "svc:scalability_mode_util", - "//third_party/libyuv", + "../../test:video_codec_tester", ] if (is_android) { @@ -1191,9 +1140,6 @@ if (rtc_include_tests) { sources = [ "chain_diff_calculator_unittest.cc", - "codecs/test/video_codec_analyzer_unittest.cc", - "codecs/test/video_codec_stats_impl_unittest.cc", - "codecs/test/video_codec_tester_impl_unittest.cc", "codecs/test/videocodec_test_fixture_config_unittest.cc", "codecs/test/videocodec_test_stats_impl_unittest.cc", "codecs/test/videoprocessor_unittest.cc", @@ -1248,7 +1194,6 @@ if (rtc_include_tests) { ":packet_buffer", ":simulcast_test_fixture_impl", ":video_codec_interface", - ":video_codec_tester", ":video_codecs_test_framework", ":video_coding", ":video_coding_legacy", @@ -1271,7 +1216,6 @@ if (rtc_include_tests) { "../../api:rtp_packet_info", "../../api:scoped_refptr", "../../api:simulcast_test_fixture_api", - "../../api:video_codec_tester_api", "../../api:videocodec_test_fixture_api", "../../api/task_queue", "../../api/task_queue:default_task_queue_factory", @@ -1297,6 +1241,7 @@ if (rtc_include_tests) { "../../common_video/generic_frame_descriptor", "../../common_video/test:utilities", "../../media:media_constants", + "../../media:rtc_internal_video_codecs", "../../media:rtc_media_base", "../../rtc_base:checks", "../../rtc_base:gunit_helpers", diff --git a/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_gn/moz.build index dd8e979e41..144097f87a 100644 --- a/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/codec_globals_headers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/codec_globals_headers_gn/moz.build index 73fce5bf02..cf74ae964c 100644 --- a/third_party/libwebrtc/modules/video_coding/codec_globals_headers_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/codec_globals_headers_gn/moz.build @@ -180,16 +180,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] -if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": - - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_gn/moz.build b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_gn/moz.build index e67bb6616d..bfe37b935d 100644 --- a/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.cc deleted file mode 100644 index 772c15734a..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.cc +++ /dev/null @@ -1,193 +0,0 @@ -/* - * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/video_coding/codecs/test/video_codec_analyzer.h" - -#include <memory> - -#include "api/task_queue/default_task_queue_factory.h" -#include "api/video/i420_buffer.h" -#include "api/video/video_codec_constants.h" -#include "api/video/video_frame.h" -#include "rtc_base/checks.h" -#include "rtc_base/event.h" -#include "rtc_base/time_utils.h" -#include "third_party/libyuv/include/libyuv/compare.h" - -namespace webrtc { -namespace test { - -namespace { -using Psnr = VideoCodecStats::Frame::Psnr; - -Psnr CalcPsnr(const I420BufferInterface& ref_buffer, - const I420BufferInterface& dec_buffer) { - RTC_CHECK_EQ(ref_buffer.width(), dec_buffer.width()); - RTC_CHECK_EQ(ref_buffer.height(), dec_buffer.height()); - - uint64_t sse_y = libyuv::ComputeSumSquareErrorPlane( - dec_buffer.DataY(), dec_buffer.StrideY(), ref_buffer.DataY(), - ref_buffer.StrideY(), dec_buffer.width(), dec_buffer.height()); - - uint64_t sse_u = libyuv::ComputeSumSquareErrorPlane( - dec_buffer.DataU(), dec_buffer.StrideU(), ref_buffer.DataU(), - ref_buffer.StrideU(), dec_buffer.width() / 2, dec_buffer.height() / 2); - - uint64_t sse_v = libyuv::ComputeSumSquareErrorPlane( - dec_buffer.DataV(), dec_buffer.StrideV(), ref_buffer.DataV(), - ref_buffer.StrideV(), dec_buffer.width() / 2, dec_buffer.height() / 2); - - int num_y_samples = dec_buffer.width() * dec_buffer.height(); - Psnr psnr; - psnr.y = libyuv::SumSquareErrorToPsnr(sse_y, num_y_samples); - psnr.u = libyuv::SumSquareErrorToPsnr(sse_u, num_y_samples / 4); - psnr.v = libyuv::SumSquareErrorToPsnr(sse_v, num_y_samples / 4); - - return psnr; -} - -} // namespace - -VideoCodecAnalyzer::VideoCodecAnalyzer( - ReferenceVideoSource* reference_video_source) - : reference_video_source_(reference_video_source), num_frames_(0) { - sequence_checker_.Detach(); -} - -void VideoCodecAnalyzer::StartEncode(const VideoFrame& input_frame) { - int64_t encode_start_us = rtc::TimeMicros(); - task_queue_.PostTask( - [this, timestamp_rtp = input_frame.timestamp(), encode_start_us]() { - RTC_DCHECK_RUN_ON(&sequence_checker_); - - RTC_CHECK(frame_num_.find(timestamp_rtp) == frame_num_.end()); - frame_num_[timestamp_rtp] = num_frames_++; - - stats_.AddFrame({.frame_num = frame_num_[timestamp_rtp], - .timestamp_rtp = timestamp_rtp, - .encode_start = Timestamp::Micros(encode_start_us)}); - }); -} - -void VideoCodecAnalyzer::FinishEncode(const EncodedImage& frame) { - int64_t encode_finished_us = rtc::TimeMicros(); - - task_queue_.PostTask([this, timestamp_rtp = frame.RtpTimestamp(), - spatial_idx = frame.SpatialIndex().value_or(0), - temporal_idx = frame.TemporalIndex().value_or(0), - width = frame._encodedWidth, - height = frame._encodedHeight, - frame_type = frame._frameType, - frame_size_bytes = frame.size(), qp = frame.qp_, - encode_finished_us]() { - RTC_DCHECK_RUN_ON(&sequence_checker_); - - if (spatial_idx > 0) { - VideoCodecStats::Frame* base_frame = - stats_.GetFrame(timestamp_rtp, /*spatial_idx=*/0); - - stats_.AddFrame({.frame_num = base_frame->frame_num, - .timestamp_rtp = timestamp_rtp, - .spatial_idx = spatial_idx, - .encode_start = base_frame->encode_start}); - } - - VideoCodecStats::Frame* fs = stats_.GetFrame(timestamp_rtp, spatial_idx); - fs->spatial_idx = spatial_idx; - fs->temporal_idx = temporal_idx; - fs->width = width; - fs->height = height; - fs->frame_size = DataSize::Bytes(frame_size_bytes); - fs->qp = qp; - fs->keyframe = frame_type == VideoFrameType::kVideoFrameKey; - fs->encode_time = Timestamp::Micros(encode_finished_us) - fs->encode_start; - fs->encoded = true; - }); -} - -void VideoCodecAnalyzer::StartDecode(const EncodedImage& frame) { - int64_t decode_start_us = rtc::TimeMicros(); - task_queue_.PostTask([this, timestamp_rtp = frame.RtpTimestamp(), - spatial_idx = frame.SpatialIndex().value_or(0), - frame_size_bytes = frame.size(), decode_start_us]() { - RTC_DCHECK_RUN_ON(&sequence_checker_); - - VideoCodecStats::Frame* fs = stats_.GetFrame(timestamp_rtp, spatial_idx); - if (fs == nullptr) { - if (frame_num_.find(timestamp_rtp) == frame_num_.end()) { - frame_num_[timestamp_rtp] = num_frames_++; - } - stats_.AddFrame({.frame_num = frame_num_[timestamp_rtp], - .timestamp_rtp = timestamp_rtp, - .spatial_idx = spatial_idx, - .frame_size = DataSize::Bytes(frame_size_bytes)}); - fs = stats_.GetFrame(timestamp_rtp, spatial_idx); - } - - fs->decode_start = Timestamp::Micros(decode_start_us); - }); -} - -void VideoCodecAnalyzer::FinishDecode(const VideoFrame& frame, - int spatial_idx) { - int64_t decode_finished_us = rtc::TimeMicros(); - task_queue_.PostTask([this, timestamp_rtp = frame.timestamp(), spatial_idx, - width = frame.width(), height = frame.height(), - decode_finished_us]() { - RTC_DCHECK_RUN_ON(&sequence_checker_); - VideoCodecStats::Frame* fs = stats_.GetFrame(timestamp_rtp, spatial_idx); - fs->decode_time = Timestamp::Micros(decode_finished_us) - fs->decode_start; - - if (!fs->encoded) { - fs->width = width; - fs->height = height; - } - - fs->decoded = true; - }); - - if (reference_video_source_ != nullptr) { - // Copy hardware-backed frame into main memory to release output buffers - // which number may be limited in hardware decoders. - rtc::scoped_refptr<I420BufferInterface> decoded_buffer = - frame.video_frame_buffer()->ToI420(); - - task_queue_.PostTask([this, decoded_buffer, - timestamp_rtp = frame.timestamp(), spatial_idx]() { - RTC_DCHECK_RUN_ON(&sequence_checker_); - VideoFrame ref_frame = reference_video_source_->GetFrame( - timestamp_rtp, {.width = decoded_buffer->width(), - .height = decoded_buffer->height()}); - rtc::scoped_refptr<I420BufferInterface> ref_buffer = - ref_frame.video_frame_buffer()->ToI420(); - - Psnr psnr = CalcPsnr(*decoded_buffer, *ref_buffer); - - VideoCodecStats::Frame* fs = - this->stats_.GetFrame(timestamp_rtp, spatial_idx); - fs->psnr = psnr; - }); - } -} - -std::unique_ptr<VideoCodecStats> VideoCodecAnalyzer::GetStats() { - std::unique_ptr<VideoCodecStats> stats; - rtc::Event ready; - task_queue_.PostTask([this, &stats, &ready]() mutable { - RTC_DCHECK_RUN_ON(&sequence_checker_); - stats.reset(new VideoCodecStatsImpl(stats_)); - ready.Set(); - }); - ready.Wait(rtc::Event::kForever); - return stats; -} - -} // namespace test -} // namespace webrtc diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.h b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.h deleted file mode 100644 index 29ca8ee2ff..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.h +++ /dev/null @@ -1,75 +0,0 @@ -/* - * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_ANALYZER_H_ -#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_ANALYZER_H_ - -#include <map> -#include <memory> - -#include "absl/types/optional.h" -#include "api/sequence_checker.h" -#include "api/test/video_codec_tester.h" -#include "api/video/encoded_image.h" -#include "api/video/resolution.h" -#include "api/video/video_frame.h" -#include "modules/video_coding/codecs/test/video_codec_stats_impl.h" -#include "rtc_base/system/no_unique_address.h" -#include "rtc_base/task_queue_for_test.h" - -namespace webrtc { -namespace test { - -// Analyzer measures and collects metrics necessary for evaluation of video -// codec quality and performance. This class is thread-safe. -class VideoCodecAnalyzer { - public: - // An interface that provides reference frames for spatial quality analysis. - class ReferenceVideoSource { - public: - virtual ~ReferenceVideoSource() = default; - - virtual VideoFrame GetFrame(uint32_t timestamp_rtp, - Resolution resolution) = 0; - }; - - explicit VideoCodecAnalyzer( - ReferenceVideoSource* reference_video_source = nullptr); - - void StartEncode(const VideoFrame& frame); - - void FinishEncode(const EncodedImage& frame); - - void StartDecode(const EncodedImage& frame); - - void FinishDecode(const VideoFrame& frame, int spatial_idx); - - std::unique_ptr<VideoCodecStats> GetStats(); - - protected: - TaskQueueForTest task_queue_; - - ReferenceVideoSource* const reference_video_source_; - - VideoCodecStatsImpl stats_ RTC_GUARDED_BY(sequence_checker_); - - // Map from RTP timestamp to frame number. - std::map<uint32_t, int> frame_num_ RTC_GUARDED_BY(sequence_checker_); - - // Processed frames counter. - int num_frames_ RTC_GUARDED_BY(sequence_checker_); - - RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_; -}; - -} // namespace test -} // namespace webrtc - -#endif // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_ANALYZER_H_ diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer_unittest.cc deleted file mode 100644 index 03146417da..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer_unittest.cc +++ /dev/null @@ -1,127 +0,0 @@ -/* - * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/video_coding/codecs/test/video_codec_analyzer.h" - -#include "absl/types/optional.h" -#include "api/video/i420_buffer.h" -#include "test/gmock.h" -#include "test/gtest.h" -#include "third_party/libyuv/include/libyuv/planar_functions.h" - -namespace webrtc { -namespace test { - -namespace { -using ::testing::Return; -using ::testing::Values; -using Psnr = VideoCodecStats::Frame::Psnr; - -const uint32_t kTimestamp = 3000; -const int kSpatialIdx = 2; - -class MockReferenceVideoSource - : public VideoCodecAnalyzer::ReferenceVideoSource { - public: - MOCK_METHOD(VideoFrame, GetFrame, (uint32_t, Resolution), (override)); -}; - -VideoFrame CreateVideoFrame(uint32_t timestamp_rtp, - uint8_t y = 0, - uint8_t u = 0, - uint8_t v = 0) { - rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(2, 2)); - - libyuv::I420Rect(buffer->MutableDataY(), buffer->StrideY(), - buffer->MutableDataU(), buffer->StrideU(), - buffer->MutableDataV(), buffer->StrideV(), 0, 0, - buffer->width(), buffer->height(), y, u, v); - - return VideoFrame::Builder() - .set_video_frame_buffer(buffer) - .set_timestamp_rtp(timestamp_rtp) - .build(); -} - -EncodedImage CreateEncodedImage(uint32_t timestamp_rtp, int spatial_idx = 0) { - EncodedImage encoded_image; - encoded_image.SetRtpTimestamp(timestamp_rtp); - encoded_image.SetSpatialIndex(spatial_idx); - return encoded_image; -} -} // namespace - -TEST(VideoCodecAnalyzerTest, StartEncode) { - VideoCodecAnalyzer analyzer; - analyzer.StartEncode(CreateVideoFrame(kTimestamp)); - - auto fs = analyzer.GetStats()->Slice(); - EXPECT_EQ(1u, fs.size()); - EXPECT_EQ(fs[0].timestamp_rtp, kTimestamp); -} - -TEST(VideoCodecAnalyzerTest, FinishEncode) { - VideoCodecAnalyzer analyzer; - analyzer.StartEncode(CreateVideoFrame(kTimestamp)); - - EncodedImage encoded_frame = CreateEncodedImage(kTimestamp, kSpatialIdx); - analyzer.FinishEncode(encoded_frame); - - auto fs = analyzer.GetStats()->Slice(); - EXPECT_EQ(2u, fs.size()); - EXPECT_EQ(kSpatialIdx, fs[1].spatial_idx); -} - -TEST(VideoCodecAnalyzerTest, StartDecode) { - VideoCodecAnalyzer analyzer; - analyzer.StartDecode(CreateEncodedImage(kTimestamp, kSpatialIdx)); - - auto fs = analyzer.GetStats()->Slice(); - EXPECT_EQ(1u, fs.size()); - EXPECT_EQ(kTimestamp, fs[0].timestamp_rtp); -} - -TEST(VideoCodecAnalyzerTest, FinishDecode) { - VideoCodecAnalyzer analyzer; - analyzer.StartDecode(CreateEncodedImage(kTimestamp, kSpatialIdx)); - VideoFrame decoded_frame = CreateVideoFrame(kTimestamp); - analyzer.FinishDecode(decoded_frame, kSpatialIdx); - - auto fs = analyzer.GetStats()->Slice(); - EXPECT_EQ(1u, fs.size()); - EXPECT_EQ(decoded_frame.width(), fs[0].width); - EXPECT_EQ(decoded_frame.height(), fs[0].height); -} - -TEST(VideoCodecAnalyzerTest, ReferenceVideoSource) { - MockReferenceVideoSource reference_video_source; - VideoCodecAnalyzer analyzer(&reference_video_source); - analyzer.StartDecode(CreateEncodedImage(kTimestamp, kSpatialIdx)); - - EXPECT_CALL(reference_video_source, GetFrame) - .WillOnce(Return(CreateVideoFrame(kTimestamp, /*y=*/0, - /*u=*/0, /*v=*/0))); - - analyzer.FinishDecode( - CreateVideoFrame(kTimestamp, /*value_y=*/1, /*value_u=*/2, /*value_v=*/3), - kSpatialIdx); - - auto fs = analyzer.GetStats()->Slice(); - EXPECT_EQ(1u, fs.size()); - EXPECT_TRUE(fs[0].psnr.has_value()); - - const Psnr& psnr = *fs[0].psnr; - EXPECT_NEAR(psnr.y, 48, 1); - EXPECT_NEAR(psnr.u, 42, 1); - EXPECT_NEAR(psnr.v, 38, 1); -} - -} // namespace test -} // namespace webrtc diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl.cc deleted file mode 100644 index 9808e2a601..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl.cc +++ /dev/null @@ -1,278 +0,0 @@ -/* - * Copyright (c) 2023 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/video_coding/codecs/test/video_codec_stats_impl.h" - -#include <algorithm> - -#include "api/numerics/samples_stats_counter.h" -#include "api/test/metrics/metrics_logger.h" -#include "rtc_base/checks.h" -#include "rtc_base/time_utils.h" - -namespace webrtc { -namespace test { -namespace { -using Frame = VideoCodecStats::Frame; -using Stream = VideoCodecStats::Stream; - -constexpr Frequency k90kHz = Frequency::Hertz(90000); - -class LeakyBucket { - public: - LeakyBucket() : level_bits_(0) {} - - // Updates bucket level and returns its current level in bits. Data is remove - // from bucket with rate equal to target bitrate of previous frame. Bucket - // level is tracked with floating point precision. Returned value of bucket - // level is rounded up. - int Update(const Frame& frame) { - RTC_CHECK(frame.target_bitrate) << "Bitrate must be specified."; - - if (prev_frame_) { - RTC_CHECK_GT(frame.timestamp_rtp, prev_frame_->timestamp_rtp) - << "Timestamp must increase."; - TimeDelta passed = - (frame.timestamp_rtp - prev_frame_->timestamp_rtp) / k90kHz; - level_bits_ -= - prev_frame_->target_bitrate->bps() * passed.us() / 1000000.0; - level_bits_ = std::max(level_bits_, 0.0); - } - - prev_frame_ = frame; - - level_bits_ += frame.frame_size.bytes() * 8; - return static_cast<int>(std::ceil(level_bits_)); - } - - private: - absl::optional<Frame> prev_frame_; - double level_bits_; -}; - -// Merges spatial layer frames into superframes. -std::vector<Frame> Merge(const std::vector<Frame>& frames) { - std::vector<Frame> superframes; - // Map from frame timestamp to index in `superframes` vector. - std::map<uint32_t, int> index; - - for (const auto& f : frames) { - if (index.find(f.timestamp_rtp) == index.end()) { - index[f.timestamp_rtp] = static_cast<int>(superframes.size()); - superframes.push_back(f); - continue; - } - - Frame& sf = superframes[index[f.timestamp_rtp]]; - - sf.width = std::max(sf.width, f.width); - sf.height = std::max(sf.height, f.height); - sf.frame_size += f.frame_size; - sf.keyframe |= f.keyframe; - - sf.encode_time = std::max(sf.encode_time, f.encode_time); - sf.decode_time = std::max(sf.decode_time, f.decode_time); - - if (f.spatial_idx > sf.spatial_idx) { - if (f.qp) { - sf.qp = f.qp; - } - if (f.psnr) { - sf.psnr = f.psnr; - } - } - - sf.spatial_idx = std::max(sf.spatial_idx, f.spatial_idx); - sf.temporal_idx = std::max(sf.temporal_idx, f.temporal_idx); - - sf.encoded |= f.encoded; - sf.decoded |= f.decoded; - } - - return superframes; -} - -Timestamp RtpToTime(uint32_t timestamp_rtp) { - return Timestamp::Micros((timestamp_rtp / k90kHz).us()); -} - -SamplesStatsCounter::StatsSample StatsSample(double value, Timestamp time) { - return SamplesStatsCounter::StatsSample{value, time}; -} - -TimeDelta CalcTotalDuration(const std::vector<Frame>& frames) { - RTC_CHECK(!frames.empty()); - TimeDelta duration = TimeDelta::Zero(); - if (frames.size() > 1) { - duration += - (frames.rbegin()->timestamp_rtp - frames.begin()->timestamp_rtp) / - k90kHz; - } - - // Add last frame duration. If target frame rate is provided, calculate frame - // duration from it. Otherwise, assume duration of last frame is the same as - // duration of preceding frame. - if (frames.rbegin()->target_framerate) { - duration += 1 / *frames.rbegin()->target_framerate; - } else { - RTC_CHECK_GT(frames.size(), 1u); - duration += (frames.rbegin()->timestamp_rtp - - std::next(frames.rbegin())->timestamp_rtp) / - k90kHz; - } - - return duration; -} -} // namespace - -std::vector<Frame> VideoCodecStatsImpl::Slice( - absl::optional<Filter> filter) const { - std::vector<Frame> frames; - for (const auto& [frame_id, f] : frames_) { - if (filter.has_value()) { - if (filter->first_frame.has_value() && - f.frame_num < *filter->first_frame) { - continue; - } - if (filter->last_frame.has_value() && f.frame_num > *filter->last_frame) { - continue; - } - if (filter->spatial_idx.has_value() && - f.spatial_idx != *filter->spatial_idx) { - continue; - } - if (filter->temporal_idx.has_value() && - f.temporal_idx > *filter->temporal_idx) { - continue; - } - } - frames.push_back(f); - } - return frames; -} - -Stream VideoCodecStatsImpl::Aggregate(const std::vector<Frame>& frames) const { - std::vector<Frame> superframes = Merge(frames); - RTC_CHECK(!superframes.empty()); - - LeakyBucket leacky_bucket; - Stream stream; - for (size_t i = 0; i < superframes.size(); ++i) { - Frame& f = superframes[i]; - Timestamp time = RtpToTime(f.timestamp_rtp); - - if (!f.frame_size.IsZero()) { - stream.width.AddSample(StatsSample(f.width, time)); - stream.height.AddSample(StatsSample(f.height, time)); - stream.frame_size_bytes.AddSample( - StatsSample(f.frame_size.bytes(), time)); - stream.keyframe.AddSample(StatsSample(f.keyframe, time)); - if (f.qp) { - stream.qp.AddSample(StatsSample(*f.qp, time)); - } - } - - if (f.encoded) { - stream.encode_time_ms.AddSample(StatsSample(f.encode_time.ms(), time)); - } - - if (f.decoded) { - stream.decode_time_ms.AddSample(StatsSample(f.decode_time.ms(), time)); - } - - if (f.psnr) { - stream.psnr.y.AddSample(StatsSample(f.psnr->y, time)); - stream.psnr.u.AddSample(StatsSample(f.psnr->u, time)); - stream.psnr.v.AddSample(StatsSample(f.psnr->v, time)); - } - - if (f.target_framerate) { - stream.target_framerate_fps.AddSample( - StatsSample(f.target_framerate->millihertz() / 1000.0, time)); - } - - if (f.target_bitrate) { - stream.target_bitrate_kbps.AddSample( - StatsSample(f.target_bitrate->bps() / 1000.0, time)); - - int buffer_level_bits = leacky_bucket.Update(f); - stream.transmission_time_ms.AddSample( - StatsSample(buffer_level_bits * rtc::kNumMillisecsPerSec / - f.target_bitrate->bps(), - RtpToTime(f.timestamp_rtp))); - } - } - - TimeDelta duration = CalcTotalDuration(superframes); - DataRate encoded_bitrate = - DataSize::Bytes(stream.frame_size_bytes.GetSum()) / duration; - - int num_encoded_frames = stream.frame_size_bytes.NumSamples(); - Frequency encoded_framerate = num_encoded_frames / duration; - - absl::optional<double> bitrate_mismatch_pct; - if (auto target_bitrate = superframes.begin()->target_bitrate; - target_bitrate) { - bitrate_mismatch_pct = 100.0 * - (encoded_bitrate.bps() - target_bitrate->bps()) / - target_bitrate->bps(); - } - - absl::optional<double> framerate_mismatch_pct; - if (auto target_framerate = superframes.begin()->target_framerate; - target_framerate) { - framerate_mismatch_pct = - 100.0 * - (encoded_framerate.millihertz() - target_framerate->millihertz()) / - target_framerate->millihertz(); - } - - for (auto& f : superframes) { - Timestamp time = RtpToTime(f.timestamp_rtp); - stream.encoded_bitrate_kbps.AddSample( - StatsSample(encoded_bitrate.bps() / 1000.0, time)); - - stream.encoded_framerate_fps.AddSample( - StatsSample(encoded_framerate.millihertz() / 1000.0, time)); - - if (bitrate_mismatch_pct) { - stream.bitrate_mismatch_pct.AddSample( - StatsSample(*bitrate_mismatch_pct, time)); - } - - if (framerate_mismatch_pct) { - stream.framerate_mismatch_pct.AddSample( - StatsSample(*framerate_mismatch_pct, time)); - } - } - - return stream; -} - -void VideoCodecStatsImpl::AddFrame(const Frame& frame) { - FrameId frame_id{.timestamp_rtp = frame.timestamp_rtp, - .spatial_idx = frame.spatial_idx}; - RTC_CHECK(frames_.find(frame_id) == frames_.end()) - << "Frame with timestamp_rtp=" << frame.timestamp_rtp - << " and spatial_idx=" << frame.spatial_idx << " already exists"; - - frames_[frame_id] = frame; -} - -Frame* VideoCodecStatsImpl::GetFrame(uint32_t timestamp_rtp, int spatial_idx) { - FrameId frame_id{.timestamp_rtp = timestamp_rtp, .spatial_idx = spatial_idx}; - if (frames_.find(frame_id) == frames_.end()) { - return nullptr; - } - return &frames_.find(frame_id)->second; -} - -} // namespace test -} // namespace webrtc diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl.h b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl.h deleted file mode 100644 index 77471d2ecd..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl.h +++ /dev/null @@ -1,62 +0,0 @@ -/* - * Copyright (c) 2023 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_STATS_IMPL_H_ -#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_STATS_IMPL_H_ - -#include <map> -#include <string> -#include <vector> - -#include "absl/types/optional.h" -#include "api/test/video_codec_stats.h" - -namespace webrtc { -namespace test { - -// Implementation of `VideoCodecStats`. This class is not thread-safe. -class VideoCodecStatsImpl : public VideoCodecStats { - public: - std::vector<Frame> Slice( - absl::optional<Filter> filter = absl::nullopt) const override; - - Stream Aggregate(const std::vector<Frame>& frames) const override; - - void AddFrame(const Frame& frame); - - // Returns raw pointers to previously added frame. If frame does not exist, - // returns `nullptr`. - Frame* GetFrame(uint32_t timestamp_rtp, int spatial_idx); - - private: - struct FrameId { - uint32_t timestamp_rtp; - int spatial_idx; - - bool operator==(const FrameId& o) const { - return timestamp_rtp == o.timestamp_rtp && spatial_idx == o.spatial_idx; - } - - bool operator<(const FrameId& o) const { - if (timestamp_rtp < o.timestamp_rtp) - return true; - if (timestamp_rtp == o.timestamp_rtp && spatial_idx < o.spatial_idx) - return true; - return false; - } - }; - - std::map<FrameId, Frame> frames_; -}; - -} // namespace test -} // namespace webrtc - -#endif // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_STATS_IMPL_H_ diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl_unittest.cc deleted file mode 100644 index ce11d5abe6..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_stats_impl_unittest.cc +++ /dev/null @@ -1,148 +0,0 @@ -/* - * Copyright (c) 2023 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/video_coding/codecs/test/video_codec_stats_impl.h" - -#include <tuple> - -#include "absl/types/optional.h" -#include "test/gmock.h" -#include "test/gtest.h" - -namespace webrtc { -namespace test { - -namespace { -using ::testing::Return; -using ::testing::Values; -using Filter = VideoCodecStats::Filter; -using Frame = VideoCodecStatsImpl::Frame; -using Stream = VideoCodecStats::Stream; -} // namespace - -TEST(VideoCodecStatsImpl, AddAndGetFrame) { - VideoCodecStatsImpl stats; - stats.AddFrame({.timestamp_rtp = 0, .spatial_idx = 0}); - stats.AddFrame({.timestamp_rtp = 0, .spatial_idx = 1}); - stats.AddFrame({.timestamp_rtp = 1, .spatial_idx = 0}); - - Frame* fs = stats.GetFrame(/*timestamp_rtp=*/0, /*spatial_idx=*/0); - ASSERT_NE(fs, nullptr); - EXPECT_EQ(fs->timestamp_rtp, 0u); - EXPECT_EQ(fs->spatial_idx, 0); - - fs = stats.GetFrame(/*timestamp_rtp=*/0, /*spatial_idx=*/1); - ASSERT_NE(fs, nullptr); - EXPECT_EQ(fs->timestamp_rtp, 0u); - EXPECT_EQ(fs->spatial_idx, 1); - - fs = stats.GetFrame(/*timestamp_rtp=*/1, /*spatial_idx=*/0); - ASSERT_NE(fs, nullptr); - EXPECT_EQ(fs->timestamp_rtp, 1u); - EXPECT_EQ(fs->spatial_idx, 0); - - fs = stats.GetFrame(/*timestamp_rtp=*/1, /*spatial_idx=*/1); - EXPECT_EQ(fs, nullptr); -} - -class VideoCodecStatsImplSlicingTest - : public ::testing::TestWithParam<std::tuple<Filter, std::vector<int>>> {}; - -TEST_P(VideoCodecStatsImplSlicingTest, Slice) { - Filter filter = std::get<0>(GetParam()); - std::vector<int> expected_frames = std::get<1>(GetParam()); - std::vector<VideoCodecStats::Frame> frames = { - {.frame_num = 0, .timestamp_rtp = 0, .spatial_idx = 0, .temporal_idx = 0}, - {.frame_num = 0, .timestamp_rtp = 0, .spatial_idx = 1, .temporal_idx = 0}, - {.frame_num = 1, .timestamp_rtp = 1, .spatial_idx = 0, .temporal_idx = 1}, - {.frame_num = 1, - .timestamp_rtp = 1, - .spatial_idx = 1, - .temporal_idx = 1}}; - - VideoCodecStatsImpl stats; - stats.AddFrame(frames[0]); - stats.AddFrame(frames[1]); - stats.AddFrame(frames[2]); - stats.AddFrame(frames[3]); - - std::vector<VideoCodecStats::Frame> slice = stats.Slice(filter); - ASSERT_EQ(slice.size(), expected_frames.size()); - for (size_t i = 0; i < expected_frames.size(); ++i) { - Frame& expected = frames[expected_frames[i]]; - EXPECT_EQ(slice[i].frame_num, expected.frame_num); - EXPECT_EQ(slice[i].timestamp_rtp, expected.timestamp_rtp); - EXPECT_EQ(slice[i].spatial_idx, expected.spatial_idx); - EXPECT_EQ(slice[i].temporal_idx, expected.temporal_idx); - } -} - -INSTANTIATE_TEST_SUITE_P( - All, - VideoCodecStatsImplSlicingTest, - ::testing::Values( - std::make_tuple(Filter{}, std::vector<int>{0, 1, 2, 3}), - std::make_tuple(Filter{.first_frame = 1}, std::vector<int>{2, 3}), - std::make_tuple(Filter{.last_frame = 0}, std::vector<int>{0, 1}), - std::make_tuple(Filter{.spatial_idx = 0}, std::vector<int>{0, 2}), - std::make_tuple(Filter{.temporal_idx = 1}, - std::vector<int>{0, 1, 2, 3}))); - -TEST(VideoCodecStatsImpl, AggregateBitrate) { - std::vector<VideoCodecStats::Frame> frames = { - {.frame_num = 0, - .timestamp_rtp = 0, - .frame_size = DataSize::Bytes(1000), - .target_bitrate = DataRate::BytesPerSec(1000)}, - {.frame_num = 1, - .timestamp_rtp = 90000, - .frame_size = DataSize::Bytes(2000), - .target_bitrate = DataRate::BytesPerSec(1000)}}; - - Stream stream = VideoCodecStatsImpl().Aggregate(frames); - EXPECT_EQ(stream.encoded_bitrate_kbps.GetAverage(), 12.0); - EXPECT_EQ(stream.bitrate_mismatch_pct.GetAverage(), 50.0); -} - -TEST(VideoCodecStatsImpl, AggregateFramerate) { - std::vector<VideoCodecStats::Frame> frames = { - {.frame_num = 0, - .timestamp_rtp = 0, - .frame_size = DataSize::Bytes(1), - .target_framerate = Frequency::Hertz(1)}, - {.frame_num = 1, - .timestamp_rtp = 90000, - .frame_size = DataSize::Zero(), - .target_framerate = Frequency::Hertz(1)}}; - - Stream stream = VideoCodecStatsImpl().Aggregate(frames); - EXPECT_EQ(stream.encoded_framerate_fps.GetAverage(), 0.5); - EXPECT_EQ(stream.framerate_mismatch_pct.GetAverage(), -50.0); -} - -TEST(VideoCodecStatsImpl, AggregateTransmissionTime) { - std::vector<VideoCodecStats::Frame> frames = { - {.frame_num = 0, - .timestamp_rtp = 0, - .frame_size = DataSize::Bytes(2), - .target_bitrate = DataRate::BytesPerSec(1)}, - {.frame_num = 1, - .timestamp_rtp = 90000, - .frame_size = DataSize::Bytes(3), - .target_bitrate = DataRate::BytesPerSec(1)}}; - - Stream stream = VideoCodecStatsImpl().Aggregate(frames); - ASSERT_EQ(stream.transmission_time_ms.NumSamples(), 2); - ASSERT_EQ(stream.transmission_time_ms.GetSamples()[0], 2000); - ASSERT_EQ(stream.transmission_time_ms.GetSamples()[1], 4000); -} - -} // namespace test -} // namespace webrtc diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc index 1c8fe97e84..60c2fcbb6e 100644 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc +++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc @@ -8,41 +8,62 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "api/video_codecs/video_codec.h" - -#include <cstddef> #include <memory> #include <string> #include <vector> #include "absl/flags/flag.h" #include "absl/functional/any_invocable.h" -#include "api/test/create_video_codec_tester.h" #include "api/test/metrics/global_metrics_logger_and_exporter.h" -#include "api/test/video_codec_tester.h" -#include "api/test/videocodec_test_stats.h" #include "api/units/data_rate.h" #include "api/units/frequency.h" -#include "api/video/encoded_image.h" -#include "api/video/i420_buffer.h" #include "api/video/resolution.h" -#include "api/video/video_frame.h" -#include "api/video_codecs/scalability_mode.h" -#include "api/video_codecs/video_decoder.h" -#include "api/video_codecs/video_encoder.h" -#include "media/engine/internal_decoder_factory.h" -#include "media/engine/internal_encoder_factory.h" -#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "modules/video_coding/include/video_error_codes.h" -#include "modules/video_coding/svc/scalability_mode_util.h" +#include "api/video_codecs/builtin_video_decoder_factory.h" +#include "api/video_codecs/builtin_video_encoder_factory.h" #if defined(WEBRTC_ANDROID) #include "modules/video_coding/codecs/test/android_codec_factory_helper.h" #endif +#include "modules/video_coding/svc/scalability_mode_util.h" #include "rtc_base/logging.h" +#include "rtc_base/strings/string_builder.h" #include "test/gtest.h" #include "test/test_flags.h" #include "test/testsupport/file_utils.h" -#include "test/testsupport/frame_reader.h" +#include "test/video_codec_tester.h" + +ABSL_FLAG(std::string, + video_name, + "FourPeople_1280x720_30", + "Name of input video sequence."); +ABSL_FLAG(std::string, + encoder, + "libaom-av1", + "Encoder: libaom-av1, libvpx-vp9, libvpx-vp8, openh264, hw-vp8, " + "hw-vp9, hw-av1, hw-h264, hw-h265"); +ABSL_FLAG(std::string, + decoder, + "dav1d", + "Decoder: dav1d, libvpx-vp9, libvpx-vp8, ffmpeg-h264, hw-vp8, " + "hw-vp9, hw-av1, hw-h264, hw-h265"); +ABSL_FLAG(std::string, scalability_mode, "L1T1", "Scalability mode."); +ABSL_FLAG(int, width, 1280, "Width."); +ABSL_FLAG(int, height, 720, "Height."); +ABSL_FLAG(std::vector<std::string>, + bitrate_kbps, + {"1024"}, + "Encode target bitrate per layer (l0t0,l0t1,...l1t0,l1t1 and so on) " + "in kbps."); +ABSL_FLAG(double, + framerate_fps, + 30.0, + "Encode target frame rate of the top temporal layer in fps."); +ABSL_FLAG(int, num_frames, 300, "Number of frames to encode and/or decode."); +ABSL_FLAG(std::string, test_name, "", "Test name."); +ABSL_FLAG(bool, dump_decoder_input, false, "Dump decoder input."); +ABSL_FLAG(bool, dump_decoder_output, false, "Dump decoder output."); +ABSL_FLAG(bool, dump_encoder_input, false, "Dump encoder input."); +ABSL_FLAG(bool, dump_encoder_output, false, "Dump encoder output."); +ABSL_FLAG(bool, write_csv, false, "Write metrics to a CSV file."); namespace webrtc { namespace test { @@ -50,6 +71,10 @@ namespace test { namespace { using ::testing::Combine; using ::testing::Values; +using VideoSourceSettings = VideoCodecTester::VideoSourceSettings; +using EncodingSettings = VideoCodecTester::EncodingSettings; +using VideoCodecStats = VideoCodecTester::VideoCodecStats; +using Filter = VideoCodecStats::Filter; using PacingMode = VideoCodecTester::PacingSettings::PacingMode; struct VideoInfo { @@ -58,405 +83,93 @@ struct VideoInfo { Frequency framerate; }; -struct LayerId { - int spatial_idx; - int temporal_idx; - - bool operator==(const LayerId& o) const { - return spatial_idx == o.spatial_idx && temporal_idx == o.temporal_idx; - } - - bool operator<(const LayerId& o) const { - if (spatial_idx < o.spatial_idx) - return true; - if (spatial_idx == o.spatial_idx && temporal_idx < o.temporal_idx) - return true; - return false; - } -}; - -struct EncodingSettings { - ScalabilityMode scalability_mode; - struct LayerSettings { - Resolution resolution; - Frequency framerate; - DataRate bitrate; - }; - std::map<LayerId, LayerSettings> layer_settings; - - bool IsSameSettings(const EncodingSettings& other) const { - if (scalability_mode != other.scalability_mode) { - return false; - } - - for (auto [layer_id, layer] : layer_settings) { - const auto& other_layer = other.layer_settings.at(layer_id); - if (layer.resolution != other_layer.resolution) { - return false; - } - } - - return true; - } - - bool IsSameRate(const EncodingSettings& other) const { - for (auto [layer_id, layer] : layer_settings) { - const auto& other_layer = other.layer_settings.at(layer_id); - if (layer.bitrate != other_layer.bitrate || - layer.framerate != other_layer.framerate) { - return false; - } - } - - return true; - } -}; - -const VideoInfo kFourPeople_1280x720_30 = { - .name = "FourPeople_1280x720_30", - .resolution = {.width = 1280, .height = 720}, - .framerate = Frequency::Hertz(30)}; - -class TestRawVideoSource : public VideoCodecTester::RawVideoSource { - public: - static constexpr Frequency k90kHz = Frequency::Hertz(90000); - - TestRawVideoSource(VideoInfo video_info, - const std::map<int, EncodingSettings>& frame_settings, - int num_frames) - : video_info_(video_info), - frame_settings_(frame_settings), - num_frames_(num_frames), - frame_num_(0), - // Start with non-zero timestamp to force using frame RTP timestamps in - // IvfFrameWriter. - timestamp_rtp_(90000) { - // Ensure settings for the first frame are provided. - RTC_CHECK_GT(frame_settings_.size(), 0u); - RTC_CHECK_EQ(frame_settings_.begin()->first, 0); - - frame_reader_ = CreateYuvFrameReader( - ResourcePath(video_info_.name, "yuv"), video_info_.resolution, - YuvFrameReaderImpl::RepeatMode::kPingPong); - RTC_CHECK(frame_reader_); - } - - // Pulls next frame. Frame RTP timestamp is set accordingly to - // `EncodingSettings::framerate`. - absl::optional<VideoFrame> PullFrame() override { - if (frame_num_ >= num_frames_) { - return absl::nullopt; // End of stream. - } - - const EncodingSettings& encoding_settings = - std::prev(frame_settings_.upper_bound(frame_num_))->second; - - Resolution resolution = - encoding_settings.layer_settings.begin()->second.resolution; - Frequency framerate = - encoding_settings.layer_settings.begin()->second.framerate; - - int pulled_frame; - auto buffer = frame_reader_->PullFrame( - &pulled_frame, resolution, - {.num = static_cast<int>(framerate.millihertz()), - .den = static_cast<int>(video_info_.framerate.millihertz())}); - RTC_CHECK(buffer) << "Cannot pull frame " << frame_num_; - - auto frame = VideoFrame::Builder() - .set_video_frame_buffer(buffer) - .set_timestamp_rtp(timestamp_rtp_) - .set_timestamp_us((timestamp_rtp_ / k90kHz).us()) - .build(); - - pulled_frames_[timestamp_rtp_] = pulled_frame; - timestamp_rtp_ += k90kHz / framerate; - ++frame_num_; - - return frame; - } - - // Reads frame specified by `timestamp_rtp`, scales it to `resolution` and - // returns. Frame with the given `timestamp_rtp` is expected to be pulled - // before. - VideoFrame GetFrame(uint32_t timestamp_rtp, Resolution resolution) override { - RTC_CHECK(pulled_frames_.find(timestamp_rtp) != pulled_frames_.end()) - << "Frame with RTP timestamp " << timestamp_rtp - << " was not pulled before"; - auto buffer = - frame_reader_->ReadFrame(pulled_frames_[timestamp_rtp], resolution); - return VideoFrame::Builder() - .set_video_frame_buffer(buffer) - .set_timestamp_rtp(timestamp_rtp) - .build(); - } - - protected: - VideoInfo video_info_; - std::unique_ptr<FrameReader> frame_reader_; - const std::map<int, EncodingSettings>& frame_settings_; - int num_frames_; - int frame_num_; - uint32_t timestamp_rtp_; - std::map<uint32_t, int> pulled_frames_; -}; - -class TestEncoder : public VideoCodecTester::Encoder, - public EncodedImageCallback { - public: - TestEncoder(std::unique_ptr<VideoEncoder> encoder, - const std::string codec_type, - const std::map<int, EncodingSettings>& frame_settings) - : encoder_(std::move(encoder)), - codec_type_(codec_type), - frame_settings_(frame_settings), - frame_num_(0) { - // Ensure settings for the first frame is provided. - RTC_CHECK_GT(frame_settings_.size(), 0u); - RTC_CHECK_EQ(frame_settings_.begin()->first, 0); - - encoder_->RegisterEncodeCompleteCallback(this); - } - - void Initialize() override { - const EncodingSettings& first_frame_settings = frame_settings_.at(0); - Configure(first_frame_settings); - SetRates(first_frame_settings); - } - - void Encode(const VideoFrame& frame, EncodeCallback callback) override { - { - MutexLock lock(&mutex_); - callbacks_[frame.timestamp()] = std::move(callback); - } - - if (auto fs = frame_settings_.find(frame_num_); - fs != frame_settings_.begin() && fs != frame_settings_.end()) { - if (!fs->second.IsSameSettings(std::prev(fs)->second)) { - Configure(fs->second); - } else if (!fs->second.IsSameRate(std::prev(fs)->second)) { - SetRates(fs->second); - } - } - - encoder_->Encode(frame, nullptr); - ++frame_num_; - } - - void Flush() override { - // TODO(webrtc:14852): For codecs which buffer frames we need a to - // flush them to get last frames. Add such functionality to VideoEncoder - // API. On Android it will map directly to `MediaCodec.flush()`. - encoder_->Release(); - } - - VideoEncoder* encoder() { return encoder_.get(); } - - protected: - Result OnEncodedImage(const EncodedImage& encoded_image, - const CodecSpecificInfo* codec_specific_info) override { - MutexLock lock(&mutex_); - auto cb = callbacks_.find(encoded_image.RtpTimestamp()); - RTC_CHECK(cb != callbacks_.end()); - cb->second(encoded_image); - - callbacks_.erase(callbacks_.begin(), cb); - return Result(Result::Error::OK); - } - - void Configure(const EncodingSettings& es) { - VideoCodec vc; - const EncodingSettings::LayerSettings& layer_settings = - es.layer_settings.begin()->second; - vc.width = layer_settings.resolution.width; - vc.height = layer_settings.resolution.height; - const DataRate& bitrate = layer_settings.bitrate; - vc.startBitrate = bitrate.kbps(); - vc.maxBitrate = bitrate.kbps(); - vc.minBitrate = 0; - vc.maxFramerate = static_cast<uint32_t>(layer_settings.framerate.hertz()); - vc.active = true; - vc.qpMax = 63; - vc.numberOfSimulcastStreams = 0; - vc.mode = webrtc::VideoCodecMode::kRealtimeVideo; - vc.SetFrameDropEnabled(true); - vc.SetScalabilityMode(es.scalability_mode); - - vc.codecType = PayloadStringToCodecType(codec_type_); - if (vc.codecType == kVideoCodecVP8) { - *(vc.VP8()) = VideoEncoder::GetDefaultVp8Settings(); - } else if (vc.codecType == kVideoCodecVP9) { - *(vc.VP9()) = VideoEncoder::GetDefaultVp9Settings(); - } else if (vc.codecType == kVideoCodecH264) { - *(vc.H264()) = VideoEncoder::GetDefaultH264Settings(); - } - - VideoEncoder::Settings ves( - VideoEncoder::Capabilities(/*loss_notification=*/false), - /*number_of_cores=*/1, - /*max_payload_size=*/1440); - - int result = encoder_->InitEncode(&vc, ves); - ASSERT_EQ(result, WEBRTC_VIDEO_CODEC_OK); - - SetRates(es); - } - - void SetRates(const EncodingSettings& es) { - VideoEncoder::RateControlParameters rc; - int num_spatial_layers = - ScalabilityModeToNumSpatialLayers(es.scalability_mode); - int num_temporal_layers = - ScalabilityModeToNumSpatialLayers(es.scalability_mode); - for (int sidx = 0; sidx < num_spatial_layers; ++sidx) { - for (int tidx = 0; tidx < num_temporal_layers; ++tidx) { - auto layer_settings = - es.layer_settings.find({.spatial_idx = sidx, .temporal_idx = tidx}); - RTC_CHECK(layer_settings != es.layer_settings.end()) - << "Bitrate for layer S=" << sidx << " T=" << tidx << " is not set"; - rc.bitrate.SetBitrate(sidx, tidx, layer_settings->second.bitrate.bps()); - } - } - - rc.framerate_fps = - es.layer_settings.begin()->second.framerate.millihertz() / 1000.0; - encoder_->SetRates(rc); - } - - std::unique_ptr<VideoEncoder> encoder_; - const std::string codec_type_; - const std::map<int, EncodingSettings>& frame_settings_; - int frame_num_; - std::map<uint32_t, EncodeCallback> callbacks_ RTC_GUARDED_BY(mutex_); - Mutex mutex_; -}; - -class TestDecoder : public VideoCodecTester::Decoder, - public DecodedImageCallback { - public: - TestDecoder(std::unique_ptr<VideoDecoder> decoder, - const std::string codec_type) - : decoder_(std::move(decoder)), codec_type_(codec_type) { - decoder_->RegisterDecodeCompleteCallback(this); - } - - void Initialize() override { - VideoDecoder::Settings ds; - ds.set_codec_type(PayloadStringToCodecType(codec_type_)); - ds.set_number_of_cores(1); - ds.set_max_render_resolution({1280, 720}); - - bool result = decoder_->Configure(ds); - ASSERT_TRUE(result); - } - - void Decode(const EncodedImage& frame, DecodeCallback callback) override { - { - MutexLock lock(&mutex_); - callbacks_[frame.RtpTimestamp()] = std::move(callback); - } - - decoder_->Decode(frame, /*render_time_ms=*/0); - } - - void Flush() override { - // TODO(webrtc:14852): For codecs which buffer frames we need a to - // flush them to get last frames. Add such functionality to VideoDecoder - // API. On Android it will map directly to `MediaCodec.flush()`. - decoder_->Release(); - } - - VideoDecoder* decoder() { return decoder_.get(); } - - protected: - int Decoded(VideoFrame& decoded_frame) override { - MutexLock lock(&mutex_); - auto cb = callbacks_.find(decoded_frame.timestamp()); - RTC_CHECK(cb != callbacks_.end()); - cb->second(decoded_frame); +const std::map<std::string, VideoInfo> kRawVideos = { + {"FourPeople_1280x720_30", + {.name = "FourPeople_1280x720_30", + .resolution = {.width = 1280, .height = 720}, + .framerate = Frequency::Hertz(30)}}, + {"vidyo1_1280x720_30", + {.name = "vidyo1_1280x720_30", + .resolution = {.width = 1280, .height = 720}, + .framerate = Frequency::Hertz(30)}}, + {"vidyo4_1280x720_30", + {.name = "vidyo4_1280x720_30", + .resolution = {.width = 1280, .height = 720}, + .framerate = Frequency::Hertz(30)}}, + {"KristenAndSara_1280x720_30", + {.name = "KristenAndSara_1280x720_30", + .resolution = {.width = 1280, .height = 720}, + .framerate = Frequency::Hertz(30)}}, + {"Johnny_1280x720_30", + {.name = "Johnny_1280x720_30", + .resolution = {.width = 1280, .height = 720}, + .framerate = Frequency::Hertz(30)}}}; + +static constexpr Frequency k90kHz = Frequency::Hertz(90000); + +std::string CodecNameToCodecType(std::string name) { + if (name.find("av1") != std::string::npos) { + return "AV1"; + } + if (name.find("vp9") != std::string::npos) { + return "VP9"; + } + if (name.find("vp8") != std::string::npos) { + return "VP8"; + } + if (name.find("h264") != std::string::npos) { + return "H264"; + } + if (name.find("h265") != std::string::npos) { + return "H265"; + } + RTC_CHECK_NOTREACHED(); +} - callbacks_.erase(callbacks_.begin(), cb); - return WEBRTC_VIDEO_CODEC_OK; +// TODO(webrtc:14852): Make Create[Encoder,Decoder]Factory to work with codec +// name directly. +std::string CodecNameToCodecImpl(std::string name) { + if (name.find("hw") != std::string::npos) { + return "mediacodec"; } - - std::unique_ptr<VideoDecoder> decoder_; - const std::string codec_type_; - std::map<uint32_t, DecodeCallback> callbacks_ RTC_GUARDED_BY(mutex_); - Mutex mutex_; -}; - -std::unique_ptr<TestRawVideoSource> CreateVideoSource( - const VideoInfo& video, - const std::map<int, EncodingSettings>& frame_settings, - int num_frames) { - return std::make_unique<TestRawVideoSource>(video, frame_settings, - num_frames); + return "builtin"; } -std::unique_ptr<TestEncoder> CreateEncoder( - std::string type, - std::string impl, - const std::map<int, EncodingSettings>& frame_settings) { - std::unique_ptr<VideoEncoderFactory> factory; +std::unique_ptr<VideoEncoderFactory> CreateEncoderFactory(std::string impl) { if (impl == "builtin") { - factory = std::make_unique<InternalEncoderFactory>(); - } else if (impl == "mediacodec") { + return CreateBuiltinVideoEncoderFactory(); + } #if defined(WEBRTC_ANDROID) - InitializeAndroidObjects(); - factory = CreateAndroidEncoderFactory(); + InitializeAndroidObjects(); + return CreateAndroidEncoderFactory(); +#else + return nullptr; #endif - } - std::unique_ptr<VideoEncoder> encoder = - factory->CreateVideoEncoder(SdpVideoFormat(type)); - if (encoder == nullptr) { - return nullptr; - } - return std::make_unique<TestEncoder>(std::move(encoder), type, - frame_settings); } -std::unique_ptr<TestDecoder> CreateDecoder(std::string type, std::string impl) { - std::unique_ptr<VideoDecoderFactory> factory; +std::unique_ptr<VideoDecoderFactory> CreateDecoderFactory(std::string impl) { if (impl == "builtin") { - factory = std::make_unique<InternalDecoderFactory>(); - } else if (impl == "mediacodec") { + return CreateBuiltinVideoDecoderFactory(); + } #if defined(WEBRTC_ANDROID) - InitializeAndroidObjects(); - factory = CreateAndroidDecoderFactory(); + InitializeAndroidObjects(); + return CreateAndroidDecoderFactory(); +#else + return nullptr; #endif - } - std::unique_ptr<VideoDecoder> decoder = - factory->CreateVideoDecoder(SdpVideoFormat(type)); - if (decoder == nullptr) { - return nullptr; - } - return std::make_unique<TestDecoder>(std::move(decoder), type); } -void SetTargetRates(const std::map<int, EncodingSettings>& frame_settings, - std::vector<VideoCodecStats::Frame>& frames) { - for (VideoCodecStats::Frame& f : frames) { - const EncodingSettings& encoding_settings = - std::prev(frame_settings.upper_bound(f.frame_num))->second; - LayerId layer_id = {.spatial_idx = f.spatial_idx, - .temporal_idx = f.temporal_idx}; - RTC_CHECK(encoding_settings.layer_settings.find(layer_id) != - encoding_settings.layer_settings.end()) - << "Frame frame_num=" << f.frame_num - << " belongs to spatial_idx=" << f.spatial_idx - << " temporal_idx=" << f.temporal_idx - << " but settings for this layer are not provided."; - const EncodingSettings::LayerSettings& layer_settings = - encoding_settings.layer_settings.at(layer_id); - f.target_bitrate = layer_settings.bitrate; - f.target_framerate = layer_settings.framerate; +std::string TestName() { + std::string test_name = absl::GetFlag(FLAGS_test_name); + if (!test_name.empty()) { + return test_name; } + return ::testing::UnitTest::GetInstance()->current_test_info()->name(); } std::string TestOutputPath() { std::string output_path = - OutputPath() + - ::testing::UnitTest::GetInstance()->current_test_info()->name(); + (rtc::StringBuilder() << OutputPath() << TestName()).str(); std::string output_dir = DirName(output_path); bool result = CreateDir(output_dir); RTC_CHECK(result) << "Cannot create " << output_dir; @@ -465,116 +178,120 @@ std::string TestOutputPath() { } // namespace std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest( - std::string codec_type, std::string codec_impl, const VideoInfo& video_info, - const std::map<int, EncodingSettings>& frame_settings, - int num_frames, - bool save_codec_input, - bool save_codec_output) { - std::unique_ptr<TestRawVideoSource> video_source = - CreateVideoSource(video_info, frame_settings, num_frames); - - std::unique_ptr<TestEncoder> encoder = - CreateEncoder(codec_type, codec_impl, frame_settings); - if (encoder == nullptr) { + const std::map<uint32_t, EncodingSettings>& encoding_settings) { + VideoSourceSettings source_settings{ + .file_path = ResourcePath(video_info.name, "yuv"), + .resolution = video_info.resolution, + .framerate = video_info.framerate}; + + const SdpVideoFormat& sdp_video_format = + encoding_settings.begin()->second.sdp_video_format; + + std::unique_ptr<VideoEncoderFactory> encoder_factory = + CreateEncoderFactory(codec_impl); + if (!encoder_factory + ->QueryCodecSupport(sdp_video_format, + /*scalability_mode=*/absl::nullopt) + .is_supported) { + RTC_LOG(LS_WARNING) << "No encoder for video format " + << sdp_video_format.ToString(); return nullptr; } - std::unique_ptr<TestDecoder> decoder = CreateDecoder(codec_type, codec_impl); - if (decoder == nullptr) { - // If platform decoder is not available try built-in one. - if (codec_impl == "builtin") { - return nullptr; - } - - decoder = CreateDecoder(codec_type, "builtin"); - if (decoder == nullptr) { + std::unique_ptr<VideoDecoderFactory> decoder_factory = + CreateDecoderFactory(codec_impl); + if (!decoder_factory + ->QueryCodecSupport(sdp_video_format, + /*reference_scaling=*/false) + .is_supported) { + decoder_factory = CreateDecoderFactory("builtin"); + if (!decoder_factory + ->QueryCodecSupport(sdp_video_format, + /*reference_scaling=*/false) + .is_supported) { + RTC_LOG(LS_WARNING) << "No decoder for video format " + << sdp_video_format.ToString(); return nullptr; } } - RTC_LOG(LS_INFO) << "Encoder implementation: " - << encoder->encoder()->GetEncoderInfo().implementation_name; - RTC_LOG(LS_INFO) << "Decoder implementation: " - << decoder->decoder()->GetDecoderInfo().implementation_name; + std::string output_path = TestOutputPath(); VideoCodecTester::EncoderSettings encoder_settings; - encoder_settings.pacing.mode = - encoder->encoder()->GetEncoderInfo().is_hardware_accelerated - ? PacingMode::kRealTime - : PacingMode::kNoPacing; + encoder_settings.pacing_settings.mode = + codec_impl == "builtin" ? PacingMode::kNoPacing : PacingMode::kRealTime; + if (absl::GetFlag(FLAGS_dump_encoder_input)) { + encoder_settings.encoder_input_base_path = output_path + "_enc_input"; + } + if (absl::GetFlag(FLAGS_dump_encoder_output)) { + encoder_settings.encoder_output_base_path = output_path + "_enc_output"; + } VideoCodecTester::DecoderSettings decoder_settings; - decoder_settings.pacing.mode = - decoder->decoder()->GetDecoderInfo().is_hardware_accelerated - ? PacingMode::kRealTime - : PacingMode::kNoPacing; - - std::string output_path = TestOutputPath(); - if (save_codec_input) { - encoder_settings.encoder_input_base_path = output_path + "_enc_input"; + decoder_settings.pacing_settings.mode = + codec_impl == "builtin" ? PacingMode::kNoPacing : PacingMode::kRealTime; + if (absl::GetFlag(FLAGS_dump_decoder_input)) { decoder_settings.decoder_input_base_path = output_path + "_dec_input"; } - if (save_codec_output) { - encoder_settings.encoder_output_base_path = output_path + "_enc_output"; + if (absl::GetFlag(FLAGS_dump_decoder_output)) { decoder_settings.decoder_output_base_path = output_path + "_dec_output"; } - std::unique_ptr<VideoCodecTester> tester = CreateVideoCodecTester(); - return tester->RunEncodeDecodeTest(video_source.get(), encoder.get(), - decoder.get(), encoder_settings, - decoder_settings); + return VideoCodecTester::RunEncodeDecodeTest( + source_settings, encoder_factory.get(), decoder_factory.get(), + encoder_settings, decoder_settings, encoding_settings); } std::unique_ptr<VideoCodecStats> RunEncodeTest( std::string codec_type, std::string codec_impl, const VideoInfo& video_info, - const std::map<int, EncodingSettings>& frame_settings, - int num_frames, - bool save_codec_input, - bool save_codec_output) { - std::unique_ptr<TestRawVideoSource> video_source = - CreateVideoSource(video_info, frame_settings, num_frames); - - std::unique_ptr<TestEncoder> encoder = - CreateEncoder(codec_type, codec_impl, frame_settings); - if (encoder == nullptr) { + const std::map<uint32_t, EncodingSettings>& encoding_settings) { + VideoSourceSettings source_settings{ + .file_path = ResourcePath(video_info.name, "yuv"), + .resolution = video_info.resolution, + .framerate = video_info.framerate}; + + const SdpVideoFormat& sdp_video_format = + encoding_settings.begin()->second.sdp_video_format; + + std::unique_ptr<VideoEncoderFactory> encoder_factory = + CreateEncoderFactory(codec_impl); + if (!encoder_factory + ->QueryCodecSupport(sdp_video_format, + /*scalability_mode=*/absl::nullopt) + .is_supported) { + RTC_LOG(LS_WARNING) << "No encoder for video format " + << sdp_video_format.ToString(); return nullptr; } - RTC_LOG(LS_INFO) << "Encoder implementation: " - << encoder->encoder()->GetEncoderInfo().implementation_name; - - VideoCodecTester::EncoderSettings encoder_settings; - encoder_settings.pacing.mode = - encoder->encoder()->GetEncoderInfo().is_hardware_accelerated - ? PacingMode::kRealTime - : PacingMode::kNoPacing; - std::string output_path = TestOutputPath(); - if (save_codec_input) { + VideoCodecTester::EncoderSettings encoder_settings; + encoder_settings.pacing_settings.mode = + codec_impl == "builtin" ? PacingMode::kNoPacing : PacingMode::kRealTime; + if (absl::GetFlag(FLAGS_dump_encoder_input)) { encoder_settings.encoder_input_base_path = output_path + "_enc_input"; } - if (save_codec_output) { + if (absl::GetFlag(FLAGS_dump_encoder_output)) { encoder_settings.encoder_output_base_path = output_path + "_enc_output"; } - std::unique_ptr<VideoCodecTester> tester = CreateVideoCodecTester(); - return tester->RunEncodeTest(video_source.get(), encoder.get(), - encoder_settings); + return VideoCodecTester::RunEncodeTest(source_settings, encoder_factory.get(), + encoder_settings, encoding_settings); } -class SpatialQualityTest : public ::testing::TestWithParam< - std::tuple</*codec_type=*/std::string, - /*codec_impl=*/std::string, - VideoInfo, - std::tuple</*width=*/int, - /*height=*/int, - /*framerate_fps=*/double, - /*bitrate_kbps=*/int, - /*min_psnr=*/double>>> { +class SpatialQualityTest : public ::testing::TestWithParam<std::tuple< + /*codec_type=*/std::string, + /*codec_impl=*/std::string, + VideoInfo, + std::tuple</*width=*/int, + /*height=*/int, + /*framerate_fps=*/double, + /*bitrate_kbps=*/int, + /*expected_min_psnr=*/double>>> { public: static std::string TestParamsToString( const ::testing::TestParamInfo<SpatialQualityTest::ParamType>& info) { @@ -590,41 +307,35 @@ class SpatialQualityTest : public ::testing::TestWithParam< TEST_P(SpatialQualityTest, SpatialQuality) { auto [codec_type, codec_impl, video_info, coding_settings] = GetParam(); - auto [width, height, framerate_fps, bitrate_kbps, psnr] = coding_settings; - - std::map<int, EncodingSettings> frame_settings = { - {0, - {.scalability_mode = ScalabilityMode::kL1T1, - .layer_settings = { - {LayerId{.spatial_idx = 0, .temporal_idx = 0}, - {.resolution = {.width = width, .height = height}, - .framerate = Frequency::MilliHertz(1000 * framerate_fps), - .bitrate = DataRate::KilobitsPerSec(bitrate_kbps)}}}}}}; - + auto [width, height, framerate_fps, bitrate_kbps, expected_min_psnr] = + coding_settings; int duration_s = 10; int num_frames = duration_s * framerate_fps; - std::unique_ptr<VideoCodecStats> stats = RunEncodeDecodeTest( - codec_type, codec_impl, video_info, frame_settings, num_frames, - /*save_codec_input=*/false, /*save_codec_output=*/false); + std::map<uint32_t, EncodingSettings> frames_settings = + VideoCodecTester::CreateEncodingSettings( + codec_type, /*scalability_mode=*/"L1T1", width, height, + {bitrate_kbps}, framerate_fps, num_frames); + + std::unique_ptr<VideoCodecStats> stats = + RunEncodeDecodeTest(codec_impl, video_info, frames_settings); VideoCodecStats::Stream stream; if (stats != nullptr) { - std::vector<VideoCodecStats::Frame> frames = stats->Slice(); - SetTargetRates(frame_settings, frames); - stream = stats->Aggregate(frames); + stream = stats->Aggregate(Filter{}); if (absl::GetFlag(FLAGS_webrtc_quick_perf_test)) { - EXPECT_GE(stream.psnr.y.GetAverage(), psnr); + EXPECT_GE(stream.psnr.y.GetAverage(), expected_min_psnr); } } stream.LogMetrics( GetGlobalMetricsLogger(), ::testing::UnitTest::GetInstance()->current_test_info()->name(), + /*prefix=*/"", /*metadata=*/ - {{"codec_type", codec_type}, - {"codec_impl", codec_impl}, - {"video_name", video_info.name}}); + {{"video_name", video_info.name}, + {"codec_type", codec_type}, + {"codec_impl", codec_impl}}); } INSTANTIATE_TEST_SUITE_P( @@ -636,7 +347,7 @@ INSTANTIATE_TEST_SUITE_P( #else Values("builtin"), #endif - Values(kFourPeople_1280x720_30), + Values(kRawVideos.at("FourPeople_1280x720_30")), Values(std::make_tuple(320, 180, 30, 32, 28), std::make_tuple(320, 180, 30, 64, 30), std::make_tuple(320, 180, 30, 128, 33), @@ -671,33 +382,32 @@ TEST_P(BitrateAdaptationTest, BitrateAdaptation) { auto [codec_type, codec_impl, video_info, bitrate_kbps] = GetParam(); int duration_s = 10; // Duration of fixed rate interval. - int first_frame = duration_s * video_info.framerate.millihertz() / 1000; - int num_frames = 2 * duration_s * video_info.framerate.millihertz() / 1000; - - std::map<int, EncodingSettings> frame_settings = { - {0, - {.layer_settings = {{LayerId{.spatial_idx = 0, .temporal_idx = 0}, - {.resolution = {.width = 640, .height = 360}, - .framerate = video_info.framerate, - .bitrate = DataRate::KilobitsPerSec( - bitrate_kbps.first)}}}}}, - {first_frame, - {.layer_settings = { - {LayerId{.spatial_idx = 0, .temporal_idx = 0}, - {.resolution = {.width = 640, .height = 360}, - .framerate = video_info.framerate, - .bitrate = DataRate::KilobitsPerSec(bitrate_kbps.second)}}}}}}; - - std::unique_ptr<VideoCodecStats> stats = RunEncodeTest( - codec_type, codec_impl, video_info, frame_settings, num_frames, - /*save_codec_input=*/false, /*save_codec_output=*/false); + int num_frames = + static_cast<int>(duration_s * video_info.framerate.hertz<double>()); + + std::map<uint32_t, EncodingSettings> encoding_settings = + VideoCodecTester::CreateEncodingSettings( + codec_type, /*scalability_mode=*/"L1T1", + /*width=*/640, /*height=*/360, {bitrate_kbps.first}, + /*framerate_fps=*/30, num_frames); + + uint32_t initial_timestamp_rtp = + encoding_settings.rbegin()->first + k90kHz / Frequency::Hertz(30); + + std::map<uint32_t, EncodingSettings> encoding_settings2 = + VideoCodecTester::CreateEncodingSettings( + codec_type, /*scalability_mode=*/"L1T1", + /*width=*/640, /*height=*/360, {bitrate_kbps.second}, + /*framerate_fps=*/30, num_frames, initial_timestamp_rtp); + + encoding_settings.merge(encoding_settings2); + + std::unique_ptr<VideoCodecStats> stats = + RunEncodeTest(codec_type, codec_impl, video_info, encoding_settings); VideoCodecStats::Stream stream; if (stats != nullptr) { - std::vector<VideoCodecStats::Frame> frames = - stats->Slice(VideoCodecStats::Filter{.first_frame = first_frame}); - SetTargetRates(frame_settings, frames); - stream = stats->Aggregate(frames); + stream = stats->Aggregate({.min_timestamp_rtp = initial_timestamp_rtp}); if (absl::GetFlag(FLAGS_webrtc_quick_perf_test)) { EXPECT_NEAR(stream.bitrate_mismatch_pct.GetAverage(), 0, 10); EXPECT_NEAR(stream.framerate_mismatch_pct.GetAverage(), 0, 10); @@ -707,6 +417,7 @@ TEST_P(BitrateAdaptationTest, BitrateAdaptation) { stream.LogMetrics( GetGlobalMetricsLogger(), ::testing::UnitTest::GetInstance()->current_test_info()->name(), + /*prefix=*/"", /*metadata=*/ {{"codec_type", codec_type}, {"codec_impl", codec_impl}, @@ -715,18 +426,18 @@ TEST_P(BitrateAdaptationTest, BitrateAdaptation) { std::to_string(bitrate_kbps.second)}}); } -INSTANTIATE_TEST_SUITE_P(All, - BitrateAdaptationTest, - Combine(Values("AV1", "VP9", "VP8", "H264", "H265"), +INSTANTIATE_TEST_SUITE_P( + All, + BitrateAdaptationTest, + Combine(Values("AV1", "VP9", "VP8", "H264", "H265"), #if defined(WEBRTC_ANDROID) - Values("builtin", "mediacodec"), + Values("builtin", "mediacodec"), #else - Values("builtin"), + Values("builtin"), #endif - Values(kFourPeople_1280x720_30), - Values(std::pair(1024, 512), - std::pair(512, 1024))), - BitrateAdaptationTest::TestParamsToString); + Values(kRawVideos.at("FourPeople_1280x720_30")), + Values(std::pair(1024, 512), std::pair(512, 1024))), + BitrateAdaptationTest::TestParamsToString); class FramerateAdaptationTest : public ::testing::TestWithParam<std::tuple</*codec_type=*/std::string, @@ -749,34 +460,34 @@ TEST_P(FramerateAdaptationTest, FramerateAdaptation) { auto [codec_type, codec_impl, video_info, framerate_fps] = GetParam(); int duration_s = 10; // Duration of fixed rate interval. - int first_frame = static_cast<int>(duration_s * framerate_fps.first); - int num_frames = static_cast<int>( - duration_s * (framerate_fps.first + framerate_fps.second)); - - std::map<int, EncodingSettings> frame_settings = { - {0, - {.layer_settings = {{LayerId{.spatial_idx = 0, .temporal_idx = 0}, - {.resolution = {.width = 640, .height = 360}, - .framerate = Frequency::MilliHertz( - 1000 * framerate_fps.first), - .bitrate = DataRate::KilobitsPerSec(512)}}}}}, - {first_frame, - {.layer_settings = { - {LayerId{.spatial_idx = 0, .temporal_idx = 0}, - {.resolution = {.width = 640, .height = 360}, - .framerate = Frequency::MilliHertz(1000 * framerate_fps.second), - .bitrate = DataRate::KilobitsPerSec(512)}}}}}}; - - std::unique_ptr<VideoCodecStats> stats = RunEncodeTest( - codec_type, codec_impl, video_info, frame_settings, num_frames, - /*save_codec_input=*/false, /*save_codec_output=*/false); + + std::map<uint32_t, EncodingSettings> encoding_settings = + VideoCodecTester::CreateEncodingSettings( + codec_type, /*scalability_mode=*/"L1T1", + /*width=*/640, /*height=*/360, + /*layer_bitrates_kbps=*/{512}, framerate_fps.first, + static_cast<int>(duration_s * framerate_fps.first)); + + uint32_t initial_timestamp_rtp = + encoding_settings.rbegin()->first + + k90kHz / Frequency::Hertz(framerate_fps.first); + + std::map<uint32_t, EncodingSettings> encoding_settings2 = + VideoCodecTester::CreateEncodingSettings( + codec_type, /*scalability_mode=*/"L1T1", /*width=*/640, + /*height=*/360, + /*layer_bitrates_kbps=*/{512}, framerate_fps.second, + static_cast<int>(duration_s * framerate_fps.second), + initial_timestamp_rtp); + + encoding_settings.merge(encoding_settings2); + + std::unique_ptr<VideoCodecStats> stats = + RunEncodeTest(codec_type, codec_impl, video_info, encoding_settings); VideoCodecStats::Stream stream; if (stats != nullptr) { - std::vector<VideoCodecStats::Frame> frames = - stats->Slice(VideoCodecStats::Filter{.first_frame = first_frame}); - SetTargetRates(frame_settings, frames); - stream = stats->Aggregate(frames); + stream = stats->Aggregate({.min_timestamp_rtp = initial_timestamp_rtp}); if (absl::GetFlag(FLAGS_webrtc_quick_perf_test)) { EXPECT_NEAR(stream.bitrate_mismatch_pct.GetAverage(), 0, 10); EXPECT_NEAR(stream.framerate_mismatch_pct.GetAverage(), 0, 10); @@ -786,6 +497,7 @@ TEST_P(FramerateAdaptationTest, FramerateAdaptation) { stream.LogMetrics( GetGlobalMetricsLogger(), ::testing::UnitTest::GetInstance()->current_test_info()->name(), + /*prefix=*/"", /*metadata=*/ {{"codec_type", codec_type}, {"codec_impl", codec_impl}, @@ -794,17 +506,71 @@ TEST_P(FramerateAdaptationTest, FramerateAdaptation) { std::to_string(framerate_fps.second)}}); } -INSTANTIATE_TEST_SUITE_P(All, - FramerateAdaptationTest, - Combine(Values("AV1", "VP9", "VP8", "H264", "H265"), +INSTANTIATE_TEST_SUITE_P( + All, + FramerateAdaptationTest, + Combine(Values("AV1", "VP9", "VP8", "H264", "H265"), #if defined(WEBRTC_ANDROID) - Values("builtin", "mediacodec"), + Values("builtin", "mediacodec"), #else - Values("builtin"), + Values("builtin"), #endif - Values(kFourPeople_1280x720_30), - Values(std::pair(30, 15), std::pair(15, 30))), - FramerateAdaptationTest::TestParamsToString); + Values(kRawVideos.at("FourPeople_1280x720_30")), + Values(std::pair(30, 15), std::pair(15, 30))), + FramerateAdaptationTest::TestParamsToString); + +TEST(VideoCodecTest, DISABLED_EncodeDecode) { + std::vector<std::string> bitrate_str = absl::GetFlag(FLAGS_bitrate_kbps); + std::vector<int> bitrate_kbps; + std::transform(bitrate_str.begin(), bitrate_str.end(), + std::back_inserter(bitrate_kbps), + [](const std::string& str) { return std::stoi(str); }); + + std::map<uint32_t, EncodingSettings> frames_settings = + VideoCodecTester::CreateEncodingSettings( + CodecNameToCodecType(absl::GetFlag(FLAGS_encoder)), + absl::GetFlag(FLAGS_scalability_mode), absl::GetFlag(FLAGS_width), + absl::GetFlag(FLAGS_height), {bitrate_kbps}, + absl::GetFlag(FLAGS_framerate_fps), absl::GetFlag(FLAGS_num_frames)); + + // TODO(webrtc:14852): Pass encoder and decoder names directly, and update + // logged test name (implies lossing history in the chromeperf dashboard). + // Sync with changes in Stream::LogMetrics (see TODOs there). + std::unique_ptr<VideoCodecStats> stats = RunEncodeDecodeTest( + CodecNameToCodecImpl(absl::GetFlag(FLAGS_encoder)), + kRawVideos.at(absl::GetFlag(FLAGS_video_name)), frames_settings); + ASSERT_NE(nullptr, stats); + + // Log unsliced metrics. + VideoCodecStats::Stream stream = stats->Aggregate(Filter{}); + stream.LogMetrics(GetGlobalMetricsLogger(), TestName(), /*prefix=*/"", + /*metadata=*/{}); + + // Log metrics sliced on spatial and temporal layer. + ScalabilityMode scalability_mode = + *ScalabilityModeFromString(absl::GetFlag(FLAGS_scalability_mode)); + int num_spatial_layers = ScalabilityModeToNumSpatialLayers(scalability_mode); + int num_temporal_layers = + ScalabilityModeToNumTemporalLayers(scalability_mode); + for (int sidx = 0; sidx < num_spatial_layers; ++sidx) { + for (int tidx = 0; tidx < num_temporal_layers; ++tidx) { + std::string metric_name_prefix = + (rtc::StringBuilder() << "s" << sidx << "t" << tidx << "_").str(); + stream = stats->Aggregate( + {.layer_id = {{.spatial_idx = sidx, .temporal_idx = tidx}}}); + stream.LogMetrics(GetGlobalMetricsLogger(), TestName(), + metric_name_prefix, + /*metadata=*/{}); + } + } + + if (absl::GetFlag(FLAGS_write_csv)) { + stats->LogMetrics( + (rtc::StringBuilder() << TestOutputPath() << ".csv").str(), + stats->Slice(Filter{}, /*merge=*/false), /*metadata=*/ + {{"test_name", TestName()}}); + } +} } // namespace test diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.cc deleted file mode 100644 index f15b1b35f3..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.cc +++ /dev/null @@ -1,437 +0,0 @@ -/* - * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/video_coding/codecs/test/video_codec_tester_impl.h" - -#include <map> -#include <memory> -#include <string> -#include <utility> - -#include "api/task_queue/default_task_queue_factory.h" -#include "api/units/frequency.h" -#include "api/units/time_delta.h" -#include "api/units/timestamp.h" -#include "api/video/encoded_image.h" -#include "api/video/i420_buffer.h" -#include "api/video/video_codec_type.h" -#include "api/video/video_frame.h" -#include "modules/video_coding/codecs/test/video_codec_analyzer.h" -#include "modules/video_coding/utility/ivf_file_writer.h" -#include "rtc_base/event.h" -#include "rtc_base/time_utils.h" -#include "system_wrappers/include/sleep.h" -#include "test/testsupport/video_frame_writer.h" - -namespace webrtc { -namespace test { - -namespace { -using RawVideoSource = VideoCodecTester::RawVideoSource; -using CodedVideoSource = VideoCodecTester::CodedVideoSource; -using Decoder = VideoCodecTester::Decoder; -using Encoder = VideoCodecTester::Encoder; -using EncoderSettings = VideoCodecTester::EncoderSettings; -using DecoderSettings = VideoCodecTester::DecoderSettings; -using PacingSettings = VideoCodecTester::PacingSettings; -using PacingMode = PacingSettings::PacingMode; - -constexpr Frequency k90kHz = Frequency::Hertz(90000); - -// A thread-safe wrapper for video source to be shared with the quality analyzer -// that reads reference frames from a separate thread. -class SyncRawVideoSource : public VideoCodecAnalyzer::ReferenceVideoSource { - public: - explicit SyncRawVideoSource(RawVideoSource* video_source) - : video_source_(video_source) {} - - absl::optional<VideoFrame> PullFrame() { - MutexLock lock(&mutex_); - return video_source_->PullFrame(); - } - - VideoFrame GetFrame(uint32_t timestamp_rtp, Resolution resolution) override { - MutexLock lock(&mutex_); - return video_source_->GetFrame(timestamp_rtp, resolution); - } - - protected: - RawVideoSource* const video_source_ RTC_GUARDED_BY(mutex_); - Mutex mutex_; -}; - -// Pacer calculates delay necessary to keep frame encode or decode call spaced -// from the previous calls by the pacing time. `Delay` is expected to be called -// as close as possible to posting frame encode or decode task. This class is -// not thread safe. -class Pacer { - public: - explicit Pacer(PacingSettings settings) - : settings_(settings), delay_(TimeDelta::Zero()) {} - Timestamp Schedule(Timestamp timestamp) { - Timestamp now = Timestamp::Micros(rtc::TimeMicros()); - if (settings_.mode == PacingMode::kNoPacing) { - return now; - } - - Timestamp scheduled = now; - if (prev_scheduled_) { - scheduled = *prev_scheduled_ + PacingTime(timestamp); - if (scheduled < now) { - scheduled = now; - } - } - - prev_timestamp_ = timestamp; - prev_scheduled_ = scheduled; - return scheduled; - } - - private: - TimeDelta PacingTime(Timestamp timestamp) { - if (settings_.mode == PacingMode::kRealTime) { - return timestamp - *prev_timestamp_; - } - RTC_CHECK_EQ(PacingMode::kConstantRate, settings_.mode); - return 1 / settings_.constant_rate; - } - - PacingSettings settings_; - absl::optional<Timestamp> prev_timestamp_; - absl::optional<Timestamp> prev_scheduled_; - TimeDelta delay_; -}; - -// Task queue that keeps the number of queued tasks below a certain limit. If -// the limit is reached, posting of a next task is blocked until execution of a -// previously posted task starts. This class is not thread-safe. -class LimitedTaskQueue { - public: - // The codec tester reads frames from video source in the main thread. - // Encoding and decoding are done in separate threads. If encoding or - // decoding is slow, the reading may go far ahead and may buffer too many - // frames in memory. To prevent this we limit the encoding/decoding queue - // size. When the queue is full, the main thread and, hence, reading frames - // from video source is blocked until a previously posted encoding/decoding - // task starts. - static constexpr int kMaxTaskQueueSize = 3; - - LimitedTaskQueue() : queue_size_(0) {} - - void PostScheduledTask(absl::AnyInvocable<void() &&> task, Timestamp start) { - ++queue_size_; - task_queue_.PostTask([this, task = std::move(task), start]() mutable { - int wait_ms = static_cast<int>(start.ms() - rtc::TimeMillis()); - if (wait_ms > 0) { - SleepMs(wait_ms); - } - - std::move(task)(); - --queue_size_; - task_executed_.Set(); - }); - - task_executed_.Reset(); - if (queue_size_ > kMaxTaskQueueSize) { - task_executed_.Wait(rtc::Event::kForever); - } - RTC_CHECK(queue_size_ <= kMaxTaskQueueSize); - } - - void WaitForPreviouslyPostedTasks() { - task_queue_.SendTask([] {}); - } - - TaskQueueForTest task_queue_; - std::atomic_int queue_size_; - rtc::Event task_executed_; -}; - -class TesterY4mWriter { - public: - explicit TesterY4mWriter(absl::string_view base_path) - : base_path_(base_path) {} - - ~TesterY4mWriter() { - task_queue_.SendTask([] {}); - } - - void Write(const VideoFrame& frame, int spatial_idx) { - task_queue_.PostTask([this, frame, spatial_idx] { - if (y4m_writers_.find(spatial_idx) == y4m_writers_.end()) { - std::string file_path = - base_path_ + "_s" + std::to_string(spatial_idx) + ".y4m"; - - Y4mVideoFrameWriterImpl* y4m_writer = new Y4mVideoFrameWriterImpl( - file_path, frame.width(), frame.height(), /*fps=*/30); - RTC_CHECK(y4m_writer); - - y4m_writers_[spatial_idx] = - std::unique_ptr<VideoFrameWriter>(y4m_writer); - } - - y4m_writers_.at(spatial_idx)->WriteFrame(frame); - }); - } - - protected: - std::string base_path_; - std::map<int, std::unique_ptr<VideoFrameWriter>> y4m_writers_; - TaskQueueForTest task_queue_; -}; - -class TesterIvfWriter { - public: - explicit TesterIvfWriter(absl::string_view base_path) - : base_path_(base_path) {} - - ~TesterIvfWriter() { - task_queue_.SendTask([] {}); - } - - void Write(const EncodedImage& encoded_frame) { - task_queue_.PostTask([this, encoded_frame] { - int spatial_idx = encoded_frame.SpatialIndex().value_or(0); - if (ivf_file_writers_.find(spatial_idx) == ivf_file_writers_.end()) { - std::string ivf_path = - base_path_ + "_s" + std::to_string(spatial_idx) + ".ivf"; - - FileWrapper ivf_file = FileWrapper::OpenWriteOnly(ivf_path); - RTC_CHECK(ivf_file.is_open()); - - std::unique_ptr<IvfFileWriter> ivf_writer = - IvfFileWriter::Wrap(std::move(ivf_file), /*byte_limit=*/0); - RTC_CHECK(ivf_writer); - - ivf_file_writers_[spatial_idx] = std::move(ivf_writer); - } - - // To play: ffplay -vcodec vp8|vp9|av1|hevc|h264 filename - ivf_file_writers_.at(spatial_idx) - ->WriteFrame(encoded_frame, VideoCodecType::kVideoCodecGeneric); - }); - } - - protected: - std::string base_path_; - std::map<int, std::unique_ptr<IvfFileWriter>> ivf_file_writers_; - TaskQueueForTest task_queue_; -}; - -class TesterDecoder { - public: - TesterDecoder(Decoder* decoder, - VideoCodecAnalyzer* analyzer, - const DecoderSettings& settings) - : decoder_(decoder), - analyzer_(analyzer), - settings_(settings), - pacer_(settings.pacing) { - RTC_CHECK(analyzer_) << "Analyzer must be provided"; - - if (settings.decoder_input_base_path) { - input_writer_ = - std::make_unique<TesterIvfWriter>(*settings.decoder_input_base_path); - } - - if (settings.decoder_output_base_path) { - output_writer_ = - std::make_unique<TesterY4mWriter>(*settings.decoder_output_base_path); - } - } - - void Initialize() { - task_queue_.PostScheduledTask([this] { decoder_->Initialize(); }, - Timestamp::Zero()); - task_queue_.WaitForPreviouslyPostedTasks(); - } - - void Decode(const EncodedImage& input_frame) { - Timestamp timestamp = - Timestamp::Micros((input_frame.RtpTimestamp() / k90kHz).us()); - - task_queue_.PostScheduledTask( - [this, input_frame] { - analyzer_->StartDecode(input_frame); - - decoder_->Decode( - input_frame, - [this, spatial_idx = input_frame.SpatialIndex().value_or(0)]( - const VideoFrame& output_frame) { - analyzer_->FinishDecode(output_frame, spatial_idx); - - if (output_writer_) { - output_writer_->Write(output_frame, spatial_idx); - } - }); - - if (input_writer_) { - input_writer_->Write(input_frame); - } - }, - pacer_.Schedule(timestamp)); - } - - void Flush() { - task_queue_.PostScheduledTask([this] { decoder_->Flush(); }, - Timestamp::Zero()); - task_queue_.WaitForPreviouslyPostedTasks(); - } - - protected: - Decoder* const decoder_; - VideoCodecAnalyzer* const analyzer_; - const DecoderSettings& settings_; - Pacer pacer_; - LimitedTaskQueue task_queue_; - std::unique_ptr<TesterIvfWriter> input_writer_; - std::unique_ptr<TesterY4mWriter> output_writer_; -}; - -class TesterEncoder { - public: - TesterEncoder(Encoder* encoder, - TesterDecoder* decoder, - VideoCodecAnalyzer* analyzer, - const EncoderSettings& settings) - : encoder_(encoder), - decoder_(decoder), - analyzer_(analyzer), - settings_(settings), - pacer_(settings.pacing) { - RTC_CHECK(analyzer_) << "Analyzer must be provided"; - if (settings.encoder_input_base_path) { - input_writer_ = - std::make_unique<TesterY4mWriter>(*settings.encoder_input_base_path); - } - - if (settings.encoder_output_base_path) { - output_writer_ = - std::make_unique<TesterIvfWriter>(*settings.encoder_output_base_path); - } - } - - void Initialize() { - task_queue_.PostScheduledTask([this] { encoder_->Initialize(); }, - Timestamp::Zero()); - task_queue_.WaitForPreviouslyPostedTasks(); - } - - void Encode(const VideoFrame& input_frame) { - Timestamp timestamp = - Timestamp::Micros((input_frame.timestamp() / k90kHz).us()); - - task_queue_.PostScheduledTask( - [this, input_frame] { - analyzer_->StartEncode(input_frame); - encoder_->Encode(input_frame, - [this](const EncodedImage& encoded_frame) { - analyzer_->FinishEncode(encoded_frame); - - if (decoder_ != nullptr) { - decoder_->Decode(encoded_frame); - } - - if (output_writer_ != nullptr) { - output_writer_->Write(encoded_frame); - } - }); - - if (input_writer_) { - input_writer_->Write(input_frame, /*spatial_idx=*/0); - } - }, - pacer_.Schedule(timestamp)); - } - - void Flush() { - task_queue_.PostScheduledTask([this] { encoder_->Flush(); }, - Timestamp::Zero()); - task_queue_.WaitForPreviouslyPostedTasks(); - } - - protected: - Encoder* const encoder_; - TesterDecoder* const decoder_; - VideoCodecAnalyzer* const analyzer_; - const EncoderSettings& settings_; - std::unique_ptr<TesterY4mWriter> input_writer_; - std::unique_ptr<TesterIvfWriter> output_writer_; - Pacer pacer_; - LimitedTaskQueue task_queue_; -}; - -} // namespace - -std::unique_ptr<VideoCodecStats> VideoCodecTesterImpl::RunDecodeTest( - CodedVideoSource* video_source, - Decoder* decoder, - const DecoderSettings& decoder_settings) { - VideoCodecAnalyzer perf_analyzer; - TesterDecoder tester_decoder(decoder, &perf_analyzer, decoder_settings); - - tester_decoder.Initialize(); - - while (auto frame = video_source->PullFrame()) { - tester_decoder.Decode(*frame); - } - - tester_decoder.Flush(); - - return perf_analyzer.GetStats(); -} - -std::unique_ptr<VideoCodecStats> VideoCodecTesterImpl::RunEncodeTest( - RawVideoSource* video_source, - Encoder* encoder, - const EncoderSettings& encoder_settings) { - SyncRawVideoSource sync_source(video_source); - VideoCodecAnalyzer perf_analyzer; - TesterEncoder tester_encoder(encoder, /*decoder=*/nullptr, &perf_analyzer, - encoder_settings); - - tester_encoder.Initialize(); - - while (auto frame = sync_source.PullFrame()) { - tester_encoder.Encode(*frame); - } - - tester_encoder.Flush(); - - return perf_analyzer.GetStats(); -} - -std::unique_ptr<VideoCodecStats> VideoCodecTesterImpl::RunEncodeDecodeTest( - RawVideoSource* video_source, - Encoder* encoder, - Decoder* decoder, - const EncoderSettings& encoder_settings, - const DecoderSettings& decoder_settings) { - SyncRawVideoSource sync_source(video_source); - VideoCodecAnalyzer perf_analyzer(&sync_source); - TesterDecoder tester_decoder(decoder, &perf_analyzer, decoder_settings); - TesterEncoder tester_encoder(encoder, &tester_decoder, &perf_analyzer, - encoder_settings); - - tester_encoder.Initialize(); - tester_decoder.Initialize(); - - while (auto frame = sync_source.PullFrame()) { - tester_encoder.Encode(*frame); - } - - tester_encoder.Flush(); - tester_decoder.Flush(); - - return perf_analyzer.GetStats(); -} - -} // namespace test -} // namespace webrtc diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.h b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.h deleted file mode 100644 index 32191b5a98..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.h +++ /dev/null @@ -1,45 +0,0 @@ -/* - * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TESTER_IMPL_H_ -#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TESTER_IMPL_H_ - -#include <memory> - -#include "api/test/video_codec_tester.h" - -namespace webrtc { -namespace test { - -// A stateless implementation of `VideoCodecTester`. This class is thread safe. -class VideoCodecTesterImpl : public VideoCodecTester { - public: - std::unique_ptr<VideoCodecStats> RunDecodeTest( - CodedVideoSource* video_source, - Decoder* decoder, - const DecoderSettings& decoder_settings) override; - - std::unique_ptr<VideoCodecStats> RunEncodeTest( - RawVideoSource* video_source, - Encoder* encoder, - const EncoderSettings& encoder_settings) override; - - std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest( - RawVideoSource* video_source, - Encoder* encoder, - Decoder* decoder, - const EncoderSettings& encoder_settings, - const DecoderSettings& decoder_settings) override; -}; - -} // namespace test -} // namespace webrtc - -#endif // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TESTER_IMPL_H_ diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl_unittest.cc deleted file mode 100644 index a8c118ef20..0000000000 --- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl_unittest.cc +++ /dev/null @@ -1,205 +0,0 @@ -/* - * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/video_coding/codecs/test/video_codec_tester_impl.h" - -#include <memory> -#include <tuple> -#include <utility> -#include <vector> - -#include "api/units/frequency.h" -#include "api/units/time_delta.h" -#include "api/video/encoded_image.h" -#include "api/video/i420_buffer.h" -#include "api/video/video_frame.h" -#include "rtc_base/fake_clock.h" -#include "rtc_base/gunit.h" -#include "rtc_base/task_queue_for_test.h" -#include "rtc_base/time_utils.h" -#include "test/gmock.h" -#include "test/gtest.h" - -namespace webrtc { -namespace test { - -namespace { -using ::testing::_; -using ::testing::Invoke; -using ::testing::InvokeWithoutArgs; -using ::testing::Return; - -using Decoder = VideoCodecTester::Decoder; -using Encoder = VideoCodecTester::Encoder; -using CodedVideoSource = VideoCodecTester::CodedVideoSource; -using RawVideoSource = VideoCodecTester::RawVideoSource; -using DecoderSettings = VideoCodecTester::DecoderSettings; -using EncoderSettings = VideoCodecTester::EncoderSettings; -using PacingSettings = VideoCodecTester::PacingSettings; -using PacingMode = PacingSettings::PacingMode; - -constexpr Frequency k90kHz = Frequency::Hertz(90000); - -struct PacingTestParams { - PacingSettings pacing_settings; - Frequency framerate; - int num_frames; - std::vector<int> expected_delta_ms; -}; - -VideoFrame CreateVideoFrame(uint32_t timestamp_rtp) { - rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(2, 2)); - return VideoFrame::Builder() - .set_video_frame_buffer(buffer) - .set_timestamp_rtp(timestamp_rtp) - .build(); -} - -EncodedImage CreateEncodedImage(uint32_t timestamp_rtp) { - EncodedImage encoded_image; - encoded_image.SetRtpTimestamp(timestamp_rtp); - return encoded_image; -} - -class MockRawVideoSource : public RawVideoSource { - public: - MockRawVideoSource(int num_frames, Frequency framerate) - : num_frames_(num_frames), frame_num_(0), framerate_(framerate) {} - - absl::optional<VideoFrame> PullFrame() override { - if (frame_num_ >= num_frames_) { - return absl::nullopt; - } - uint32_t timestamp_rtp = frame_num_ * k90kHz / framerate_; - ++frame_num_; - return CreateVideoFrame(timestamp_rtp); - } - - MOCK_METHOD(VideoFrame, - GetFrame, - (uint32_t timestamp_rtp, Resolution), - (override)); - - private: - int num_frames_; - int frame_num_; - Frequency framerate_; -}; - -class MockCodedVideoSource : public CodedVideoSource { - public: - MockCodedVideoSource(int num_frames, Frequency framerate) - : num_frames_(num_frames), frame_num_(0), framerate_(framerate) {} - - absl::optional<EncodedImage> PullFrame() override { - if (frame_num_ >= num_frames_) { - return absl::nullopt; - } - uint32_t timestamp_rtp = frame_num_ * k90kHz / framerate_; - ++frame_num_; - return CreateEncodedImage(timestamp_rtp); - } - - private: - int num_frames_; - int frame_num_; - Frequency framerate_; -}; - -class MockDecoder : public Decoder { - public: - MOCK_METHOD(void, Initialize, (), (override)); - MOCK_METHOD(void, - Decode, - (const EncodedImage& frame, DecodeCallback callback), - (override)); - MOCK_METHOD(void, Flush, (), (override)); -}; - -class MockEncoder : public Encoder { - public: - MOCK_METHOD(void, Initialize, (), (override)); - MOCK_METHOD(void, - Encode, - (const VideoFrame& frame, EncodeCallback callback), - (override)); - MOCK_METHOD(void, Flush, (), (override)); -}; - -} // namespace - -class VideoCodecTesterImplPacingTest - : public ::testing::TestWithParam<PacingTestParams> { - public: - VideoCodecTesterImplPacingTest() : test_params_(GetParam()) {} - - protected: - PacingTestParams test_params_; -}; - -TEST_P(VideoCodecTesterImplPacingTest, PaceEncode) { - MockRawVideoSource video_source(test_params_.num_frames, - test_params_.framerate); - MockEncoder encoder; - EncoderSettings encoder_settings; - encoder_settings.pacing = test_params_.pacing_settings; - - VideoCodecTesterImpl tester; - auto fs = - tester.RunEncodeTest(&video_source, &encoder, encoder_settings)->Slice(); - ASSERT_EQ(static_cast<int>(fs.size()), test_params_.num_frames); - - for (size_t i = 1; i < fs.size(); ++i) { - int delta_ms = (fs[i].encode_start - fs[i - 1].encode_start).ms(); - EXPECT_NEAR(delta_ms, test_params_.expected_delta_ms[i - 1], 10); - } -} - -TEST_P(VideoCodecTesterImplPacingTest, PaceDecode) { - MockCodedVideoSource video_source(test_params_.num_frames, - test_params_.framerate); - MockDecoder decoder; - DecoderSettings decoder_settings; - decoder_settings.pacing = test_params_.pacing_settings; - - VideoCodecTesterImpl tester; - auto fs = - tester.RunDecodeTest(&video_source, &decoder, decoder_settings)->Slice(); - ASSERT_EQ(static_cast<int>(fs.size()), test_params_.num_frames); - - for (size_t i = 1; i < fs.size(); ++i) { - int delta_ms = (fs[i].decode_start - fs[i - 1].decode_start).ms(); - EXPECT_NEAR(delta_ms, test_params_.expected_delta_ms[i - 1], 20); - } -} - -INSTANTIATE_TEST_SUITE_P( - DISABLED_All, - VideoCodecTesterImplPacingTest, - ::testing::ValuesIn( - {// No pacing. - PacingTestParams({.pacing_settings = {.mode = PacingMode::kNoPacing}, - .framerate = Frequency::Hertz(10), - .num_frames = 3, - .expected_delta_ms = {0, 0}}), - // Real-time pacing. - PacingTestParams({.pacing_settings = {.mode = PacingMode::kRealTime}, - .framerate = Frequency::Hertz(10), - .num_frames = 3, - .expected_delta_ms = {100, 100}}), - // Pace with specified constant rate. - PacingTestParams( - {.pacing_settings = {.mode = PacingMode::kConstantRate, - .constant_rate = Frequency::Hertz(20)}, - .framerate = Frequency::Hertz(10), - .num_frames = 3, - .expected_delta_ms = {50, 50}})})); -} // namespace test -} // namespace webrtc diff --git a/third_party/libwebrtc/modules/video_coding/encoded_frame_gn/moz.build b/third_party/libwebrtc/modules/video_coding/encoded_frame_gn/moz.build index 9b8e33b7d5..31e83f9c31 100644 --- a/third_party/libwebrtc/modules/video_coding/encoded_frame_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/encoded_frame_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_gn/moz.build index 487fc5b4d6..1ad9c574ad 100644 --- a/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/frame_helpers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/frame_helpers_gn/moz.build index dd901a5371..ccac90f50d 100644 --- a/third_party/libwebrtc/modules/video_coding/frame_helpers_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/frame_helpers_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/generic_decoder.cc b/third_party/libwebrtc/modules/video_coding/generic_decoder.cc index fc356e7a44..00585abbc9 100644 --- a/third_party/libwebrtc/modules/video_coding/generic_decoder.cc +++ b/third_party/libwebrtc/modules/video_coding/generic_decoder.cc @@ -329,18 +329,7 @@ int32_t VCMGenericDecoder::Decode(const EncodedImage& frame, } _callback->OnDecoderInfoChanged(std::move(decoder_info)); } - if (ret < WEBRTC_VIDEO_CODEC_OK) { - const absl::optional<uint32_t> ssrc = - !frame_info.packet_infos.empty() - ? absl::make_optional(frame_info.packet_infos[0].ssrc()) - : absl::nullopt; - RTC_LOG(LS_WARNING) << "Failed to decode frame with timestamp " - << frame.RtpTimestamp() << ", ssrc " - << (ssrc ? rtc::ToString(*ssrc) : "<not set>") - << ", error code: " << ret; - _callback->ClearTimestampMap(); - } else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT) { - // No output. + if (ret < WEBRTC_VIDEO_CODEC_OK || ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT) { _callback->ClearTimestampMap(); } return ret; diff --git a/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.h b/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.h index c6522fcc6b..987e1b623e 100644 --- a/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.h +++ b/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.h @@ -50,7 +50,9 @@ struct CodecSpecificInfoVP8 { size_t updatedBuffers[kBuffersCount]; size_t updatedBuffersCount; }; -static_assert(std::is_pod<CodecSpecificInfoVP8>::value, ""); +static_assert(std::is_trivial_v<CodecSpecificInfoVP8> && + std::is_standard_layout_v<CodecSpecificInfoVP8>, + ""); // Hack alert - the code assumes that thisstruct is memset when constructed. struct CodecSpecificInfoVP9 { @@ -79,7 +81,9 @@ struct CodecSpecificInfoVP9 { uint8_t num_ref_pics; uint8_t p_diff[kMaxVp9RefPics]; }; -static_assert(std::is_pod<CodecSpecificInfoVP9>::value, ""); +static_assert(std::is_trivial_v<CodecSpecificInfoVP9> && + std::is_standard_layout_v<CodecSpecificInfoVP9>, + ""); // Hack alert - the code assumes that thisstruct is memset when constructed. struct CodecSpecificInfoH264 { @@ -88,14 +92,18 @@ struct CodecSpecificInfoH264 { bool base_layer_sync; bool idr_frame; }; -static_assert(std::is_pod<CodecSpecificInfoH264>::value, ""); +static_assert(std::is_trivial_v<CodecSpecificInfoH264> && + std::is_standard_layout_v<CodecSpecificInfoH264>, + ""); union CodecSpecificInfoUnion { CodecSpecificInfoVP8 VP8; CodecSpecificInfoVP9 VP9; CodecSpecificInfoH264 H264; }; -static_assert(std::is_pod<CodecSpecificInfoUnion>::value, ""); +static_assert(std::is_trivial_v<CodecSpecificInfoUnion> && + std::is_standard_layout_v<CodecSpecificInfoUnion>, + ""); // Note: if any pointers are added to this struct or its sub-structs, it // must be fitted with a copy-constructor. This is because it is copied diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester_gn/moz.build b/third_party/libwebrtc/modules/video_coding/nack_requester_gn/moz.build index 0f6654f1ab..d50ed75a00 100644 --- a/third_party/libwebrtc/modules/video_coding/nack_requester_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/nack_requester_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/packet_buffer_gn/moz.build b/third_party/libwebrtc/modules/video_coding/packet_buffer_gn/moz.build index f3f85aacaa..2c161989c1 100644 --- a/third_party/libwebrtc/modules/video_coding/packet_buffer_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/packet_buffer_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_gn/moz.build index 8a1dfd6377..80eb00a991 100644 --- a/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structures_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/scalability_structures_gn/moz.build index a3ea8b3495..931dfe8d89 100644 --- a/third_party/libwebrtc/modules/video_coding/svc/scalability_structures_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structures_gn/moz.build @@ -202,7 +202,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -212,10 +211,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_gn/moz.build index a285154a79..18aa68e696 100644 --- a/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_gn/moz.build index 412f719d18..bbb5a75959 100644 --- a/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_gn/moz.build @@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/timing/decode_time_percentile_filter_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/decode_time_percentile_filter_gn/moz.build index 36867642c7..2347b0937c 100644 --- a/third_party/libwebrtc/modules/video_coding/timing/decode_time_percentile_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/timing/decode_time_percentile_filter_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_gn/moz.build index caf0efc165..274023c6e7 100644 --- a/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_variation_calculator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_variation_calculator_gn/moz.build index 8c6e826a4a..d4ec330ed1 100644 --- a/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_variation_calculator_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_variation_calculator_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_gn/moz.build index c7ca3c7fd8..e540f00f8c 100644 --- a/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_gn/moz.build index f3993a17b1..18a30a6ede 100644 --- a/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_gn/moz.build index ad8a6874e4..4c2a6eed62 100644 --- a/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/timing/timing_module_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/timing_module_gn/moz.build index 60cc81a229..76c4cfe664 100644 --- a/third_party/libwebrtc/modules/video_coding/timing/timing_module_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/timing/timing_module_gn/moz.build @@ -199,7 +199,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -209,10 +208,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/video_codec_interface_gn/moz.build b/third_party/libwebrtc/modules/video_coding/video_codec_interface_gn/moz.build index b14bef2dec..141def9090 100644 --- a/third_party/libwebrtc/modules/video_coding/video_codec_interface_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/video_codec_interface_gn/moz.build @@ -196,7 +196,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -206,10 +205,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/video_coding_gn/moz.build b/third_party/libwebrtc/modules/video_coding/video_coding_gn/moz.build index 5af51f1238..923ac7785a 100644 --- a/third_party/libwebrtc/modules/video_coding/video_coding_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/video_coding_gn/moz.build @@ -214,7 +214,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -224,10 +223,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/video_coding_utility_gn/moz.build b/third_party/libwebrtc/modules/video_coding/video_coding_utility_gn/moz.build index d42eb284cd..bc1510e0ba 100644 --- a/third_party/libwebrtc/modules/video_coding/video_coding_utility_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/video_coding_utility_gn/moz.build @@ -211,7 +211,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -221,10 +220,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_libvpx_interface_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_libvpx_interface_gn/moz.build index 81c9b9d404..8cb4b64625 100644 --- a/third_party/libwebrtc/modules/video_coding/webrtc_libvpx_interface_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/webrtc_libvpx_interface_gn/moz.build @@ -191,7 +191,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -201,10 +200,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_gn/moz.build index 82a4d24e97..21d5eeee9f 100644 --- a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_gn/moz.build @@ -206,7 +206,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -216,10 +215,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_scalability_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_scalability_gn/moz.build index 6799224dff..92fd7cf630 100644 --- a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_scalability_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_scalability_gn/moz.build @@ -188,7 +188,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -198,10 +197,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_temporal_layers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_temporal_layers_gn/moz.build index 2423950ba5..caf91a5d2c 100644 --- a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_temporal_layers_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_temporal_layers_gn/moz.build @@ -205,7 +205,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -215,10 +214,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp9_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_gn/moz.build index 5bb64f3412..707d563559 100644 --- a/third_party/libwebrtc/modules/video_coding/webrtc_vp9_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_gn/moz.build @@ -208,7 +208,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -218,10 +217,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp9_helpers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_helpers_gn/moz.build index 6f1575870e..883e5c70b2 100644 --- a/third_party/libwebrtc/modules/video_coding/webrtc_vp9_helpers_gn/moz.build +++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_helpers_gn/moz.build @@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": OS_LIBS += [ - "android_support", "unwind" ] @@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": "-msse2" ] - OS_LIBS += [ - "android_support" - ] - if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": DEFINES["_GNU_SOURCE"] = True |