diff options
Diffstat (limited to 'third_party/libwebrtc/moz-patch-stack/0060.patch')
| -rw-r--r-- | third_party/libwebrtc/moz-patch-stack/0060.patch | 163 |
1 files changed, 163 insertions, 0 deletions
diff --git a/third_party/libwebrtc/moz-patch-stack/0060.patch b/third_party/libwebrtc/moz-patch-stack/0060.patch new file mode 100644 index 0000000000..81458c04df --- /dev/null +++ b/third_party/libwebrtc/moz-patch-stack/0060.patch @@ -0,0 +1,163 @@ +From: Michael Froman <mjfroman@mac.com> +Date: Mon, 4 Apr 2022 12:25:26 -0500 +Subject: Bug 1766646 - (fix) breakout Call::Stats and SharedModuleThread into + seperate files + +--- + call/BUILD.gn | 6 ++++++ + call/call.cc | 13 ------------- + call/call.h | 13 ++----------- + call/call_basic_stats.cc | 20 ++++++++++++++++++++ + call/call_basic_stats.h | 21 +++++++++++++++++++++ + video/video_send_stream.h | 1 - + 6 files changed, 49 insertions(+), 25 deletions(-) + create mode 100644 call/call_basic_stats.cc + create mode 100644 call/call_basic_stats.h + +diff --git a/call/BUILD.gn b/call/BUILD.gn +index 0e52e8fb3f..26618aee80 100644 +--- a/call/BUILD.gn ++++ b/call/BUILD.gn +@@ -33,6 +33,12 @@ rtc_library("call_interfaces") { + "syncable.cc", + "syncable.h", + ] ++ if (build_with_mozilla) { ++ sources += [ ++ "call_basic_stats.cc", ++ "call_basic_stats.h", ++ ] ++ } + + deps = [ + ":audio_sender_interface", +diff --git a/call/call.cc b/call/call.cc +index a63087f5c1..4c3f4b63fc 100644 +--- a/call/call.cc ++++ b/call/call.cc +@@ -472,19 +472,6 @@ class Call final : public webrtc::Call, + }; + } // namespace internal + +-std::string Call::Stats::ToString(int64_t time_ms) const { +- char buf[1024]; +- rtc::SimpleStringBuilder ss(buf); +- ss << "Call stats: " << time_ms << ", {"; +- ss << "send_bw_bps: " << send_bandwidth_bps << ", "; +- ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; +- ss << "max_pad_bps: " << max_padding_bitrate_bps << ", "; +- ss << "pacer_delay_ms: " << pacer_delay_ms << ", "; +- ss << "rtt_ms: " << rtt_ms; +- ss << '}'; +- return ss.str(); +-} +- + /* Mozilla: Avoid this since it could use GetRealTimeClock(). + Call* Call::Create(const Call::Config& config) { + Clock* clock = Clock::GetRealTimeClock(); +diff --git a/call/call.h b/call/call.h +index 366978392e..42daa95a6c 100644 +--- a/call/call.h ++++ b/call/call.h +@@ -21,6 +21,7 @@ + #include "api/task_queue/task_queue_base.h" + #include "call/audio_receive_stream.h" + #include "call/audio_send_stream.h" ++#include "call/call_basic_stats.h" + #include "call/call_config.h" + #include "call/flexfec_receive_stream.h" + #include "call/packet_receiver.h" +@@ -30,7 +31,6 @@ + #include "rtc_base/copy_on_write_buffer.h" + #include "rtc_base/network/sent_packet.h" + #include "rtc_base/network_route.h" +-#include "rtc_base/ref_count.h" + + namespace webrtc { + +@@ -47,16 +47,7 @@ namespace webrtc { + class Call { + public: + using Config = CallConfig; +- +- struct Stats { +- std::string ToString(int64_t time_ms) const; +- +- int send_bandwidth_bps = 0; // Estimated available send bandwidth. +- int max_padding_bitrate_bps = 0; // Cumulative configured max padding. +- int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. +- int64_t pacer_delay_ms = 0; +- int64_t rtt_ms = -1; +- }; ++ using Stats = CallBasicStats; + + static Call* Create(const Call::Config& config); + static Call* Create(const Call::Config& config, +diff --git a/call/call_basic_stats.cc b/call/call_basic_stats.cc +new file mode 100644 +index 0000000000..74333a663b +--- /dev/null ++++ b/call/call_basic_stats.cc +@@ -0,0 +1,20 @@ ++#include "call/call_basic_stats.h" ++ ++#include "rtc_base/strings/string_builder.h" ++ ++namespace webrtc { ++ ++std::string CallBasicStats::ToString(int64_t time_ms) const { ++ char buf[1024]; ++ rtc::SimpleStringBuilder ss(buf); ++ ss << "Call stats: " << time_ms << ", {"; ++ ss << "send_bw_bps: " << send_bandwidth_bps << ", "; ++ ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; ++ ss << "max_pad_bps: " << max_padding_bitrate_bps << ", "; ++ ss << "pacer_delay_ms: " << pacer_delay_ms << ", "; ++ ss << "rtt_ms: " << rtt_ms; ++ ss << '}'; ++ return ss.str(); ++} ++ ++} // namespace webrtc +diff --git a/call/call_basic_stats.h b/call/call_basic_stats.h +new file mode 100644 +index 0000000000..98febe9405 +--- /dev/null ++++ b/call/call_basic_stats.h +@@ -0,0 +1,21 @@ ++#ifndef CALL_CALL_BASIC_STATS_H_ ++#define CALL_CALL_BASIC_STATS_H_ ++ ++#include <string> ++ ++namespace webrtc { ++ ++// named to avoid conflicts with video/call_stats.h ++struct CallBasicStats { ++ std::string ToString(int64_t time_ms) const; ++ ++ int send_bandwidth_bps = 0; // Estimated available send bandwidth. ++ int max_padding_bitrate_bps = 0; // Cumulative configured max padding. ++ int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. ++ int64_t pacer_delay_ms = 0; ++ int64_t rtt_ms = -1; ++}; ++ ++} // namespace webrtc ++ ++#endif // CALL_CALL_BASIC_STATS_H_ +diff --git a/video/video_send_stream.h b/video/video_send_stream.h +index a7ce112b21..404873fd39 100644 +--- a/video/video_send_stream.h ++++ b/video/video_send_stream.h +@@ -37,7 +37,6 @@ namespace test { + class VideoSendStreamPeer; + } // namespace test + +-class CallStats; + class IvfFileWriter; + class RateLimiter; + class RtpRtcp; +-- +2.34.1 + |