From d8bbc7858622b6d9c278469aab701ca0b609cddf Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 15 May 2024 05:35:49 +0200 Subject: Merging upstream version 126.0. Signed-off-by: Daniel Baumann --- .../rtp_rtcp/source/receive_statistics_impl.cc | 8 +- .../rtp_rtcp/source/receive_statistics_unittest.cc | 17 ++ .../modules/rtp_rtcp/source/rtcp_receiver.cc | 4 +- .../source/rtp_dependency_descriptor_extension.cc | 13 ++ .../source/rtp_dependency_descriptor_extension.h | 10 ++ .../modules/rtp_rtcp/source/rtp_format_h264.cc | 27 +-- .../rtp_rtcp/source/rtp_format_h264_unittest.cc | 186 +++++++++++++++++++++ .../modules/rtp_rtcp/source/rtp_format_vp8.cc | 4 +- .../rtp_rtcp/source/rtp_format_vp8_unittest.cc | 12 ++ .../modules/rtp_rtcp/source/rtp_format_vp9.cc | 5 +- .../rtp_rtcp/source/rtp_format_vp9_unittest.cc | 5 + .../libwebrtc/modules/rtp_rtcp/source/rtp_packet.h | 8 +- .../modules/rtp_rtcp/source/rtp_packet_to_send.cc | 11 ++ .../modules/rtp_rtcp/source/rtp_packet_to_send.h | 10 +- .../modules/rtp_rtcp/source/rtp_packet_unittest.cc | 35 ++++ .../rtp_rtcp/source/rtp_packetizer_av1_unittest.cc | 6 + 16 files changed, 337 insertions(+), 24 deletions(-) (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source') diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc index 0e5e40f502..1dc56bb96f 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc @@ -18,6 +18,7 @@ #include "api/units/time_delta.h" #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" @@ -159,16 +160,15 @@ void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet, int32_t time_diff_samples = receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_); - time_diff_samples = std::abs(time_diff_samples); - ReviseFrequencyAndJitter(packet.payload_type_frequency()); // lib_jingle sometimes deliver crazy jumps in TS for the same stream. // If this happens, don't update jitter value. Use 5 secs video frequency // as the threshold. - if (time_diff_samples < 450000) { + if (time_diff_samples < 5 * kVideoPayloadTypeFrequency && + time_diff_samples > -5 * kVideoPayloadTypeFrequency) { // Note we calculate in Q4 to avoid using float. - int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_; + int32_t jitter_diff_q4 = (std::abs(time_diff_samples) << 4) - jitter_q4_; jitter_q4_ += ((jitter_diff_q4 + 8) >> 4); } } diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc index a2558545f0..8b31912f0f 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc @@ -898,5 +898,22 @@ TEST(ReviseJitterTest, EXPECT_EQ(GetJitter(*statistics), 172U); } +TEST(ReviseJitterTest, TwoPacketsWithMaximumRtpTimestampDifference) { + SimulatedClock clock(0); + std::unique_ptr statistics = + ReceiveStatistics::Create(&clock); + RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/90'000, + /*timestamp=*/0x01234567); + RtpPacketReceived packet2 = + MakeNextRtpPacket(packet1, + /*payload_type_frequency=*/90'000, + /*timestamp=*/0x81234567); + statistics->OnRtpPacket(packet1); + statistics->OnRtpPacket(packet2); + + // Expect large jump in RTP timestamp is ignored for jitter calculation. + EXPECT_EQ(GetJitter(*statistics), 0U); +} + } // namespace } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc index bda6ad9a52..756136866d 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -1190,11 +1190,11 @@ std::vector RTCPReceiver::TmmbrReceived() { MutexLock lock(&rtcp_receiver_lock_); std::vector candidates; - Timestamp timeout = clock_->CurrentTime() - kTmmbrTimeoutInterval; + Timestamp now = clock_->CurrentTime(); for (auto& kv : tmmbr_infos_) { for (auto it = kv.second.tmmbr.begin(); it != kv.second.tmmbr.end();) { - if (it->second.last_updated < timeout) { + if (now - it->second.last_updated > kTmmbrTimeoutInterval) { // Erase timeout entries. it = kv.second.tmmbr.erase(it); } else { diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc index fd42b798d4..27b0420926 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc @@ -52,4 +52,17 @@ bool RtpDependencyDescriptorExtension::Write( return writer.Write(); } +bool RtpDependencyDescriptorExtensionMandatory::Parse( + rtc::ArrayView data, + DependencyDescriptorMandatory* descriptor) { + if (data.size() < 3) { + return false; + } + descriptor->set_first_packet_in_frame(data[0] & 0b1000'0000); + descriptor->set_last_packet_in_frame(data[0] & 0b0100'0000); + descriptor->set_template_id(data[0] & 0b0011'1111); + descriptor->set_frame_number((uint16_t{data[1]} << 8) | data[2]); + return true; +} + } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h index 8d6e4b8d37..a3e415917c 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h @@ -54,6 +54,16 @@ class RtpDependencyDescriptorExtension { static constexpr std::bitset<32> kAllChainsAreActive = ~uint32_t{0}; }; +// Trait to only read the mandatory part of the descriptor. +class RtpDependencyDescriptorExtensionMandatory { + public: + static constexpr webrtc::RTPExtensionType kId = + webrtc::RtpDependencyDescriptorExtension::kId; + + static bool Parse(rtc::ArrayView data, + DependencyDescriptorMandatory* descriptor); +}; + } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_EXTENSION_H_ diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc index d066bafb90..9c1dc4edb8 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc @@ -19,6 +19,7 @@ #include #include +#include "absl/algorithm/container.h" #include "absl/types/optional.h" #include "absl/types/variant.h" #include "common_video/h264/h264_common.h" @@ -52,11 +53,14 @@ RtpPacketizerH264::RtpPacketizerH264(rtc::ArrayView payload, input_fragments_.push_back( payload.subview(nalu.payload_start_offset, nalu.payload_size)); } - - if (!GeneratePackets(packetization_mode)) { - // 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(). + bool has_empty_fragments = absl::c_any_of( + input_fragments_, [](const rtc::ArrayView fragment) { + return fragment.empty(); + }); + if (has_empty_fragments || !GeneratePackets(packetization_mode)) { + // If empty fragments were found or we failed to generate all the packets, + // discard already generated packets in case the caller would ignore the + // return value and still try to call NextPacket(). num_packets_left_ = 0; while (!packets_.empty()) { packets_.pop(); @@ -73,6 +77,7 @@ size_t RtpPacketizerH264::NumPackets() const { bool RtpPacketizerH264::GeneratePackets( H264PacketizationMode packetization_mode) { for (size_t i = 0; i < input_fragments_.size();) { + RTC_DCHECK(!input_fragments_[i].empty()); switch (packetization_mode) { case H264PacketizationMode::SingleNalUnit: if (!PacketizeSingleNalu(i)) @@ -173,11 +178,11 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) { return fragment_size; } }; - while (payload_size_left >= payload_size_needed()) { RTC_CHECK_GT(fragment.size(), 0); - packets_.push(PacketUnit(fragment, aggregated_fragments == 0, false, true, - fragment[0])); + + packets_.push(PacketUnit(fragment, /*first=*/aggregated_fragments == 0, + /*last=*/false, /*aggregated=*/true, fragment[0])); payload_size_left -= fragment.size(); payload_size_left -= fragment_headers_length; @@ -218,9 +223,9 @@ bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) { << limits_.max_payload_len; return false; } - RTC_CHECK_GT(fragment.size(), 0u); - packets_.push(PacketUnit(fragment, true /* first */, true /* last */, - false /* aggregated */, fragment[0])); + RTC_CHECK(!fragment.empty()); + packets_.push(PacketUnit(fragment, /*first=*/true, /*last=*/true, + /*aggregated=*/false, fragment[0])); ++num_packets_left_; return true; } diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc index 18311c6e8c..3920c4acd5 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc @@ -116,6 +116,7 @@ std::vector FetchAllPackets(RtpPacketizerH264* packetizer) { RtpPacketToSend packet(kNoExtensions); while (packetizer->NextPacket(&packet)) { result.push_back(packet); + packet.Clear(); } EXPECT_THAT(result, SizeIs(num_packets)); return result; @@ -527,5 +528,190 @@ TEST(RtpPacketizerH264Test, RejectsOverlongDataInPacketizationMode0) { EXPECT_THAT(packets, IsEmpty()); } + +TEST(RtpPacketizerH264Test, DoesNotPacketizeWithEmptyNalUnit) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = kMaxPayloadSize; + + uint8_t empty_nal_input[] = {0x00, 0x00, 0x01, /* empty NAL unit data */ + 0x00, 0x00, 0x01, 0x01}; + RtpPacketizerH264 packetizer(empty_nal_input, limits, + H264PacketizationMode::NonInterleaved); + EXPECT_EQ(packetizer.NumPackets(), 0u); +} + +TEST(RtpPacketizerH264Test, MultipleStapA) { + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = kMaxPayloadSize; + limits.first_packet_reduction_len = 0; + limits.last_packet_reduction_len = 0; + limits.single_packet_reduction_len = 0; + // A lot of small NAL units that will result in two STAP-A being generated. + // Input data must exceed the size of a single RTP packet. + uint8_t long_input[] = { + 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, + 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, + 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, + 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, + 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, + 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x01, + 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01, + 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0x0b, 0x01, 0x04, 0x00, + 0x19, 0x00, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, + 0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, + 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, + 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 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0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, + 0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, + 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, + 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, + 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, + 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x01, 0x00, 0x00, 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, + 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, + 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, + 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x10, 0x00, 0x00, + 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, + 0x00, 0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x00, + 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, + 0x08, 0xfe, 0xfb, 0xff, 0xff, 0xf4, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0x04, 0x00, + 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, + 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, + 0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, + 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, + 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, + 0x8e, 0x00, 0x04, 0x00, 0x00, 0x01, 0x00, 0x04, 0x00, 0x00, 0x01, 0x00, + 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x00, 0x00, 0x11, 0xd4, 0x00, + 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, + 0x8e, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, + 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, + 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, + 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x01, + 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01, + 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, + 0x19, 0x00, 0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, + 0x00, 0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, + 0x01, 0x01, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, 0x01, + 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, + 0x26, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, + 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, + 0x00, 0x00, 0x01, 0x00, 0x01, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, + 0x04, 0x00, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, + 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, + 0x00, 0x00, 0x00, 0x11, 0xd4, 0x00, 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x01, 0x04, 0x00, 0x00, + 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, + 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, + 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, + 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0xf9, 0x01, 0x00, + 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, + 0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, + 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x01}; + RtpPacketizerH264 packetizer(long_input, limits, + H264PacketizationMode::NonInterleaved); + EXPECT_EQ(packetizer.NumPackets(), 2u); + EXPECT_THAT(FetchAllPackets(&packetizer), SizeIs(2)); +} + } // namespace } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc index ae5f4e50a4..34b3fd9d2f 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc @@ -63,7 +63,9 @@ RtpPacketizerVp8::RtpPacketizerVp8(rtc::ArrayView payload, const RTPVideoHeaderVP8& hdr_info) : hdr_(BuildHeader(hdr_info)), remaining_payload_(payload) { limits.max_payload_len -= hdr_.size(); - payload_sizes_ = SplitAboutEqually(payload.size(), limits); + if (!payload.empty()) { + payload_sizes_ = SplitAboutEqually(payload.size(), limits); + } current_packet_ = payload_sizes_.begin(); } diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc index 7934ff8ea9..cab7fc4422 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc @@ -21,6 +21,18 @@ namespace { constexpr RtpPacketizer::PayloadSizeLimits kNoSizeLimits; +TEST(RtpPacketizerVp8Test, EmptyPayload) { + RTPVideoHeaderVP8 hdr_info; + hdr_info.InitRTPVideoHeaderVP8(); + hdr_info.pictureId = 200; + RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30); + + RtpPacketizer::PayloadSizeLimits limits; + limits.max_payload_len = 12; // Small enough to produce 4 packets. + RtpPacketizerVp8 packetizer({}, limits, hdr_info); + EXPECT_EQ(packetizer.NumPackets(), 0u); +} + TEST(RtpPacketizerVp8Test, ResultPacketsAreAlmostEqualSize) { RTPVideoHeaderVP8 hdr_info; hdr_info.InitRTPVideoHeaderVP8(); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc index 9ad4aa97c3..5e037859ce 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc @@ -321,8 +321,9 @@ RtpPacketizerVp9::RtpPacketizerVp9(rtc::ArrayView payload, limits.max_payload_len -= header_size_; limits.first_packet_reduction_len += first_packet_extra_header_size_; limits.single_packet_reduction_len += first_packet_extra_header_size_; - - payload_sizes_ = SplitAboutEqually(payload.size(), limits); + if (!payload.empty()) { + payload_sizes_ = SplitAboutEqually(payload.size(), limits); + } current_packet_ = payload_sizes_.begin(); } diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc index e18b8a803f..948bcf3bdb 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc @@ -186,6 +186,11 @@ class RtpPacketizerVp9Test : public ::testing::Test { } }; +TEST_F(RtpPacketizerVp9Test, EmptyPayload) { + RTPVideoHeader video_header; + VideoRtpDepacketizerVp9::ParseRtpPayload({}, &video_header); +} + TEST_F(RtpPacketizerVp9Test, TestEqualSizedMode_OnePacket) { const size_t kFrameSize = 25; const size_t kPacketSize = 26; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h index e91ec6368b..c002e51de6 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h @@ -11,6 +11,7 @@ #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_ #include +#include #include #include "absl/types/optional.h" @@ -127,7 +128,7 @@ class RtpPacket { bool IsRegistered() const; template - bool GetExtension(FirstValue, Values...) const; + bool GetExtension(FirstValue&&, Values&&...) const; template absl::optional GetExtension() const; @@ -231,11 +232,12 @@ bool RtpPacket::IsRegistered() const { } template -bool RtpPacket::GetExtension(FirstValue first, Values... values) const { +bool RtpPacket::GetExtension(FirstValue&& first, Values&&... values) const { auto raw = FindExtension(Extension::kId); if (raw.empty()) return false; - return Extension::Parse(raw, first, values...); + return Extension::Parse(raw, std::forward(first), + std::forward(values)...); } template diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc index b55e74aaf0..691a243c5f 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc @@ -12,6 +12,8 @@ #include +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" + namespace webrtc { RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions) @@ -28,4 +30,13 @@ RtpPacketToSend& RtpPacketToSend::operator=(RtpPacketToSend&& packet) = default; RtpPacketToSend::~RtpPacketToSend() = default; +void RtpPacketToSend::set_packet_type(RtpPacketMediaType type) { + if (packet_type_ == RtpPacketMediaType::kAudio) { + original_packet_type_ = OriginalType::kAudio; + } else if (packet_type_ == RtpPacketMediaType::kVideo) { + original_packet_type_ = OriginalType::kVideo; + } + packet_type_ = type; +} + } // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h index 438ca354ed..64f9ee1ab1 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h @@ -49,11 +49,18 @@ class RtpPacketToSend : public RtpPacket { webrtc::Timestamp capture_time() const { return capture_time_; } void set_capture_time(webrtc::Timestamp time) { capture_time_ = time; } - void set_packet_type(RtpPacketMediaType type) { packet_type_ = type; } + void set_packet_type(RtpPacketMediaType type); + absl::optional packet_type() const { return packet_type_; } + enum class OriginalType { kAudio, kVideo }; + // Original type does not change if packet type is changed to kRetransmission. + absl::optional original_packet_type() const { + return original_packet_type_; + } + // If this is a retransmission, indicates the sequence number of the original // media packet that this packet represents. If RTX is used this will likely // be different from SequenceNumber(). @@ -133,6 +140,7 @@ class RtpPacketToSend : public RtpPacket { private: webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero(); absl::optional packet_type_; + absl::optional original_packet_type_; bool allow_retransmission_ = false; absl::optional retransmitted_sequence_number_; rtc::scoped_refptr additional_data_; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc index b3a9452df9..44f1a9e742 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc @@ -933,6 +933,41 @@ TEST(RtpPacketTest, GetUncopyableExtension) { EXPECT_TRUE(rtp_packet.GetExtension(&value2)); } +struct ParseByReferenceExtension { + static constexpr RTPExtensionType kId = kRtpExtensionDependencyDescriptor; + static constexpr absl::string_view Uri() { return "uri"; } + + static size_t ValueSize(uint8_t value1, uint8_t value2) { return 2; } + static bool Write(rtc::ArrayView data, + uint8_t value1, + uint8_t value2) { + data[0] = value1; + data[1] = value2; + return true; + } + static bool Parse(rtc::ArrayView data, + uint8_t& value1, + uint8_t& value2) { + value1 = data[0]; + value2 = data[1]; + return true; + } +}; + +TEST(RtpPacketTest, GetExtensionByReference) { + RtpHeaderExtensionMap extensions; + extensions.Register(1); + RtpPacket rtp_packet(&extensions); + rtp_packet.SetExtension(13, 42); + + uint8_t value1 = 1; + uint8_t value2 = 1; + EXPECT_TRUE( + rtp_packet.GetExtension(value1, value2)); + EXPECT_EQ(int{value1}, 13); + EXPECT_EQ(int{value2}, 42); +} + TEST(RtpPacketTest, CreateAndParseTimingFrameExtension) { // Create a packet with video frame timing extension populated. RtpPacketToSend::ExtensionManager send_extensions; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc index 2151a59295..83a2be24ea 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc @@ -99,6 +99,12 @@ Av1Frame ReassembleFrame(rtc::ArrayView rtp_payloads) { return Av1Frame(VideoRtpDepacketizerAv1().AssembleFrame(payloads)); } +TEST(RtpPacketizerAv1Test, EmptyPayload) { + RtpPacketizer::PayloadSizeLimits limits; + RtpPacketizerAv1 packetizer({}, limits, VideoFrameType::kVideoFrameKey, true); + EXPECT_EQ(packetizer.NumPackets(), 0u); +} + TEST(RtpPacketizerAv1Test, PacketizeOneObuWithoutSizeAndExtension) { auto kFrame = BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame) .WithoutSize() -- cgit v1.2.3