diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc index 336a117f4e32..d5e8bdcccbd5 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc @@ -558,6 +558,39 @@ std::unique_ptr RTPSender::AllocatePacket() const { return packet; } +size_t RTPSender::RtxPacketOverhead() const { + MutexLock lock(&send_mutex_); + if (rtx_ == kRtxOff) { + return 0; + } + size_t overhead = 0; + + // Count space for the RTP header extensions that might need to be added to + // the RTX packet. + if (!always_send_mid_and_rid_ && (!rtx_ssrc_has_acked_ && ssrc_has_acked_)) { + // Prefer to reserve extra byte in case two byte header rtp header + // extensions are used. + static constexpr int kRtpExtensionHeaderSize = 2; + + // Rtx packets hasn't been acked and would need to have mid and rrsid rtp + // header extensions, while media packets no longer needs to include mid and + // rsid extensions. + if (!mid_.empty()) { + overhead += (kRtpExtensionHeaderSize + mid_.size()); + } + if (!rid_.empty()) { + overhead += (kRtpExtensionHeaderSize + rid_.size()); + } + // RTP header extensions are rounded up to 4 bytes. Depending on already + // present extensions adding mid & rrsid may add up to 3 bytes of padding. + overhead += 3; + } + + // Add two bytes for the original sequence number in the RTP payload. + overhead += kRtxHeaderSize; + return overhead; +} + void RTPSender::SetSendingMediaStatus(bool enabled) { MutexLock lock(&send_mutex_); sending_media_ = enabled; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.h index 55dee7f219aa..b49afe0dec42 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.h +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.h @@ -106,6 +106,9 @@ class RTPSender { absl::optional RtxSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) { return rtx_ssrc_; } + // Returns expected size difference between an RTX packet and media packet + // that RTX packet is created from. Returns 0 if RTX is disabled. + size_t RtxPacketOverhead() const; void SetRtxPayloadType(int payload_type, int associated_payload_type) RTC_LOCKS_EXCLUDED(send_mutex_); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc index e1ac4e41c3ac..99a00025c159 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -493,6 +493,13 @@ bool RTPSenderVideo::SendVideo( // Backward compatibility for older receivers without temporal layer logic. retransmission_settings = kRetransmitBaseLayer | kRetransmitHigherLayers; } + const uint8_t temporal_id = GetTemporalId(video_header); + // TODO(bugs.webrtc.org/10714): retransmission_settings_ should generally be + // replaced by expected_retransmission_time_ms.has_value(). + const bool allow_retransmission = + expected_retransmission_time_ms.has_value() && + AllowRetransmission(temporal_id, retransmission_settings, + *expected_retransmission_time_ms); MaybeUpdateCurrentPlayoutDelay(video_header); if (video_header.frame_type == VideoFrameType::kVideoFrameKey) { @@ -514,16 +521,19 @@ bool RTPSenderVideo::SendVideo( video_header.generic->frame_id, video_header.generic->chain_diffs); } - const uint8_t temporal_id = GetTemporalId(video_header); // No FEC protection for upper temporal layers, if used. const bool use_fec = fec_type_.has_value() && (temporal_id == 0 || temporal_id == kNoTemporalIdx); // Maximum size of packet including rtp headers. // Extra space left in case packet will be resent using fec or rtx. - int packet_capacity = rtp_sender_->MaxRtpPacketSize() - - (use_fec ? FecPacketOverhead() : 0) - - (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0); + int packet_capacity = rtp_sender_->MaxRtpPacketSize(); + if (use_fec) { + packet_capacity -= FecPacketOverhead(); + } + if (allow_retransmission) { + packet_capacity -= rtp_sender_->RtxPacketOverhead(); + } absl::optional capture_time; if (capture_time_ms > 0) { @@ -652,14 +662,6 @@ bool RTPSenderVideo::SendVideo( std::unique_ptr packetizer = RtpPacketizer::Create(codec_type, payload, limits, video_header); - // TODO(bugs.webrtc.org/10714): retransmission_settings_ should generally be - // replaced by expected_retransmission_time_ms.has_value(). For now, though, - // only VP8 with an injected frame buffer controller actually controls it. - const bool allow_retransmission = - expected_retransmission_time_ms.has_value() - ? AllowRetransmission(temporal_id, retransmission_settings, - expected_retransmission_time_ms.value()) - : false; const size_t num_packets = packetizer->NumPackets(); if (num_packets == 0) diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc index 72dfd0238d39..d6fbba7bd80b 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc @@ -29,6 +29,9 @@ #include "api/video/video_timing.h" #include "modules/rtp_rtcp/include/rtp_cvo.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" +#include "modules/rtp_rtcp/source/rtcp_packet/nack.h" +#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" +#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h" #include "modules/rtp_rtcp/source/rtp_format_video_generic.h" #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h" @@ -57,6 +60,7 @@ using ::testing::ElementsAre; using ::testing::ElementsAreArray; using ::testing::IsEmpty; using ::testing::NiceMock; +using ::testing::Not; using ::testing::Return; using ::testing::ReturnArg; using ::testing::SaveArg; @@ -81,6 +85,7 @@ constexpr VideoCodecType kType = VideoCodecType::kVideoCodecGeneric; constexpr uint32_t kTimestamp = 10; constexpr uint16_t kSeqNum = 33; constexpr uint32_t kSsrc = 725242; +constexpr uint32_t kRtxSsrc = 912364; constexpr int kMaxPacketLength = 1500; constexpr Timestamp kStartTime = Timestamp::Millis(123456789); constexpr int64_t kDefaultExpectedRetransmissionTimeMs = 125; @@ -182,6 +187,8 @@ class RtpSenderVideoTest : public ::testing::Test { config.retransmission_rate_limiter = &retransmission_rate_limiter_; config.field_trials = &field_trials_; config.local_media_ssrc = kSsrc; + config.rtx_send_ssrc = kRtxSsrc; + config.rid = "rid"; return config; }())), rtp_sender_video_( @@ -505,6 +512,66 @@ TEST_F(RtpSenderVideoTest, ConditionalRetransmitLimit) { rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs)); } +TEST_F(RtpSenderVideoTest, + ReservesEnoughSpaceForRtxPacketWhenMidAndRsidAreRegistered) { + constexpr int kMediaPayloadId = 100; + constexpr int kRtxPayloadId = 101; + constexpr size_t kMaxPacketSize = 1'000; + + rtp_module_->SetMaxRtpPacketSize(kMaxPacketSize); + rtp_module_->RegisterRtpHeaderExtension(RtpMid::Uri(), 1); + rtp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(), 2); + rtp_module_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(), 3); + rtp_module_->RegisterRtpHeaderExtension(AbsoluteSendTime::Uri(), 4); + rtp_module_->SetMid("long_mid"); + rtp_module_->SetRtxSendPayloadType(kRtxPayloadId, kMediaPayloadId); + rtp_module_->SetStorePacketsStatus(/*enable=*/true, 10); + rtp_module_->SetRtxSendStatus(kRtxRetransmitted); + + RTPVideoHeader header; + header.codec = kVideoCodecVP8; + header.frame_type = VideoFrameType::kVideoFrameDelta; + auto& vp8_header = header.video_type_header.emplace(); + vp8_header.temporalIdx = 0; + + uint8_t kPayload[kMaxPacketSize] = {}; + EXPECT_TRUE(rtp_sender_video_->SendVideo( + kMediaPayloadId, /*codec_type=*/kVideoCodecVP8, /*rtp_timestamp=*/0, + /*capture_time_ms=*/1'000, kPayload, header, + /*expected_retransmission_time_ms=*/absl::nullopt, /*csrcs=*/{})); + ASSERT_THAT(transport_.sent_packets(), Not(IsEmpty())); + // Ack media ssrc, but not rtx ssrc. + rtcp::ReceiverReport rr; + rtcp::ReportBlock rb; + rb.SetMediaSsrc(kSsrc); + rb.SetExtHighestSeqNum(transport_.last_sent_packet().SequenceNumber()); + rr.AddReportBlock(rb); + rtp_module_->IncomingRtcpPacket(rr.Build()); + + // Test for various frame size close to `kMaxPacketSize` to catch edge cases + // when rtx packet barely fit. + for (size_t frame_size = 800; frame_size < kMaxPacketSize; ++frame_size) { + SCOPED_TRACE(frame_size); + rtc::ArrayView payload(kPayload, frame_size); + + EXPECT_TRUE(rtp_sender_video_->SendVideo( + kMediaPayloadId, /*codec_type=*/kVideoCodecVP8, /*rtp_timestamp=*/0, + /*capture_time_ms=*/1'000, payload, header, + /*expected_retransmission_time_ms=*/1'000, /*csrcs=*/{})); + const RtpPacketReceived& media_packet = transport_.last_sent_packet(); + EXPECT_EQ(media_packet.Ssrc(), kSsrc); + + rtcp::Nack nack; + nack.SetMediaSsrc(kSsrc); + nack.SetPacketIds({media_packet.SequenceNumber()}); + rtp_module_->IncomingRtcpPacket(nack.Build()); + + const RtpPacketReceived& rtx_packet = transport_.last_sent_packet(); + EXPECT_EQ(rtx_packet.Ssrc(), kRtxSsrc); + EXPECT_LE(rtx_packet.size(), kMaxPacketSize); + } +} + TEST_F(RtpSenderVideoTest, SendsDependencyDescriptorWhenVideoStructureIsSet) { const int64_t kFrameId = 100000; uint8_t kFrame[100];