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From: Michael Froman <mfroman@mozilla.com>
Date: Wed, 3 May 2023 14:41:00 +0000
Subject: Bug 1685245 - cherry pick upstream libwebrtc commit 6aba07e5fe. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D176944
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0a46882336b7e5e97b54492e361f4bd9b33f8a39
---
modules/rtp_rtcp/source/rtp_sender.cc | 33 +++++++++
modules/rtp_rtcp/source/rtp_sender.h | 3 +
modules/rtp_rtcp/source/rtp_sender_video.cc | 26 +++----
.../source/rtp_sender_video_unittest.cc | 67 +++++++++++++++++++
4 files changed, 117 insertions(+), 12 deletions(-)
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 336a117f4e..d5e8bdcccb 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -558,6 +558,39 @@ std::unique_ptr<RtpPacketToSend> 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/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 55dee7f219..b49afe0dec 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -106,6 +106,9 @@ class RTPSender {
absl::optional<uint32_t> 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/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index e1ac4e41c3..99a00025c1 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/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<Timestamp> capture_time;
if (capture_time_ms > 0) {
@@ -652,14 +662,6 @@ bool RTPSenderVideo::SendVideo(
std::unique_ptr<RtpPacketizer> 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/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 72dfd0238d..d6fbba7bd8 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/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<RTPVideoHeaderVP8>();
+ 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<const uint8_t> 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];
--
2.34.1
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