Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
Reason for revert:
Identified a configuration problem in the video quality tests. Intend to fix and reland.
Original issue's description:
> Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
>
> Reason for revert:
> This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec
>
> Original issue's description:
> > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
> >
> > Reason for revert:
> > Intend to fix perf failures and reland.
> >
> > Original issue's description:
> > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
> > >
> > > Reason for revert:
> > > A few perf tests broken, including
> > >
> > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
> > >
> > >
> > > Original issue's description:
> > > > Use RtxReceiveStream.
> > > >
> > > > This also has the beneficial side-effect that when a media stream
> > > > which is protected by FlexFEC receives an RTX retransmission, the
> > > > retransmitted media packet is passed into the FlexFEC machinery,
> > > > which should improve its ability to recover packets via FEC.
> > > >
> > > > BUG=webrtc:7135
> > > >
> > > > Review-Url: https://codereview.webrtc.org/3008773002
> > > > Cr-Commit-Position: refs/heads/master@{#19649}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7
> > >
> > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7135
> > >
> > > Review-Url: https://codereview.webrtc.org/3010983002
> > > Cr-Commit-Position: refs/heads/master@{#19653}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b
> >
> > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/3006063002
> > Cr-Commit-Position: refs/heads/master@{#19715}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78
>
> TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3007303002
> Cr-Commit-Position: refs/heads/master@{#19744}
> Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107
TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3012963002
Cr-Original-Commit-Position: refs/heads/master@{#19765}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: ca5706d8b5576ba48eb34a7b4366a2aa6f164d0e
diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc
index ed44dad..2b85452 100644
--- a/call/rampup_tests.cc
+++ b/call/rampup_tests.cc
@@ -207,8 +207,9 @@
recv_config.rtp.ulpfec.ulpfec_payload_type =
send_config->rtp.ulpfec.ulpfec_payload_type;
if (rtx_) {
- recv_config.rtp.ulpfec.red_rtx_payload_type =
- send_config->rtp.ulpfec.red_rtx_payload_type;
+ recv_config.rtp.rtx_associated_payload_types
+ [send_config->rtp.ulpfec.red_rtx_payload_type] =
+ send_config->rtp.ulpfec.red_payload_type;
}
}
diff --git a/call/rtx_receive_stream.cc b/call/rtx_receive_stream.cc
index 1646352..6a5432f 100644
--- a/call/rtx_receive_stream.cc
+++ b/call/rtx_receive_stream.cc
@@ -11,17 +11,21 @@
#include <utility>
#include "webrtc/call/rtx_receive_stream.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/rtc_base/logging.h"
namespace webrtc {
-RtxReceiveStream::RtxReceiveStream(RtpPacketSinkInterface* media_sink,
- std::map<int, int> associated_payload_types,
- uint32_t media_ssrc)
+RtxReceiveStream::RtxReceiveStream(
+ RtpPacketSinkInterface* media_sink,
+ std::map<int, int> associated_payload_types,
+ uint32_t media_ssrc,
+ ReceiveStatistics* rtp_receive_statistics /* = nullptr */)
: media_sink_(media_sink),
associated_payload_types_(std::move(associated_payload_types)),
- media_ssrc_(media_ssrc) {
+ media_ssrc_(media_ssrc),
+ rtp_receive_statistics_(rtp_receive_statistics) {
if (associated_payload_types_.empty()) {
LOG(LS_WARNING)
<< "RtxReceiveStream created with empty payload type mapping.";
@@ -31,6 +35,12 @@
RtxReceiveStream::~RtxReceiveStream() = default;
void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
+ if (rtp_receive_statistics_) {
+ RTPHeader header;
+ rtx_packet.GetHeader(&header);
+ rtp_receive_statistics_->IncomingPacket(header, rtx_packet.size(),
+ false /* retransmitted */);
+ }
rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
if (payload.size() < kRtxHeaderSize) {
diff --git a/call/rtx_receive_stream.h b/call/rtx_receive_stream.h
index 418775c..c288a27 100644
--- a/call/rtx_receive_stream.h
+++ b/call/rtx_receive_stream.h
@@ -17,13 +17,20 @@
namespace webrtc {
+class ReceiveStatistics;
+
// This class is responsible for RTX decapsulation. The resulting media packets
// are passed on to a sink representing the associated media stream.
class RtxReceiveStream : public RtpPacketSinkInterface {
public:
RtxReceiveStream(RtpPacketSinkInterface* media_sink,
std::map<int, int> associated_payload_types,
- uint32_t media_ssrc);
+ uint32_t media_ssrc,
+ // TODO(nisse): Delete this argument, and
+ // corresponding member variable, by moving the
+ // responsibility for rtcp feedback to
+ // RtpStreamReceiverController.
+ ReceiveStatistics* rtp_receive_statistics = nullptr);
~RtxReceiveStream() override;
// RtpPacketSinkInterface.
void OnRtpPacket(const RtpPacketReceived& packet) override;
@@ -35,6 +42,7 @@
// TODO(nisse): Ultimately, the media receive stream shouldn't care about the
// ssrc, and we should delete this.
const uint32_t media_ssrc_;
+ ReceiveStatistics* const rtp_receive_statistics_;
};
} // namespace webrtc
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index e51184e..f199ca1 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -167,6 +167,11 @@
NackConfig nack;
// See UlpfecConfig for description.
+ // TODO(nisse): UlpfecConfig includes the field red_rtx_payload_type,
+ // which duplicates info in the rtx_associated_payload_types mapping. So
+ // delete the use of UlpfecConfig here, and replace by the values which
+ // make sense in this context, likely those are ulpfec_payload_type_ and
+ // red_payload_type_.
UlpfecConfig ulpfec;
// SSRC for retransmissions.
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index 9c0031a..a613dc6 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -2187,6 +2187,11 @@
config_.rtp.nack.rtp_history_ms =
HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
+ if (config_.rtp.ulpfec.red_rtx_payload_type != -1) {
+ config_.rtp
+ .rtx_associated_payload_types[config_.rtp.ulpfec.red_rtx_payload_type] =
+ config_.rtp.ulpfec.red_payload_type;
+ }
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureFlexfecCodec(
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index c6adfd3..1306082 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -83,6 +83,32 @@
return false;
}
+// TODO(nisse): Duplicated in call.cc.
+const int* FindKeyByValue(const std::map<int, int>& m, int v) {
+ for (const auto& kv : m) {
+ if (kv.second == v)
+ return &kv.first;
+ }
+ return nullptr;
+}
+
+bool HasRtxReceiveAssociation(
+ const webrtc::VideoReceiveStream::Config& config,
+ int payload_type) {
+ return FindKeyByValue(config.rtp.rtx_associated_payload_types,
+ payload_type) != nullptr;
+}
+
+// Check that there's an Rtx payload type for each decoder.
+bool VerifyRtxReceiveAssociations(
+ const webrtc::VideoReceiveStream::Config& config) {
+ for (const auto& decoder : config.decoders) {
+ if (!HasRtxReceiveAssociation(config, decoder.payload_type))
+ return false;
+ }
+ return true;
+}
+
rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer(
int width,
int height) {
@@ -112,15 +138,6 @@
return media_config;
}
-// TODO(nisse): Duplicated in call.cc.
-const int* FindKeyByValue(const std::map<int, int>& m, int v) {
- for (const auto& kv : m) {
- if (kv.second == v)
- return &kv.first;
- }
- return nullptr;
-}
-
} // namespace
namespace cricket {
@@ -1316,9 +1333,12 @@
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
EXPECT_FALSE(
recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
- EXPECT_EQ(recv_stream->GetConfig().decoders.size(),
- recv_stream->GetConfig().rtp.rtx_associated_payload_types.size())
+ EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
+ EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
+ GetEngineCodec("red").id))
+ << "RTX should be mapped for the RED payload type";
+
EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
}
@@ -1329,6 +1349,12 @@
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
FakeVideoReceiveStream* recv_stream = AddRecvStream(params);
EXPECT_EQ(kRtxSsrcs1[0], recv_stream->GetConfig().rtp.rtx_ssrc);
+
+ EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
+ << "RTX should be mapped for all decoders/payload types.";
+ EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
+ GetEngineCodec("red").id))
+ << "RTX should be mapped for the RED payload type";
}
TEST_F(WebRtcVideoChannelTest, RecvStreamNoRtx) {
@@ -3796,9 +3822,11 @@
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_FALSE(
recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
- EXPECT_EQ(recv_stream->GetConfig().decoders.size(),
- recv_stream->GetConfig().rtp.rtx_associated_payload_types.size())
+ EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
+ EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
+ GetEngineCodec("red").id))
+ << "RTX should be mapped also for the RED payload type";
EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
}
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 7756ff4..16f6bd0 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -61,6 +61,9 @@
"../call:call_interfaces",
"../call:rtp_interfaces",
"../call:video_stream_api",
+
+ # For RtxReceiveStream.
+ "../call:rtp_receiver",
"../common_video",
"../logging:rtc_event_log_api",
"../media:rtc_media_base",
diff --git a/video/end_to_end_tests.cc b/video/end_to_end_tests.cc
index ffd52fd..e92d8fe 100644
--- a/video/end_to_end_tests.cc
+++ b/video/end_to_end_tests.cc
@@ -1184,7 +1184,9 @@
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
(*receive_configs)[0]
- .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
+ .rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType)
+ ? kRtxRedPayloadType
+ : kSendRtxPayloadType] =
payload_type_;
}
// Configure encoding and decoding with VP8, since generic packetization
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index a47055a..1dbd869 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -86,6 +86,7 @@
RtcpRttStats* rtt_stats,
PacketRouter* packet_router,
const VideoReceiveStream::Config* config,
+ ReceiveStatistics* rtp_receive_statistics,
ReceiveStatisticsProxy* receive_stats_proxy,
ProcessThread* process_thread,
NackSender* nack_sender,
@@ -102,12 +103,11 @@
this,
this,
&rtp_payload_registry_)),
- rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
+ rtp_receive_statistics_(rtp_receive_statistics),
ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, this)),
receiving_(false),
- restored_packet_in_use_(false),
last_packet_log_ms_(-1),
- rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
+ rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_,
transport,
rtt_stats,
receive_stats_proxy,
@@ -146,12 +146,8 @@
rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
if (config_.rtp.rtx_ssrc) {
+ // Needed for rtp_payload_registry_.RtxEnabled().
rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc);
-
- for (const auto& kv : config_.rtp.rtx_associated_payload_types) {
- RTC_DCHECK_NE(kv.first, 0);
- rtp_payload_registry_.SetRtxPayloadType(kv.first, kv.second);
- }
}
if (IsUlpfecEnabled()) {
@@ -168,11 +164,6 @@
strncpy(red_codec.plName, "red", sizeof(red_codec.plName));
red_codec.plType = config_.rtp.ulpfec.red_payload_type;
RTC_CHECK(AddReceiveCodec(red_codec));
- if (config_.rtp.ulpfec.red_rtx_payload_type != -1) {
- rtp_payload_registry_.SetRtxPayloadType(
- config_.rtp.ulpfec.red_rtx_payload_type,
- config_.rtp.ulpfec.red_payload_type);
- }
}
if (config_.rtp.rtcp_xr.receiver_reference_time_report)
@@ -495,31 +486,7 @@
}
ulpfec_receiver_->ProcessReceivedFec();
} else if (rtp_payload_registry_.IsRtx(header)) {
- if (header.headerLength + header.paddingLength == packet_length) {
- // This is an empty packet and should be silently dropped before trying to
- // parse the RTX header.
- return;
- }
- // Remove the RTX header and parse the original RTP header.
- if (packet_length < header.headerLength)
- return;
- if (packet_length > sizeof(restored_packet_))
- return;
- if (restored_packet_in_use_) {
- LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
- return;
- }
- if (!rtp_payload_registry_.RestoreOriginalPacket(
- restored_packet_, packet, &packet_length, config_.rtp.remote_ssrc,
- header)) {
- LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
- << header.ssrc << " payload type: "
- << static_cast<int>(header.payloadType);
- return;
- }
- restored_packet_in_use_ = true;
- OnRecoveredPacket(restored_packet_, packet_length);
- restored_packet_in_use_ = false;
+ LOG(LS_WARNING) << "Unexpected RTX packet on media ssrc";
}
}
diff --git a/video/rtp_video_stream_receiver.h b/video/rtp_video_stream_receiver.h
index 3c8b2aa..994ef55 100644
--- a/video/rtp_video_stream_receiver.h
+++ b/video/rtp_video_stream_receiver.h
@@ -72,6 +72,7 @@
RtcpRttStats* rtt_stats,
PacketRouter* packet_router,
const VideoReceiveStream::Config* config,
+ ReceiveStatistics* rtp_receive_statistics,
ReceiveStatisticsProxy* receive_stats_proxy,
ProcessThread* process_thread,
NackSender* nack_sender,
@@ -180,13 +181,11 @@
const std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
const std::unique_ptr<RtpReceiver> rtp_receiver_;
- const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+ ReceiveStatistics* const rtp_receive_statistics_;
std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
rtc::SequencedTaskChecker worker_task_checker_;
bool receiving_ RTC_GUARDED_BY(worker_task_checker_);
- uint8_t restored_packet_[IP_PACKET_SIZE] RTC_GUARDED_BY(worker_task_checker_);
- bool restored_packet_in_use_ RTC_GUARDED_BY(worker_task_checker_);
int64_t last_packet_log_ms_ RTC_GUARDED_BY(worker_task_checker_);
const std::unique_ptr<RtpRtcp> rtp_rtcp_;
diff --git a/video/rtp_video_stream_receiver_unittest.cc b/video/rtp_video_stream_receiver_unittest.cc
index d309397..4f2dca7 100644
--- a/video/rtp_video_stream_receiver_unittest.cc
+++ b/video/rtp_video_stream_receiver_unittest.cc
@@ -126,11 +126,14 @@
process_thread_(ProcessThread::Create("TestThread")) {}
void SetUp() {
- rtp_video_stream_receiver_.reset(new RtpVideoStreamReceiver(
+ rtp_receive_statistics_ =
+ rtc::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
+ rtp_video_stream_receiver_ = rtc::MakeUnique<RtpVideoStreamReceiver>(
&mock_transport_, nullptr, &packet_router_, &config_,
- nullptr, process_thread_.get(), &mock_nack_sender_,
+ rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
+ &mock_nack_sender_,
&mock_key_frame_request_sender_, &mock_on_complete_frame_callback_,
- &timing_));
+ &timing_);
}
WebRtcRTPHeader GetDefaultPacket() {
@@ -196,6 +199,7 @@
PacketRouter packet_router_;
VCMTiming timing_;
std::unique_ptr<ProcessThread> process_thread_;
+ std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_;
};
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 11e0605..6294184 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -1555,6 +1555,9 @@
video_send_config_.rtp.ulpfec.ulpfec_payload_type;
it->rtp.ulpfec.red_rtx_payload_type =
video_send_config_.rtp.ulpfec.red_rtx_payload_type;
+ it->rtp.rtx_associated_payload_types[video_send_config_.rtp.ulpfec
+ .red_rtx_payload_type] =
+ video_send_config_.rtp.ulpfec.red_payload_type;
}
} else {
video_receive_configs_[params_.ss.selected_stream]
@@ -1566,6 +1569,10 @@
video_receive_configs_[params_.ss.selected_stream]
.rtp.ulpfec.red_rtx_payload_type =
video_send_config_.rtp.ulpfec.red_rtx_payload_type;
+ video_receive_configs_[params_.ss.selected_stream]
+ .rtp.rtx_associated_payload_types[video_send_config_.rtp.ulpfec
+ .red_rtx_payload_type] =
+ video_send_config_.rtp.ulpfec.red_payload_type;
}
}
}
diff --git a/video/video_receive_stream.cc b/video/video_receive_stream.cc
index 74c9013..a89e3ce 100644
--- a/video/video_receive_stream.cc
+++ b/video/video_receive_stream.cc
@@ -18,6 +18,7 @@
#include "webrtc/api/optional.h"
#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
+#include "webrtc/call/rtx_receive_stream.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/h264/profile_level_id.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@@ -32,6 +33,7 @@
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/location.h"
#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/trace_event.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/field_trial.h"
@@ -88,6 +90,7 @@
"DecodingThread",
rtc::kHighestPriority),
call_stats_(call_stats),
+ rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
timing_(new VCMTiming(clock_)),
video_receiver_(clock_, nullptr, this, timing_.get(), this, this),
stats_proxy_(&config_, clock_),
@@ -95,6 +98,7 @@
call_stats_->rtcp_rtt_stats(),
packet_router,
&config_,
+ rtp_receive_statistics_.get(),
&stats_proxy_,
process_thread_,
this, // NackSender
@@ -120,7 +124,7 @@
decoder_payload_types.insert(decoder.payload_type);
}
- video_receiver_.SetRenderDelay(config.render_delay_ms);
+ video_receiver_.SetRenderDelay(config_.render_delay_ms);
jitter_estimator_.reset(new VCMJitterEstimator(clock_));
frame_buffer_.reset(new video_coding::FrameBuffer(
@@ -131,9 +135,12 @@
// Register with RtpStreamReceiverController.
media_receiver_ = receiver_controller->CreateReceiver(
config_.rtp.remote_ssrc, &rtp_video_stream_receiver_);
- if (config.rtp.rtx_ssrc) {
+ if (config_.rtp.rtx_ssrc) {
+ rtx_receive_stream_ = rtc::MakeUnique<RtxReceiveStream>(
+ &rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types,
+ config_.rtp.remote_ssrc, rtp_receive_statistics_.get());
rtx_receiver_ = receiver_controller->CreateReceiver(
- config_.rtp.rtx_ssrc, &rtp_video_stream_receiver_);
+ config_.rtp.rtx_ssrc, rtx_receive_stream_.get());
}
}
diff --git a/video/video_receive_stream.h b/video/video_receive_stream.h
index 32ce812..8b680ae 100644
--- a/video/video_receive_stream.h
+++ b/video/video_receive_stream.h
@@ -38,6 +38,7 @@
class RTPFragmentationHeader;
class RtpStreamReceiverInterface;
class RtpStreamReceiverControllerInterface;
+class RtxReceiveStream;
class VCMTiming;
class VCMJitterEstimator;
@@ -125,6 +126,10 @@
CallStats* const call_stats_;
+ // Shared by media and rtx stream receivers, since the latter has no RtpRtcp
+ // module of its own.
+ const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+
std::unique_ptr<VCMTiming> timing_; // Jitter buffer experiment.
vcm::VideoReceiver video_receiver_;
std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
@@ -141,6 +146,7 @@
std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
+ std::unique_ptr<RtxReceiveStream> rtx_receive_stream_;
std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
// Whenever we are in an undecodable state (stream has just started or due to