From 175f06f112a7b896a6789236037a1dd46d99a77e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Henrik=20Bostr=C3=B6m?= Date: Thu, 5 Jan 2023 08:53:16 +0100 Subject: [PATCH] Reland "Remove 'trackId' dependency in stats selector algorithm." MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is a reland of commit 81aab488781c1a736c9d85ff1532631be2989523 See diff between Patch Set 1 and latest Patch Set. The original CL broke this WPT[1] because getStats() with the receiver as the selector stopped working in the event of unsignalled SSRCs due to the receiver not knowing what the SSRC was. This fix is to query media_channel_ for the unsignalled SSRC in the event that the receiver does not know the SSRC. [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html Original change's description: > Remove 'trackId' dependency in stats selector algorithm. > > In preparation for the deletion of deprecated 'track' stats, the > stats selector algorithm needs to be rewritten not to use 'trackId'. > > This is achieved by finding RTP stats by their SSRC, as obtained via > getParameters(). This unfortunately adds a block-invoke (in the sender > case the block-invoke happens inside GetParametersInternal and in the > receiver case the block-invoke is explicit at the calling place), but > it can't be helped and it's just once per getStats() call and only if > the selector argument is used. > > Bug: webrtc:14175 > Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101 > Reviewed-by: Harald Alvestrand > Commit-Queue: Henrik Boström > Cr-Commit-Position: refs/heads/main@{#38981} Bug: webrtc:14175, webrtc:14811 Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600 Commit-Queue: Henrik Boström Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#39010} --- media/base/fake_media_engine.h | 3 ++ media/base/media_channel.h | 2 + media/base/media_channel_impl.h | 6 +++ media/engine/webrtc_video_engine.cc | 36 ++++++++--------- media/engine/webrtc_video_engine.h | 3 +- media/engine/webrtc_voice_engine.cc | 9 +++++ media/engine/webrtc_voice_engine.h | 1 + pc/audio_rtp_receiver.cc | 7 +++- pc/audio_rtp_receiver.h | 2 +- pc/legacy_stats_collector.cc | 2 +- pc/rtc_stats_collector.cc | 58 +++++++++++++--------------- pc/rtc_stats_collector.h | 6 +++ pc/rtc_stats_collector_unittest.cc | 8 +++- pc/rtp_receiver.h | 2 +- pc/test/mock_rtp_receiver_internal.h | 2 +- pc/test/mock_voice_media_channel.h | 4 ++ pc/video_rtp_receiver.cc | 7 +++- pc/video_rtp_receiver.h | 2 +- 18 files changed, 98 insertions(+), 62 deletions(-) diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h index ef7701400e..e1376c287b 100644 --- a/media/base/fake_media_engine.h +++ b/media/base/fake_media_engine.h @@ -121,6 +121,9 @@ class RtpHelper : public Base { return RemoveStreamBySsrc(&send_streams_, ssrc); } virtual void ResetUnsignaledRecvStream() {} + virtual absl::optional GetUnsignaledSsrc() const { + return absl::nullopt; + } virtual void OnDemuxerCriteriaUpdatePending() {} virtual void OnDemuxerCriteriaUpdateComplete() {} diff --git a/media/base/media_channel.h b/media/base/media_channel.h index 9dced9485e..30e7571ab8 100644 --- a/media/base/media_channel.h +++ b/media/base/media_channel.h @@ -282,6 +282,8 @@ class MediaReceiveChannelInterface // Resets any cached StreamParams for an unsignaled RecvStream, and removes // any existing unsignaled streams. virtual void ResetUnsignaledRecvStream() = 0; + // Gets the current unsignaled receive stream's SSRC, if there is one. + virtual absl::optional GetUnsignaledSsrc() const = 0; // This is currently a workaround because of the demuxer state being managed // across two separate threads. Once the state is consistently managed on // the same thread (network), this workaround can be removed. diff --git a/media/base/media_channel_impl.h b/media/base/media_channel_impl.h index b4263f4a86..91b91c118a 100644 --- a/media/base/media_channel_impl.h +++ b/media/base/media_channel_impl.h @@ -420,6 +420,9 @@ class VoiceMediaReceiveChannel : public VoiceMediaReceiveChannelInterface { void ResetUnsignaledRecvStream() override { return impl()->ResetUnsignaledRecvStream(); } + absl::optional GetUnsignaledSsrc() const override { + return impl()->GetUnsignaledSsrc(); + } void OnDemuxerCriteriaUpdatePending() override { impl()->OnDemuxerCriteriaUpdatePending(); } @@ -614,6 +617,9 @@ class VideoMediaReceiveChannel : public VideoMediaReceiveChannelInterface { void ResetUnsignaledRecvStream() override { return impl()->ResetUnsignaledRecvStream(); } + absl::optional GetUnsignaledSsrc() const override { + return impl()->GetUnsignaledSsrc(); + } void OnDemuxerCriteriaUpdatePending() override { impl()->OnDemuxerCriteriaUpdatePending(); } diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc index d023f21da2..cd80b3f027 100644 --- a/media/engine/webrtc_video_engine.cc +++ b/media/engine/webrtc_video_engine.cc @@ -548,8 +548,7 @@ UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc( WebRtcVideoChannel* channel, uint32_t ssrc, absl::optional rtx_ssrc) { - absl::optional default_recv_ssrc = - channel->GetDefaultReceiveStreamSsrc(); + absl::optional default_recv_ssrc = channel->GetUnsignaledSsrc(); if (default_recv_ssrc) { RTC_LOG(LS_INFO) << "Destroying old default receive stream for SSRC=" @@ -588,8 +587,7 @@ void DefaultUnsignalledSsrcHandler::SetDefaultSink( WebRtcVideoChannel* channel, rtc::VideoSinkInterface* sink) { default_sink_ = sink; - absl::optional default_recv_ssrc = - channel->GetDefaultReceiveStreamSsrc(); + absl::optional default_recv_ssrc = channel->GetUnsignaledSsrc(); if (default_recv_ssrc) { channel->SetSink(*default_recv_ssrc, default_sink_); } @@ -1566,6 +1564,18 @@ void WebRtcVideoChannel::ResetUnsignaledRecvStream() { } } +absl::optional WebRtcVideoChannel::GetUnsignaledSsrc() const { + RTC_DCHECK_RUN_ON(&thread_checker_); + absl::optional ssrc; + for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) { + if (it->second->IsDefaultStream()) { + ssrc.emplace(it->first); + break; + } + } + return ssrc; +} + void WebRtcVideoChannel::OnDemuxerCriteriaUpdatePending() { RTC_DCHECK_RUN_ON(&thread_checker_); ++demuxer_criteria_id_; @@ -1756,7 +1766,7 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet, // stream, which will be associated with unsignaled media stream. // It is not possible to update the ssrcs of a receive stream, so we // recreate it insead if found. - auto default_ssrc = GetDefaultReceiveStreamSsrc(); + auto default_ssrc = GetUnsignaledSsrc(); if (!default_ssrc) { return; } @@ -1919,7 +1929,7 @@ void WebRtcVideoChannel::SetVideoCodecSwitchingEnabled(bool enabled) { bool WebRtcVideoChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) { RTC_DCHECK_RUN_ON(&thread_checker_); - absl::optional default_ssrc = GetDefaultReceiveStreamSsrc(); + absl::optional default_ssrc = GetUnsignaledSsrc(); // SSRC of 0 represents the default receive stream. if (ssrc == 0) { @@ -1961,18 +1971,6 @@ absl::optional WebRtcVideoChannel::GetBaseMinimumPlayoutDelayMs( } } -absl::optional WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() { - RTC_DCHECK_RUN_ON(&thread_checker_); - absl::optional ssrc; - for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) { - if (it->second->IsDefaultStream()) { - ssrc.emplace(it->first); - break; - } - } - return ssrc; -} - std::vector WebRtcVideoChannel::GetSources( uint32_t ssrc) const { RTC_DCHECK_RUN_ON(&thread_checker_); @@ -3496,7 +3494,7 @@ WebRtcVideoChannel::MapCodecs(const std::vector& codecs) { WebRtcVideoChannel::WebRtcVideoReceiveStream* WebRtcVideoChannel::FindReceiveStream(uint32_t ssrc) { if (ssrc == 0) { - absl::optional default_ssrc = GetDefaultReceiveStreamSsrc(); + absl::optional default_ssrc = GetUnsignaledSsrc(); if (!default_ssrc) { return nullptr; } diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h index 8b7e4561bd..bf27defc92 100644 --- a/media/engine/webrtc_video_engine.h +++ b/media/engine/webrtc_video_engine.h @@ -165,6 +165,7 @@ class WebRtcVideoChannel : public VideoMediaChannel, bool AddRecvStream(const StreamParams& sp, bool default_stream); bool RemoveRecvStream(uint32_t ssrc) override; void ResetUnsignaledRecvStream() override; + absl::optional GetUnsignaledSsrc() const override; void OnDemuxerCriteriaUpdatePending() override; void OnDemuxerCriteriaUpdateComplete() override; bool SetSink(uint32_t ssrc, @@ -216,8 +217,6 @@ class WebRtcVideoChannel : public VideoMediaChannel, return sending_; } - absl::optional GetDefaultReceiveStreamSsrc(); - StreamParams unsignaled_stream_params() { RTC_DCHECK_RUN_ON(&thread_checker_); return unsignaled_stream_params_; diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index c87eca5fdf..694b8b9196 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -1966,6 +1966,15 @@ void WebRtcVoiceMediaChannel::ResetUnsignaledRecvStream() { } } +absl::optional WebRtcVoiceMediaChannel::GetUnsignaledSsrc() const { + if (unsignaled_recv_ssrcs_.empty()) { + return absl::nullopt; + } + // In the event of multiple unsignaled ssrcs, the last in the vector will be + // the most recent one (the one forwarded to the MediaStreamTrack). + return unsignaled_recv_ssrcs_.back(); +} + // Not implemented. // TODO(https://crbug.com/webrtc/12676): Implement a fix for the unsignalled // SSRC race that can happen when an m= section goes from receiving to not diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h index 35dad9f679..835be360e7 100644 --- a/media/engine/webrtc_voice_engine.h +++ b/media/engine/webrtc_voice_engine.h @@ -173,6 +173,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, bool AddRecvStream(const StreamParams& sp) override; bool RemoveRecvStream(uint32_t ssrc) override; void ResetUnsignaledRecvStream() override; + absl::optional GetUnsignaledSsrc() const override; void OnDemuxerCriteriaUpdatePending() override; void OnDemuxerCriteriaUpdateComplete() override; diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc index 7af460b80e..9e687b9b55 100644 --- a/pc/audio_rtp_receiver.cc +++ b/pc/audio_rtp_receiver.cc @@ -212,9 +212,12 @@ void AudioRtpReceiver::SetupUnsignaledMediaChannel() { RestartMediaChannel(absl::nullopt); } -uint32_t AudioRtpReceiver::ssrc() const { +absl::optional AudioRtpReceiver::ssrc() const { RTC_DCHECK_RUN_ON(worker_thread_); - return ssrc_.value_or(0); + if (!ssrc_.has_value() && media_channel_) { + return media_channel_->GetUnsignaledSsrc(); + } + return ssrc_; } void AudioRtpReceiver::set_stream_ids(std::vector stream_ids) { diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h index 2e0f77c85c..6ef8f61efd 100644 --- a/pc/audio_rtp_receiver.h +++ b/pc/audio_rtp_receiver.h @@ -100,7 +100,7 @@ class AudioRtpReceiver : public ObserverInterface, void Stop() override; void SetupMediaChannel(uint32_t ssrc) override; void SetupUnsignaledMediaChannel() override; - uint32_t ssrc() const override; + absl::optional ssrc() const override; void NotifyFirstPacketReceived() override; void set_stream_ids(std::vector stream_ids) override; void set_transport( diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc index 0df1957fcb..6829e359b8 100644 --- a/pc/legacy_stats_collector.cc +++ b/pc/legacy_stats_collector.cc @@ -1231,7 +1231,7 @@ void LegacyStatsCollector::ExtractMediaInfo( for (const auto& receiver : transceiver->internal()->receivers()) { gatherer->receiver_track_id_by_ssrc.insert(std::make_pair( - receiver->internal()->ssrc(), receiver->track()->id())); + receiver->internal()->ssrc().value_or(0), receiver->track()->id())); } } diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc index da86afa52d..78e7f4d96b 100644 --- a/pc/rtc_stats_collector.cc +++ b/pc/rtc_stats_collector.cc @@ -1249,7 +1249,10 @@ void ProduceReceiverMediaTrackStats( } } -rtc::scoped_refptr CreateReportFilteredBySelector( +} // namespace + +rtc::scoped_refptr +RTCStatsCollector::CreateReportFilteredBySelector( bool filter_by_sender_selector, rtc::scoped_refptr report, rtc::scoped_refptr sender_selector, @@ -1258,40 +1261,35 @@ rtc::scoped_refptr CreateReportFilteredBySelector( if (filter_by_sender_selector) { // Filter mode: RTCStatsCollector::RequestInfo::kSenderSelector if (sender_selector) { - // Find outbound-rtp(s) of the sender, i.e. the outbound-rtp(s) that - // reference the sender stats. - // Because we do not implement sender stats, we look at outbound-rtp(s) - // that reference the track attachment stats for the sender instead. - std::string track_id = - DEPRECATED_RTCMediaStreamTrackStatsIDFromDirectionAndAttachment( - kDirectionOutbound, sender_selector->AttachmentId()); - for (const auto& stats : *report) { - if (stats.type() != RTCOutboundRTPStreamStats::kType) - continue; - const auto& outbound_rtp = stats.cast_to(); - if (outbound_rtp.track_id.is_defined() && - *outbound_rtp.track_id == track_id) { - rtpstream_ids.push_back(outbound_rtp.id()); + // Find outbound-rtp(s) of the sender using ssrc lookup. + auto encodings = sender_selector->GetParametersInternal().encodings; + for (const auto* outbound_rtp : + report->GetStatsOfType()) { + RTC_DCHECK(outbound_rtp->ssrc.is_defined()); + auto it = std::find_if( + encodings.begin(), encodings.end(), + [ssrc = + *outbound_rtp->ssrc](const RtpEncodingParameters& encoding) { + return encoding.ssrc.has_value() && encoding.ssrc.value() == ssrc; + }); + if (it != encodings.end()) { + rtpstream_ids.push_back(outbound_rtp->id()); } } } } else { // Filter mode: RTCStatsCollector::RequestInfo::kReceiverSelector if (receiver_selector) { - // Find inbound-rtp(s) of the receiver, i.e. the inbound-rtp(s) that - // reference the receiver stats. - // Because we do not implement receiver stats, we look at inbound-rtp(s) - // that reference the track attachment stats for the receiver instead. - std::string track_id = - DEPRECATED_RTCMediaStreamTrackStatsIDFromDirectionAndAttachment( - kDirectionInbound, receiver_selector->AttachmentId()); - for (const auto& stats : *report) { - if (stats.type() != RTCInboundRTPStreamStats::kType) - continue; - const auto& inbound_rtp = stats.cast_to(); - if (inbound_rtp.track_id.is_defined() && - *inbound_rtp.track_id == track_id) { - rtpstream_ids.push_back(inbound_rtp.id()); + // Find the inbound-rtp of the receiver using ssrc lookup. + absl::optional ssrc; + worker_thread_->BlockingCall([&] { ssrc = receiver_selector->ssrc(); }); + if (ssrc.has_value()) { + for (const auto* inbound_rtp : + report->GetStatsOfType()) { + RTC_DCHECK(inbound_rtp->ssrc.is_defined()); + if (*inbound_rtp->ssrc == *ssrc) { + rtpstream_ids.push_back(inbound_rtp->id()); + } } } } @@ -1301,8 +1299,6 @@ rtc::scoped_refptr CreateReportFilteredBySelector( return TakeReferencedStats(report->Copy(), rtpstream_ids); } -} // namespace - RTCStatsCollector::CertificateStatsPair RTCStatsCollector::CertificateStatsPair::Copy() const { CertificateStatsPair copy; diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h index b3e60ef379..be366140c2 100644 --- a/pc/rtc_stats_collector.h +++ b/pc/rtc_stats_collector.h @@ -244,6 +244,12 @@ class RTCStatsCollector : public rtc::RefCountInterface, // This is a NO-OP if `network_report_` is null. void MergeNetworkReport_s(); + rtc::scoped_refptr CreateReportFilteredBySelector( + bool filter_by_sender_selector, + rtc::scoped_refptr report, + rtc::scoped_refptr sender_selector, + rtc::scoped_refptr receiver_selector); + // Slots for signals (sigslot) that are wired up to `pc_`. void OnSctpDataChannelCreated(SctpDataChannel* channel); // Slots for signals (sigslot) that are wired up to `channel`. diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc index 1916b92cb8..b75a15f322 100644 --- a/pc/rtc_stats_collector_unittest.cc +++ b/pc/rtc_stats_collector_unittest.cc @@ -388,7 +388,10 @@ rtc::scoped_refptr CreateMockSender( EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track)); EXPECT_CALL(*sender, ssrc()).WillRepeatedly(Return(ssrc)); EXPECT_CALL(*sender, media_type()).WillRepeatedly(Return(media_type)); - EXPECT_CALL(*sender, GetParameters()).WillRepeatedly(Invoke([ssrc]() { + EXPECT_CALL(*sender, GetParameters()) + .WillRepeatedly( + Invoke([s = sender.get()]() { return s->GetParametersInternal(); })); + EXPECT_CALL(*sender, GetParametersInternal()).WillRepeatedly(Invoke([ssrc]() { RtpParameters params; params.encodings.push_back(RtpEncodingParameters()); params.encodings[0].ssrc = ssrc; @@ -406,6 +409,9 @@ rtc::scoped_refptr CreateMockReceiver( int attachment_id) { auto receiver = rtc::make_ref_counted(); EXPECT_CALL(*receiver, track()).WillRepeatedly(Return(track)); + EXPECT_CALL(*receiver, ssrc()).WillRepeatedly(Invoke([ssrc]() { + return ssrc; + })); EXPECT_CALL(*receiver, streams()) .WillRepeatedly( Return(std::vector>({}))); diff --git a/pc/rtp_receiver.h b/pc/rtp_receiver.h index 7622139f83..16ab011f14 100644 --- a/pc/rtp_receiver.h +++ b/pc/rtp_receiver.h @@ -68,7 +68,7 @@ class RtpReceiverInternal : public RtpReceiverInterface { rtc::scoped_refptr dtls_transport) = 0; // This SSRC is used as an identifier for the receiver between the API layer // and the WebRtcVideoEngine, WebRtcVoiceEngine layer. - virtual uint32_t ssrc() const = 0; + virtual absl::optional ssrc() const = 0; // Call this to notify the RtpReceiver when the first packet has been received // on the corresponding channel. diff --git a/pc/test/mock_rtp_receiver_internal.h b/pc/test/mock_rtp_receiver_internal.h index e2a81c0dd3..e76b56755d 100644 --- a/pc/test/mock_rtp_receiver_internal.h +++ b/pc/test/mock_rtp_receiver_internal.h @@ -63,7 +63,7 @@ class MockRtpReceiverInternal : public RtpReceiverInternal { (override)); MOCK_METHOD(void, SetupMediaChannel, (uint32_t), (override)); MOCK_METHOD(void, SetupUnsignaledMediaChannel, (), (override)); - MOCK_METHOD(uint32_t, ssrc, (), (const, override)); + MOCK_METHOD(absl::optional, ssrc, (), (const, override)); MOCK_METHOD(void, NotifyFirstPacketReceived, (), (override)); MOCK_METHOD(void, set_stream_ids, (std::vector), (override)); MOCK_METHOD(void, diff --git a/pc/test/mock_voice_media_channel.h b/pc/test/mock_voice_media_channel.h index e89e7e7892..e01e235a6f 100644 --- a/pc/test/mock_voice_media_channel.h +++ b/pc/test/mock_voice_media_channel.h @@ -53,6 +53,10 @@ class MockVoiceMediaChannel : public VoiceMediaChannel { MOCK_METHOD(bool, AddRecvStream, (const StreamParams& sp), (override)); MOCK_METHOD(bool, RemoveRecvStream, (uint32_t ssrc), (override)); MOCK_METHOD(void, ResetUnsignaledRecvStream, (), (override)); + MOCK_METHOD(absl::optional, + GetUnsignaledSsrc, + (), + (const, override)); MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override)); MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override)); MOCK_METHOD(int, GetRtpSendTimeExtnId, (), (const, override)); diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc index 18dfc82a2e..e7e7726ab0 100644 --- a/pc/video_rtp_receiver.cc +++ b/pc/video_rtp_receiver.cc @@ -184,9 +184,12 @@ void VideoRtpReceiver::SetupUnsignaledMediaChannel() { RestartMediaChannel(absl::nullopt); } -uint32_t VideoRtpReceiver::ssrc() const { +absl::optional VideoRtpReceiver::ssrc() const { RTC_DCHECK_RUN_ON(worker_thread_); - return ssrc_.value_or(0); + if (!ssrc_.has_value() && media_channel_) { + return media_channel_->GetUnsignaledSsrc(); + } + return ssrc_; } void VideoRtpReceiver::set_stream_ids(std::vector stream_ids) { diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h index 086246daae..caf035e36f 100644 --- a/pc/video_rtp_receiver.h +++ b/pc/video_rtp_receiver.h @@ -89,7 +89,7 @@ class VideoRtpReceiver : public RtpReceiverInternal { void Stop() override; void SetupMediaChannel(uint32_t ssrc) override; void SetupUnsignaledMediaChannel() override; - uint32_t ssrc() const override; + absl::optional ssrc() const override; void NotifyFirstPacketReceived() override; void set_stream_ids(std::vector stream_ids) override; void set_transport(