From 67008dfb366237469fe088a61b62c0cad852c024 Mon Sep 17 00:00:00 2001 From: Artem Titov Date: Thu, 4 Jul 2019 07:14:11 +0000 Subject: [PATCH] Revert "Replace the implementation of `GetContributingSources()` on the audio side." This reverts commit 8fa7151e4bbad40fec1f964fe0c003b8787bb78a. Reason for revert: Speculative revert to fix roll of webrtc into chrome. Right now tests related to RTCRtpReceiver failing and looks like it is main candidate, who can affect that behavior. Original change's description: > Replace the implementation of `GetContributingSources()` on the audio side. > > This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the audio side with the spec-compliant `SourceTracker`-implementation. > > The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network. > > This change is almost identical to the previous video side change at: https://webrtc-review.googlesource.com/c/src/+/143177 > > Bug: webrtc:10545 > Change-Id: Ife7f08ee8ca1346099b7466837a3756947085fc5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144422 > Reviewed-by: Oskar Sundbom > Commit-Queue: Chen Xing > Cr-Commit-Position: refs/heads/master@{#28459} TBR=ossu@webrtc.org,chxg@google.com Change-Id: I5c631d4dcfb39601055ffce9b104f45eea871fd3 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10545 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144562 Reviewed-by: Artem Titov Commit-Queue: Artem Titov Cr-Commit-Position: refs/heads/master@{#28478} --- audio/audio_receive_stream.cc | 13 +++------- audio/audio_receive_stream.h | 2 -- audio/channel_receive.cc | 49 ++++++++++++++++++++++++++++++----- audio/channel_receive.h | 2 ++ 4 files changed, 47 insertions(+), 19 deletions(-) diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 1a55adbe46..0ff2b0c0e3 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -113,9 +113,7 @@ AudioReceiveStream::AudioReceiveStream( const rtc::scoped_refptr& audio_state, webrtc::RtcEventLog* event_log, std::unique_ptr channel_receive) - : audio_state_(audio_state), - channel_receive_(std::move(channel_receive)), - source_tracker_(clock) { + : audio_state_(audio_state), channel_receive_(std::move(channel_receive)) { RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc; RTC_DCHECK(config.decoder_factory); RTC_DCHECK(config.rtcp_send_transport); @@ -269,18 +267,13 @@ int AudioReceiveStream::GetBaseMinimumPlayoutDelayMs() const { std::vector AudioReceiveStream::GetSources() const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); - return source_tracker_.GetSources(); + return channel_receive_->GetSources(); } AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo( int sample_rate_hz, AudioFrame* audio_frame) { - AudioMixer::Source::AudioFrameInfo audio_frame_info = - channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame); - if (audio_frame_info != AudioMixer::Source::AudioFrameInfo::kError) { - source_tracker_.OnFrameDelivered(audio_frame->packet_infos_); - } - return audio_frame_info; + return channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame); } int AudioReceiveStream::Ssrc() const { diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h index 49969a2779..0924c03d5c 100644 --- a/audio/audio_receive_stream.h +++ b/audio/audio_receive_stream.h @@ -19,7 +19,6 @@ #include "audio/audio_state.h" #include "call/audio_receive_stream.h" #include "call/syncable.h" -#include "modules/rtp_rtcp/source/source_tracker.h" #include "rtc_base/constructor_magic.h" #include "rtc_base/thread_checker.h" #include "system_wrappers/include/clock.h" @@ -108,7 +107,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, webrtc::AudioReceiveStream::Config config_; rtc::scoped_refptr audio_state_; const std::unique_ptr channel_receive_; - SourceTracker source_tracker_; AudioSendStream* associated_send_stream_ = nullptr; bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false; diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index 7567e3b2e3..8b9dd2d7f2 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc @@ -30,6 +30,7 @@ #include "modules/rtp_rtcp/include/receive_statistics.h" #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h" #include "modules/rtp_rtcp/include/rtp_rtcp.h" +#include "modules/rtp_rtcp/source/contributing_sources.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" @@ -154,6 +155,8 @@ class ChannelReceive : public ChannelReceiveInterface, // Used for obtaining RTT for a receive-only channel. void SetAssociatedSendChannel(const ChannelSendInterface* channel) override; + std::vector GetSources() const override; + // TODO(sukhanov): Return const pointer. It requires making media transport // getters like GetLatestTargetTransferRate to be also const. MediaTransportInterface* media_transport() const { @@ -210,13 +213,16 @@ class ChannelReceive : public ChannelReceiveInterface, std::unique_ptr _rtpRtcpModule; const uint32_t remote_ssrc_; - // Info for GetSyncInfo is updated on network or worker thread, and queried on - // the worker thread. - rtc::CriticalSection sync_info_lock_; + // Info for GetSources and GetSyncInfo is updated on network or worker thread, + // queried on the worker thread. + rtc::CriticalSection rtp_sources_lock_; + ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_); absl::optional last_received_rtp_timestamp_ - RTC_GUARDED_BY(&sync_info_lock_); + RTC_GUARDED_BY(&rtp_sources_lock_); absl::optional last_received_rtp_system_time_ms_ - RTC_GUARDED_BY(&sync_info_lock_); + RTC_GUARDED_BY(&rtp_sources_lock_); + absl::optional last_received_rtp_audio_level_ + RTC_GUARDED_BY(&rtp_sources_lock_); std::unique_ptr audio_coding_; AudioSinkInterface* audio_sink_ = nullptr; @@ -549,6 +555,24 @@ absl::optional> return audio_coding_->ReceiveCodec(); } +std::vector ChannelReceive::GetSources() const { + RTC_DCHECK(worker_thread_checker_.IsCurrent()); + int64_t now_ms = rtc::TimeMillis(); + std::vector sources; + { + rtc::CritScope cs(&rtp_sources_lock_); + sources = contributing_sources_.GetSources(now_ms); + if (last_received_rtp_system_time_ms_ >= + now_ms - ContributingSources::kHistoryMs) { + RTC_DCHECK(last_received_rtp_timestamp_.has_value()); + sources.emplace_back(*last_received_rtp_system_time_ms_, remote_ssrc_, + RtpSourceType::SSRC, last_received_rtp_audio_level_, + *last_received_rtp_timestamp_); + } + } + return sources; +} + void ChannelReceive::SetReceiveCodecs( const std::map& codecs) { RTC_DCHECK(worker_thread_checker_.IsCurrent()); @@ -562,11 +586,22 @@ void ChannelReceive::SetReceiveCodecs( // May be called on either worker thread or network thread. void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) { int64_t now_ms = rtc::TimeMillis(); + uint8_t audio_level; + bool voice_activity; + bool has_audio_level = + packet.GetExtension<::webrtc::AudioLevel>(&voice_activity, &audio_level); { - rtc::CritScope cs(&sync_info_lock_); + rtc::CritScope cs(&rtp_sources_lock_); last_received_rtp_timestamp_ = packet.Timestamp(); last_received_rtp_system_time_ms_ = now_ms; + if (has_audio_level) + last_received_rtp_audio_level_ = audio_level; + std::vector csrcs = packet.Csrcs(); + contributing_sources_.Update( + now_ms, csrcs, + has_audio_level ? absl::optional(audio_level) : absl::nullopt, + packet.Timestamp()); } // Store playout timestamp for the received RTP packet @@ -840,7 +875,7 @@ absl::optional ChannelReceive::GetSyncInfo() const { return absl::nullopt; } { - rtc::CritScope cs(&sync_info_lock_); + rtc::CritScope cs(&rtp_sources_lock_); if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) { return absl::nullopt; } diff --git a/audio/channel_receive.h b/audio/channel_receive.h index 0780a63692..1b0c81c314 100644 --- a/audio/channel_receive.h +++ b/audio/channel_receive.h @@ -135,6 +135,8 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface { // Used for obtaining RTT for a receive-only channel. virtual void SetAssociatedSendChannel( const ChannelSendInterface* channel) = 0; + + virtual std::vector GetSources() const = 0; }; std::unique_ptr CreateChannelReceive(