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Reland "Replace the implementation of GetContributingSources() on the audio side."

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This reverts commit 67008dfb366237469fe088a61b62c0cad852c024.

Reason for revert: Tests in the Chromium repo have been changed to accomodate this CL: https://chromium-review.googlesource.com/c/chromium/src/+/1728565

Original change's description:
> 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 <ossu@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > 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 <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28478}

TBR=ossu@webrtc.org,titovartem@webrtc.org,chxg@google.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10545
Change-Id: I609cca4f0ca4e1d31a156ba9eb44407518409f57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147865
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28746}
This commit is contained in:
Chen Xing 2019-08-02 10:29:26 +00:00 committed by Commit Bot
parent 59bbd65561
commit 054e3bbbe7
4 changed files with 19 additions and 47 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

13
audio/audio_receive_stream.cc
View File

@ -113,7 +113,9 @@ AudioReceiveStream::AudioReceiveStream(
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
: audio_state_(audio_state), channel_receive_(std::move(channel_receive)) {
: audio_state_(audio_state),
channel_receive_(std::move(channel_receive)),
source_tracker_(clock) {
RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
RTC_DCHECK(config.decoder_factory);
RTC_DCHECK(config.rtcp_send_transport);
@ -267,13 +269,18 @@ int AudioReceiveStream::GetBaseMinimumPlayoutDelayMs() const {
std::vector<RtpSource> AudioReceiveStream::GetSources() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_receive_->GetSources();
return source_tracker_.GetSources();
}
AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
return channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, 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;
}
int AudioReceiveStream::Ssrc() const {

2
audio/audio_receive_stream.h
View File

@ -19,6 +19,7 @@
#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"
@ -107,6 +108,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
SourceTracker source_tracker_;
AudioSendStream* associated_send_stream_ = nullptr;
bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;

49
audio/channel_receive.cc
View File

@ -30,7 +30,6 @@
#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"
@ -161,8 +160,6 @@ class ChannelReceive : public ChannelReceiveInterface,
// Used for obtaining RTT for a receive-only channel.
void SetAssociatedSendChannel(const ChannelSendInterface* channel) override;
std::vector<RtpSource> GetSources() const override;
// TODO(sukhanov): Return const pointer. It requires making media transport
// getters like GetLatestTargetTransferRate to be also const.
MediaTransportInterface* media_transport() const {
@ -219,16 +216,13 @@ class ChannelReceive : public ChannelReceiveInterface,
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
const uint32_t remote_ssrc_;
// 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_);
// Info for GetSyncInfo is updated on network or worker thread, and queried on
// the worker thread.
rtc::CriticalSection sync_info_lock_;
absl::optional<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(&rtp_sources_lock_);
RTC_GUARDED_BY(&sync_info_lock_);
absl::optional<int64_t> last_received_rtp_system_time_ms_
RTC_GUARDED_BY(&rtp_sources_lock_);
absl::optional<uint8_t> last_received_rtp_audio_level_
RTC_GUARDED_BY(&rtp_sources_lock_);
RTC_GUARDED_BY(&sync_info_lock_);
std::unique_ptr<AudioCodingModule> audio_coding_;
AudioSinkInterface* audio_sink_ = nullptr;
@ -565,24 +559,6 @@ absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()
return audio_coding_->ReceiveCodec();
}
std::vector<webrtc::RtpSource> ChannelReceive::GetSources() const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
int64_t now_ms = rtc::TimeMillis();
std::vector<RtpSource> 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<int, SdpAudioFormat>& codecs) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
@ -596,22 +572,11 @@ 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(&rtp_sources_lock_);
rtc::CritScope cs(&sync_info_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<uint32_t> csrcs = packet.Csrcs();
contributing_sources_.Update(
now_ms, csrcs,
has_audio_level ? absl::optional<uint8_t>(audio_level) : absl::nullopt,
packet.Timestamp());
}
// Store playout timestamp for the received RTP packet
@ -887,7 +852,7 @@ absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
return absl::nullopt;
}
{
rtc::CritScope cs(&rtp_sources_lock_);
rtc::CritScope cs(&sync_info_lock_);
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
return absl::nullopt;
}

2
audio/channel_receive.h
View File

@ -135,8 +135,6 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
// Used for obtaining RTT for a receive-only channel.
virtual void SetAssociatedSendChannel(
const ChannelSendInterface* channel) = 0;
virtual std::vector<RtpSource> GetSources() const = 0;
};
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(