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

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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}
This commit is contained in:
Artem Titov
2019-07-04 07:14:11 +00:00
committed by Commit Bot
parent c6c730b7fd
commit 67008dfb36
4 changed files with 47 additions and 19 deletions
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13
audio/audio_receive_stream.cc
View File

@ -113,9 +113,7 @@ 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)),
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<RtpSource> 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 {

2
audio/audio_receive_stream.h
View File

@ -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<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

@ -30,6 +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<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 {
@ -210,13 +213,16 @@ class ChannelReceive : public ChannelReceiveInterface,
std::unique_ptr<RtpRtcp> _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<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(&sync_info_lock_);
RTC_GUARDED_BY(&rtp_sources_lock_);
absl::optional<int64_t> last_received_rtp_system_time_ms_
RTC_GUARDED_BY(&sync_info_lock_);
RTC_GUARDED_BY(&rtp_sources_lock_);
absl::optional<uint8_t> last_received_rtp_audio_level_
RTC_GUARDED_BY(&rtp_sources_lock_);
std::unique_ptr<AudioCodingModule> audio_coding_;
AudioSinkInterface* audio_sink_ = nullptr;
@ -549,6 +555,24 @@ absl::optional<std::pair<int, SdpAudioFormat>>
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());
@ -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<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
@ -840,7 +875,7 @@ absl::optional<Syncable::Info> 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;
}

2
audio/channel_receive.h
View File

@ -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<RtpSource> GetSources() const = 0;
};
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(