bef7b058f5
Setting a minimum delay can fail in some cases. It is important that the AV sync code is aware of failures and can act accordingly to recover and prevent sync delays that keep increasing indefinitely. Bug: webrtc:11805 Change-Id: I0deed951dc6c6d0905536a949af875e0a6d9f7fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183360 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32062}
122 lines
4.5 KiB
C++
122 lines
4.5 KiB
C++
/*
|
|
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef AUDIO_AUDIO_RECEIVE_STREAM_H_
|
|
#define AUDIO_AUDIO_RECEIVE_STREAM_H_
|
|
|
|
#include <memory>
|
|
#include <vector>
|
|
|
|
#include "api/audio/audio_mixer.h"
|
|
#include "api/neteq/neteq_factory.h"
|
|
#include "api/rtp_headers.h"
|
|
#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"
|
|
|
|
namespace webrtc {
|
|
class PacketRouter;
|
|
class ProcessThread;
|
|
class RtcEventLog;
|
|
class RtpPacketReceived;
|
|
class RtpStreamReceiverControllerInterface;
|
|
class RtpStreamReceiverInterface;
|
|
|
|
namespace voe {
|
|
class ChannelReceiveInterface;
|
|
} // namespace voe
|
|
|
|
namespace internal {
|
|
class AudioSendStream;
|
|
|
|
class AudioReceiveStream final : public webrtc::AudioReceiveStream,
|
|
public AudioMixer::Source,
|
|
public Syncable {
|
|
public:
|
|
AudioReceiveStream(Clock* clock,
|
|
RtpStreamReceiverControllerInterface* receiver_controller,
|
|
PacketRouter* packet_router,
|
|
ProcessThread* module_process_thread,
|
|
NetEqFactory* neteq_factory,
|
|
const webrtc::AudioReceiveStream::Config& config,
|
|
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
|
|
webrtc::RtcEventLog* event_log);
|
|
// For unit tests, which need to supply a mock channel receive.
|
|
AudioReceiveStream(
|
|
Clock* clock,
|
|
RtpStreamReceiverControllerInterface* receiver_controller,
|
|
PacketRouter* packet_router,
|
|
const webrtc::AudioReceiveStream::Config& config,
|
|
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
|
|
webrtc::RtcEventLog* event_log,
|
|
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
|
|
~AudioReceiveStream() override;
|
|
|
|
// webrtc::AudioReceiveStream implementation.
|
|
void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override;
|
|
void Start() override;
|
|
void Stop() override;
|
|
webrtc::AudioReceiveStream::Stats GetStats() const override;
|
|
void SetSink(AudioSinkInterface* sink) override;
|
|
void SetGain(float gain) override;
|
|
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
|
|
int GetBaseMinimumPlayoutDelayMs() const override;
|
|
std::vector<webrtc::RtpSource> GetSources() const override;
|
|
|
|
// AudioMixer::Source
|
|
AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
|
|
AudioFrame* audio_frame) override;
|
|
int Ssrc() const override;
|
|
int PreferredSampleRate() const override;
|
|
|
|
// Syncable
|
|
uint32_t id() const override;
|
|
absl::optional<Syncable::Info> GetInfo() const override;
|
|
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
|
|
int64_t* time_ms) const override;
|
|
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
|
|
int64_t time_ms) override;
|
|
bool SetMinimumPlayoutDelay(int delay_ms) override;
|
|
|
|
void AssociateSendStream(AudioSendStream* send_stream);
|
|
void DeliverRtcp(const uint8_t* packet, size_t length);
|
|
const webrtc::AudioReceiveStream::Config& config() const;
|
|
const AudioSendStream* GetAssociatedSendStreamForTesting() const;
|
|
|
|
private:
|
|
static void ConfigureStream(AudioReceiveStream* stream,
|
|
const Config& new_config,
|
|
bool first_time);
|
|
|
|
AudioState* audio_state() const;
|
|
|
|
rtc::ThreadChecker worker_thread_checker_;
|
|
rtc::ThreadChecker module_process_thread_checker_;
|
|
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;
|
|
|
|
std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
|
|
|
|
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
|
|
};
|
|
} // namespace internal
|
|
} // namespace webrtc
|
|
|
|
#endif // AUDIO_AUDIO_RECEIVE_STREAM_H_
|