Index: webrtc/audio/audio_receive_stream.h |
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h |
index 7dfc5d6bb86ca723cba13a6c92bc7c6a34c3648e..6721c7ee6531e4af9d1299074983e65ce697ef6b 100644 |
--- a/webrtc/audio/audio_receive_stream.h |
+++ b/webrtc/audio/audio_receive_stream.h |
@@ -18,13 +18,13 @@ |
#include "webrtc/base/constructormagic.h" |
#include "webrtc/base/thread_checker.h" |
#include "webrtc/call/audio_receive_stream.h" |
-#include "webrtc/call/audio_state.h" |
+#include "webrtc/call/syncable.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
namespace webrtc { |
+class PacketRouter; |
class RemoteBitrateEstimator; |
class RtcEventLog; |
-class PacketRouter; |
namespace voe { |
class ChannelProxy; |
@@ -34,7 +34,8 @@ namespace internal { |
class AudioSendStream; |
class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
- public AudioMixer::Source { |
+ public AudioMixer::Source, |
+ public Syncable { |
public: |
AudioReceiveStream(PacketRouter* packet_router, |
RemoteBitrateEstimator* remote_bitrate_estimator, |
@@ -50,6 +51,18 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
void SetGain(float gain) override; |
+ // AudioMixer::Source |
+ AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
+ AudioFrame* audio_frame) override; |
+ int Ssrc() const override; |
+ int PreferredSampleRate() const override; |
+ |
+ // Syncable |
+ int id() const override; |
+ rtc::Optional<Syncable::Info> GetInfo() const override; |
+ uint32_t GetPlayoutTimestamp() const override; |
+ void SetMinimumPlayoutDelay(int delay_ms) override; |
+ |
void AssociateSendStream(AudioSendStream* send_stream); |
void SignalNetworkState(NetworkState state); |
bool DeliverRtcp(const uint8_t* packet, size_t length); |
@@ -58,25 +71,20 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
const PacketTime& packet_time); |
const webrtc::AudioReceiveStream::Config& config() const; |
- // AudioMixer::Source |
- AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
- AudioFrame* audio_frame) override; |
- int PreferredSampleRate() const override; |
- int Ssrc() const override; |
- |
private: |
VoiceEngine* voice_engine() const; |
AudioState* audio_state() const; |
int SetVoiceEnginePlayout(bool playout); |
- rtc::ThreadChecker thread_checker_; |
+ rtc::ThreadChecker worker_thread_checker_; |
+ rtc::ThreadChecker module_process_thread_checker_; |
RemoteBitrateEstimator* const remote_bitrate_estimator_; |
const webrtc::AudioReceiveStream::Config config_; |
rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
- bool playing_ ACCESS_ON(thread_checker_) = false; |
+ bool playing_ ACCESS_ON(worker_thread_checker_) = false; |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
}; |