Index: webrtc/audio/audio_send_stream.h |
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h |
index 436c49824cc2325921c560e59f2f56bc3d439f99..f50f7c4d020966ee31e9c11c8309bd65ee666609 100644 |
--- a/webrtc/audio/audio_send_stream.h |
+++ b/webrtc/audio/audio_send_stream.h |
@@ -12,12 +12,15 @@ |
#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
#include <memory> |
+#include <vector> |
#include "webrtc/base/constructormagic.h" |
#include "webrtc/base/thread_checker.h" |
#include "webrtc/call/audio_send_stream.h" |
#include "webrtc/call/audio_state.h" |
#include "webrtc/call/bitrate_allocator.h" |
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
+#include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h" |
namespace webrtc { |
class SendSideCongestionController; |
@@ -33,7 +36,8 @@ class ChannelProxy; |
namespace internal { |
class AudioSendStream final : public webrtc::AudioSendStream, |
- public webrtc::BitrateAllocatorObserver { |
+ public webrtc::BitrateAllocatorObserver, |
+ public webrtc::PacketFeedbackObserver { |
public: |
AudioSendStream(const webrtc::AudioSendStream::Config& config, |
const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
@@ -62,6 +66,11 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
int64_t rtt, |
int64_t probing_interval_ms) override; |
+ // From PacketFeedbackObserver. |
+ void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; |
+ void OnPacketFeedbackVector( |
+ const std::vector<PacketFeedback>& packet_feedback_vector) override; |
+ |
const webrtc::AudioSendStream::Config& config() const; |
void SetTransportOverhead(int transport_overhead_per_packet); |
@@ -70,7 +79,8 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
bool SetupSendCodec(); |
- rtc::ThreadChecker thread_checker_; |
+ rtc::ThreadChecker worker_thread_checker_; |
+ rtc::ThreadChecker pacer_thread_checker_; |
rtc::TaskQueue* worker_queue_; |
const webrtc::AudioSendStream::Config config_; |
rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
@@ -80,6 +90,10 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
SendSideCongestionController* const send_side_cc_; |
std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; |
+ rtc::CriticalSection packet_loss_tracker_cs_; |
+ TransportFeedbackPacketLossTracker packet_loss_tracker_ |
+ GUARDED_BY(&packet_loss_tracker_cs_); |
+ |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
}; |
} // namespace internal |