Chromium Code Reviews| 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..f7ab1af3f986aab88714db2f6e59612290fd41ef 100644 |
| --- a/webrtc/audio/audio_send_stream.h |
| +++ b/webrtc/audio/audio_send_stream.h |
| @@ -18,6 +18,8 @@ |
| #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 +35,8 @@ class ChannelProxy; |
| namespace internal { |
| class AudioSendStream final : public webrtc::AudioSendStream, |
| - public webrtc::BitrateAllocatorObserver { |
| + public webrtc::BitrateAllocatorObserver, |
| + public webrtc::TransportFeedbackAdapterObserver { |
| public: |
| AudioSendStream(const webrtc::AudioSendStream::Config& config, |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| @@ -62,6 +65,11 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
| int64_t rtt, |
| int64_t probing_interval_ms) override; |
| + // From TransportFeedbackAdapterObserver |
|
the sun
2017/03/22 12:06:42
super nit: comments end in .
elad.alon_webrtc.org
2017/03/22 14:34:47
Since both you an Minyue comment on this, I'll ali
|
| + void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; |
| + void OnNewTransportFeedbackVector( |
|
stefan-webrtc
2017/03/22 12:03:01
Let's drop "New" if you don't think it provides va
elad.alon_webrtc.org
2017/03/22 14:34:47
(I've renamed elsewhere as a follow-up to Fredrick
|
| + const std::vector<PacketFeedback>& packet_feedback_vector) override; |
| + |
| const webrtc::AudioSendStream::Config& config() const; |
| void SetTransportOverhead(int transport_overhead_per_packet); |
| @@ -70,6 +78,7 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
| bool SetupSendCodec(); |
| + const Clock* const clock_; |
|
the sun
2017/03/22 12:06:42
Please, no more Clock*. Use the functions in webrt
elad.alon_webrtc.org
2017/03/22 14:34:47
Done.
|
| rtc::ThreadChecker thread_checker_; |
| rtc::TaskQueue* worker_queue_; |
| const webrtc::AudioSendStream::Config config_; |
| @@ -80,6 +89,10 @@ 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 |