 Chromium Code Reviews
 Chromium Code Reviews Issue 2638083002:
  Attach TransportFeedbackPacketLossTracker to ANA (PLR only)  (Closed)
    
  
    Issue 2638083002:
  Attach TransportFeedbackPacketLossTracker to ANA (PLR only)  (Closed) 
  | OLD | NEW | 
|---|---|
| 1 /* | 1 /* | 
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 
| 3 * | 3 * | 
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license | 
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source | 
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found | 
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may | 
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. | 
| 9 */ | 9 */ | 
| 10 | 10 | 
| 11 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 11 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 
| 12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 
| 13 | 13 | 
| 14 #include <memory> | 14 #include <memory> | 
| 15 | 15 | 
| 16 #include "webrtc/base/constructormagic.h" | 16 #include "webrtc/base/constructormagic.h" | 
| 17 #include "webrtc/base/thread_checker.h" | 17 #include "webrtc/base/thread_checker.h" | 
| 18 #include "webrtc/call/audio_send_stream.h" | 18 #include "webrtc/call/audio_send_stream.h" | 
| 19 #include "webrtc/call/audio_state.h" | 19 #include "webrtc/call/audio_state.h" | 
| 20 #include "webrtc/call/bitrate_allocator.h" | 20 #include "webrtc/call/bitrate_allocator.h" | 
| 21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | |
| 22 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h" | |
| 21 | 23 | 
| 22 namespace webrtc { | 24 namespace webrtc { | 
| 23 class SendSideCongestionController; | 25 class SendSideCongestionController; | 
| 
minyue-webrtc
2017/03/22 07:51:39
just a note: please try rebasing in a separate pat
 
elad.alon_webrtc.org
2017/03/22 09:36:30
Sure, will do.
 | |
| 24 class VoiceEngine; | 26 class VoiceEngine; | 
| 25 class RtcEventLog; | 27 class RtcEventLog; | 
| 26 class RtcpBandwidthObserver; | 28 class RtcpBandwidthObserver; | 
| 27 class RtcpRttStats; | 29 class RtcpRttStats; | 
| 28 class PacketRouter; | 30 class PacketRouter; | 
| 29 | 31 | 
| 30 namespace voe { | 32 namespace voe { | 
| 31 class ChannelProxy; | 33 class ChannelProxy; | 
| 32 } // namespace voe | 34 } // namespace voe | 
| 33 | 35 | 
| 34 namespace internal { | 36 namespace internal { | 
| 35 class AudioSendStream final : public webrtc::AudioSendStream, | 37 class AudioSendStream final : public webrtc::AudioSendStream, | 
| 36 public webrtc::BitrateAllocatorObserver { | 38 public webrtc::BitrateAllocatorObserver, | 
| 39 public webrtc::TransportFeedbackAdapterObserver { | |
| 37 public: | 40 public: | 
| 38 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 41 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 
| 39 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 42 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 
| 40 rtc::TaskQueue* worker_queue, | 43 rtc::TaskQueue* worker_queue, | 
| 41 PacketRouter* packet_router, | 44 PacketRouter* packet_router, | 
| 42 SendSideCongestionController* send_side_cc, | 45 SendSideCongestionController* send_side_cc, | 
| 43 BitrateAllocator* bitrate_allocator, | 46 BitrateAllocator* bitrate_allocator, | 
| 44 RtcEventLog* event_log, | 47 RtcEventLog* event_log, | 
| 45 RtcpRttStats* rtcp_rtt_stats); | 48 RtcpRttStats* rtcp_rtt_stats); | 
| 46 ~AudioSendStream() override; | 49 ~AudioSendStream() override; | 
| 47 | 50 | 
| 48 // webrtc::AudioSendStream implementation. | 51 // webrtc::AudioSendStream implementation. | 
| 49 void Start() override; | 52 void Start() override; | 
| 50 void Stop() override; | 53 void Stop() override; | 
| 51 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, | 54 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, | 
| 52 int duration_ms) override; | 55 int duration_ms) override; | 
| 53 void SetMuted(bool muted) override; | 56 void SetMuted(bool muted) override; | 
| 54 webrtc::AudioSendStream::Stats GetStats() const override; | 57 webrtc::AudioSendStream::Stats GetStats() const override; | 
| 55 | 58 | 
| 56 void SignalNetworkState(NetworkState state); | 59 void SignalNetworkState(NetworkState state); | 
| 57 bool DeliverRtcp(const uint8_t* packet, size_t length); | 60 bool DeliverRtcp(const uint8_t* packet, size_t length); | 
| 58 | 61 | 
| 59 // Implements BitrateAllocatorObserver. | 62 // Implements BitrateAllocatorObserver. | 
| 60 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, | 63 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, | 
| 61 uint8_t fraction_loss, | 64 uint8_t fraction_loss, | 
| 62 int64_t rtt, | 65 int64_t rtt, | 
| 63 int64_t probing_interval_ms) override; | 66 int64_t probing_interval_ms) override; | 
| 64 | 67 | 
| 68 // From TransportFeedbackAdapterObserver | |
| 69 void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; | |
| 70 void OnNewTransportFeedbackVector( | |
| 71 const std::vector<PacketFeedback>& packet_feedback_vector) override; | |
| 72 | |
| 65 const webrtc::AudioSendStream::Config& config() const; | 73 const webrtc::AudioSendStream::Config& config() const; | 
| 66 void SetTransportOverhead(int transport_overhead_per_packet); | 74 void SetTransportOverhead(int transport_overhead_per_packet); | 
| 67 | 75 | 
| 68 private: | 76 private: | 
| 69 VoiceEngine* voice_engine() const; | 77 VoiceEngine* voice_engine() const; | 
| 70 | 78 | 
| 71 bool SetupSendCodec(); | 79 bool SetupSendCodec(); | 
| 72 | 80 | 
| 81 const Clock* const clock_; | |
| 73 rtc::ThreadChecker thread_checker_; | 82 rtc::ThreadChecker thread_checker_; | 
| 74 rtc::TaskQueue* worker_queue_; | 83 rtc::TaskQueue* worker_queue_; | 
| 75 const webrtc::AudioSendStream::Config config_; | 84 const webrtc::AudioSendStream::Config config_; | 
| 76 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 85 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 
| 77 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 86 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 
| 78 | 87 | 
| 79 BitrateAllocator* const bitrate_allocator_; | 88 BitrateAllocator* const bitrate_allocator_; | 
| 80 SendSideCongestionController* const send_side_cc_; | 89 SendSideCongestionController* const send_side_cc_; | 
| 81 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; | 90 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; | 
| 82 | 91 | 
| 92 rtc::CriticalSection packet_loss_tracker_cs_; | |
| 93 TransportFeedbackPacketLossTracker packet_loss_tracker_ | |
| 94 GUARDED_BY(&packet_loss_tracker_cs_); | |
| 95 | |
| 83 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 96 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 
| 84 }; | 97 }; | 
| 85 } // namespace internal | 98 } // namespace internal | 
| 86 } // namespace webrtc | 99 } // namespace webrtc | 
| 87 | 100 | 
| 88 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 101 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 
| OLD | NEW |