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| 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 |
| (...skipping 15 matching lines...) Expand all Loading... |
| 26 namespace voe { | 26 namespace voe { |
| 27 class ChannelProxy; | 27 class ChannelProxy; |
| 28 } // namespace voe | 28 } // namespace voe |
| 29 | 29 |
| 30 namespace internal { | 30 namespace internal { |
| 31 class AudioSendStream final : public webrtc::AudioSendStream, | 31 class AudioSendStream final : public webrtc::AudioSendStream, |
| 32 public webrtc::BitrateAllocatorObserver { | 32 public webrtc::BitrateAllocatorObserver { |
| 33 public: | 33 public: |
| 34 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 34 AudioSendStream(const webrtc::AudioSendStream::Config& config, |
| 35 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 35 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| 36 rtc::TaskQueue* worker_queue, | |
| 37 CongestionController* congestion_controller, | 36 CongestionController* congestion_controller, |
| 38 BitrateAllocator* bitrate_allocator); | 37 BitrateAllocator* bitrate_allocator); |
| 39 ~AudioSendStream() override; | 38 ~AudioSendStream() override; |
| 40 | 39 |
| 41 // webrtc::AudioSendStream implementation. | 40 // webrtc::AudioSendStream implementation. |
| 42 void Start() override; | 41 void Start() override; |
| 43 void Stop() override; | 42 void Stop() override; |
| 44 bool SendTelephoneEvent(int payload_type, int event, | 43 bool SendTelephoneEvent(int payload_type, int event, |
| 45 int duration_ms) override; | 44 int duration_ms) override; |
| 46 void SetMuted(bool muted) override; | 45 void SetMuted(bool muted) override; |
| 47 webrtc::AudioSendStream::Stats GetStats() const override; | 46 webrtc::AudioSendStream::Stats GetStats() const override; |
| 48 | 47 |
| 49 void SignalNetworkState(NetworkState state); | 48 void SignalNetworkState(NetworkState state); |
| 50 bool DeliverRtcp(const uint8_t* packet, size_t length); | 49 bool DeliverRtcp(const uint8_t* packet, size_t length); |
| 51 | 50 |
| 52 // Implements BitrateAllocatorObserver. | 51 // Implements BitrateAllocatorObserver. |
| 53 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, | 52 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, |
| 54 uint8_t fraction_loss, | 53 uint8_t fraction_loss, |
| 55 int64_t rtt) override; | 54 int64_t rtt) override; |
| 56 | 55 |
| 57 const webrtc::AudioSendStream::Config& config() const; | 56 const webrtc::AudioSendStream::Config& config() const; |
| 58 | 57 |
| 59 private: | 58 private: |
| 60 VoiceEngine* voice_engine() const; | 59 VoiceEngine* voice_engine() const; |
| 61 | 60 |
| 62 rtc::ThreadChecker thread_checker_; | 61 rtc::ThreadChecker thread_checker_; |
| 63 rtc::TaskQueue* worker_queue_; | |
| 64 const webrtc::AudioSendStream::Config config_; | 62 const webrtc::AudioSendStream::Config config_; |
| 65 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 63 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| 66 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 64 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
| 67 | 65 |
| 68 BitrateAllocator* const bitrate_allocator_; | 66 BitrateAllocator* const bitrate_allocator_; |
| 69 | 67 |
| 70 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 68 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
| 71 }; | 69 }; |
| 72 } // namespace internal | 70 } // namespace internal |
| 73 } // namespace webrtc | 71 } // namespace webrtc |
| 74 | 72 |
| 75 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 73 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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