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