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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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49 int duration_ms) override; | 49 int duration_ms) override; |
50 void SetMuted(bool muted) override; | 50 void SetMuted(bool muted) override; |
51 webrtc::AudioSendStream::Stats GetStats() const override; | 51 webrtc::AudioSendStream::Stats GetStats() const override; |
52 | 52 |
53 void SignalNetworkState(NetworkState state); | 53 void SignalNetworkState(NetworkState state); |
54 bool DeliverRtcp(const uint8_t* packet, size_t length); | 54 bool DeliverRtcp(const uint8_t* packet, size_t length); |
55 | 55 |
56 // Implements BitrateAllocatorObserver. | 56 // Implements BitrateAllocatorObserver. |
57 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, | 57 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, |
58 uint8_t fraction_loss, | 58 uint8_t fraction_loss, |
59 int64_t rtt) override; | 59 int64_t rtt, |
| 60 int64_t probing_interval_ms) override; |
60 | 61 |
61 const webrtc::AudioSendStream::Config& config() const; | 62 const webrtc::AudioSendStream::Config& config() const; |
62 void SetTransportOverhead(int transport_overhead_per_packet); | 63 void SetTransportOverhead(int transport_overhead_per_packet); |
63 | 64 |
64 private: | 65 private: |
65 VoiceEngine* voice_engine() const; | 66 VoiceEngine* voice_engine() const; |
66 | 67 |
67 bool SetupSendCodec(); | 68 bool SetupSendCodec(); |
68 | 69 |
69 rtc::ThreadChecker thread_checker_; | 70 rtc::ThreadChecker thread_checker_; |
70 rtc::TaskQueue* worker_queue_; | 71 rtc::TaskQueue* worker_queue_; |
71 const webrtc::AudioSendStream::Config config_; | 72 const webrtc::AudioSendStream::Config config_; |
72 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 73 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
73 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 74 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
74 | 75 |
75 BitrateAllocator* const bitrate_allocator_; | 76 BitrateAllocator* const bitrate_allocator_; |
76 | 77 |
77 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 78 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
78 }; | 79 }; |
79 } // namespace internal | 80 } // namespace internal |
80 } // namespace webrtc | 81 } // namespace webrtc |
81 | 82 |
82 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 83 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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