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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 |
| 11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
| 12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
| 13 | 13 |
| 14 #include <memory> | 14 #include <memory> |
| 15 #include <vector> | |
| 16 | 15 |
| 17 #include "webrtc/api/audio/audio_mixer.h" | 16 #include "webrtc/api/audio/audio_mixer.h" |
| 18 #include "webrtc/audio/audio_state.h" | 17 #include "webrtc/audio/audio_state.h" |
| 19 #include "webrtc/base/constructormagic.h" | 18 #include "webrtc/base/constructormagic.h" |
| 20 #include "webrtc/base/thread_checker.h" | 19 #include "webrtc/base/thread_checker.h" |
| 21 #include "webrtc/call/audio_receive_stream.h" | 20 #include "webrtc/call/audio_receive_stream.h" |
| 22 #include "webrtc/call/syncable.h" | 21 #include "webrtc/call/syncable.h" |
| 23 | 22 |
| 24 namespace webrtc { | 23 namespace webrtc { |
| 25 class PacketRouter; | 24 class PacketRouter; |
| (...skipping 17 matching lines...) Expand all Loading... |
| 43 webrtc::RtcEventLog* event_log); | 42 webrtc::RtcEventLog* event_log); |
| 44 ~AudioReceiveStream() override; | 43 ~AudioReceiveStream() override; |
| 45 | 44 |
| 46 // webrtc::AudioReceiveStream implementation. | 45 // webrtc::AudioReceiveStream implementation. |
| 47 void Start() override; | 46 void Start() override; |
| 48 void Stop() override; | 47 void Stop() override; |
| 49 webrtc::AudioReceiveStream::Stats GetStats() const override; | 48 webrtc::AudioReceiveStream::Stats GetStats() const override; |
| 50 int GetOutputLevel() const override; | 49 int GetOutputLevel() const override; |
| 51 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; | 50 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
| 52 void SetGain(float gain) override; | 51 void SetGain(float gain) override; |
| 53 std::vector<webrtc::RtpSource> GetSources() const override; | |
| 54 | 52 |
| 55 // TODO(nisse): Intended to be part of an RtpPacketReceiver interface. | 53 // TODO(nisse): Intended to be part of an RtpPacketReceiver interface. |
| 56 void OnRtpPacket(const RtpPacketReceived& packet); | 54 void OnRtpPacket(const RtpPacketReceived& packet); |
| 57 | 55 |
| 58 // AudioMixer::Source | 56 // AudioMixer::Source |
| 59 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, | 57 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
| 60 AudioFrame* audio_frame) override; | 58 AudioFrame* audio_frame) override; |
| 61 int Ssrc() const override; | 59 int Ssrc() const override; |
| 62 int PreferredSampleRate() const override; | 60 int PreferredSampleRate() const override; |
| 63 | 61 |
| (...skipping 20 matching lines...) Expand all Loading... |
| 84 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 82 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
| 85 | 83 |
| 86 bool playing_ ACCESS_ON(worker_thread_checker_) = false; | 84 bool playing_ ACCESS_ON(worker_thread_checker_) = false; |
| 87 | 85 |
| 88 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 86 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
| 89 }; | 87 }; |
| 90 } // namespace internal | 88 } // namespace internal |
| 91 } // namespace webrtc | 89 } // namespace webrtc |
| 92 | 90 |
| 93 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 91 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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