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Side by Side Diff: webrtc/audio_receive_stream.h

Issue 1604563002: Add send-side BWE to WebRtcVoiceEngine under a finch experiment. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 4 years, 10 months ago
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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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95 // Identifier for an A/V synchronization group. Empty string to disable. 95 // Identifier for an A/V synchronization group. Empty string to disable.
96 // TODO(pbos): Synchronize streams in a sync group, not just one video 96 // TODO(pbos): Synchronize streams in a sync group, not just one video
97 // stream to one audio stream. Tracked by issue webrtc:4762. 97 // stream to one audio stream. Tracked by issue webrtc:4762.
98 std::string sync_group; 98 std::string sync_group;
99 99
100 // Decoders for every payload that we can receive. Call owns the 100 // Decoders for every payload that we can receive. Call owns the
101 // AudioDecoder instances once the Config is submitted to 101 // AudioDecoder instances once the Config is submitted to
102 // Call::CreateReceiveStream(). 102 // Call::CreateReceiveStream().
103 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. 103 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
104 std::map<uint8_t, AudioDecoder*> decoder_map; 104 std::map<uint8_t, AudioDecoder*> decoder_map;
105
106 // TODO(pbos): Remove config option once combined A/V BWE is always on.
107 bool combined_audio_video_bwe = false;
108 }; 105 };
109 106
110 virtual Stats GetStats() const = 0; 107 virtual Stats GetStats() const = 0;
111 108
112 // Sets an audio sink that receives unmixed audio from the receive stream. 109 // Sets an audio sink that receives unmixed audio from the receive stream.
113 // Ownership of the sink is passed to the stream and can be used by the 110 // Ownership of the sink is passed to the stream and can be used by the
114 // caller to do lifetime management (i.e. when the sink's dtor is called). 111 // caller to do lifetime management (i.e. when the sink's dtor is called).
115 // Only one sink can be set and passing a null sink clears an existing one. 112 // Only one sink can be set and passing a null sink clears an existing one.
116 // NOTE: Audio must still somehow be pulled through AudioTransport for audio 113 // NOTE: Audio must still somehow be pulled through AudioTransport for audio
117 // to stream through this sink. In practice, this happens if mixed audio 114 // to stream through this sink. In practice, this happens if mixed audio
118 // is being pulled+rendered and/or if audio is being pulled for the purposes 115 // is being pulled+rendered and/or if audio is being pulled for the purposes
119 // of feeding to the AEC. 116 // of feeding to the AEC.
120 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0; 117 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
121 }; 118 };
122 } // namespace webrtc 119 } // namespace webrtc
123 120
124 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ 121 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_
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