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Side by Side Diff: webrtc/audio_receive_stream.h

Issue 1949533002: WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Retained Channel API by adding overloads; also add intended AudioReceiveStream API Created 4 years, 7 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
11 #ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_ 11 #ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_ 12 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
13 13
14 #include <map> 14 #include <map>
15 #include <memory> 15 #include <memory>
16 #include <string> 16 #include <string>
17 #include <vector> 17 #include <vector>
18 18
19 #include "webrtc/common_types.h" 19 #include "webrtc/common_types.h"
20 #include "webrtc/config.h" 20 #include "webrtc/config.h"
21 #include "webrtc/transport.h" 21 #include "webrtc/transport.h"
22 #include "webrtc/typedefs.h" 22 #include "webrtc/typedefs.h"
23 23
24 namespace webrtc { 24 namespace webrtc {
25 25
26 class AudioDecoder; 26 class AudioDecoder;
27 class AudioDecoderFactory;
27 class AudioSinkInterface; 28 class AudioSinkInterface;
28 29
29 // WORK IN PROGRESS 30 // WORK IN PROGRESS
30 // This class is under development and is not yet intended for for use outside 31 // This class is under development and is not yet intended for for use outside
31 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. 32 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
32 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 33 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
33 34
34 class AudioReceiveStream { 35 class AudioReceiveStream {
35 public: 36 public:
36 struct Stats { 37 struct Stats {
(...skipping 57 matching lines...) Expand 10 before | Expand all | Expand 10 after
94 // Identifier for an A/V synchronization group. Empty string to disable. 95 // Identifier for an A/V synchronization group. Empty string to disable.
95 // 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
96 // stream to one audio stream. Tracked by issue webrtc:4762. 97 // stream to one audio stream. Tracked by issue webrtc:4762.
97 std::string sync_group; 98 std::string sync_group;
98 99
99 // Decoders for every payload that we can receive. Call owns the 100 // Decoders for every payload that we can receive. Call owns the
100 // AudioDecoder instances once the Config is submitted to 101 // AudioDecoder instances once the Config is submitted to
101 // Call::CreateReceiveStream(). 102 // Call::CreateReceiveStream().
102 // 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.
103 std::map<uint8_t, AudioDecoder*> decoder_map; 104 std::map<uint8_t, AudioDecoder*> decoder_map;
105
106 std::shared_ptr<AudioDecoderFactory> decoder_factory;
104 }; 107 };
105 108
106 // Starts stream activity. 109 // Starts stream activity.
107 // When a stream is active, it can receive, process and deliver packets. 110 // When a stream is active, it can receive, process and deliver packets.
108 virtual void Start() = 0; 111 virtual void Start() = 0;
109 // Stops stream activity. 112 // Stops stream activity.
110 // When a stream is stopped, it can't receive, process or deliver packets. 113 // When a stream is stopped, it can't receive, process or deliver packets.
111 virtual void Stop() = 0; 114 virtual void Stop() = 0;
112 115
113 virtual Stats GetStats() const = 0; 116 virtual Stats GetStats() const = 0;
114 117
115 // Sets an audio sink that receives unmixed audio from the receive stream. 118 // Sets an audio sink that receives unmixed audio from the receive stream.
116 // Ownership of the sink is passed to the stream and can be used by the 119 // Ownership of the sink is passed to the stream and can be used by the
117 // caller to do lifetime management (i.e. when the sink's dtor is called). 120 // caller to do lifetime management (i.e. when the sink's dtor is called).
118 // Only one sink can be set and passing a null sink clears an existing one. 121 // Only one sink can be set and passing a null sink clears an existing one.
119 // NOTE: Audio must still somehow be pulled through AudioTransport for audio 122 // NOTE: Audio must still somehow be pulled through AudioTransport for audio
120 // to stream through this sink. In practice, this happens if mixed audio 123 // to stream through this sink. In practice, this happens if mixed audio
121 // is being pulled+rendered and/or if audio is being pulled for the purposes 124 // is being pulled+rendered and/or if audio is being pulled for the purposes
122 // of feeding to the AEC. 125 // of feeding to the AEC.
123 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; 126 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
124 127
125 protected: 128 protected:
126 virtual ~AudioReceiveStream() {} 129 virtual ~AudioReceiveStream() {}
127 }; 130 };
128 } // namespace webrtc 131 } // namespace webrtc
129 132
130 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ 133 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_
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