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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/config.h" | 20 #include "webrtc/config.h" |
20 #include "webrtc/stream.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 AudioSinkInterface; | 27 class AudioSinkInterface; |
28 | 28 |
29 // WORK IN PROGRESS | 29 // WORK IN PROGRESS |
30 // This class is under development and is not yet intended for for use outside | 30 // This class is under development and is not yet intended for for use outside |
31 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | 31 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
32 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 | 32 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
33 | 33 |
34 class AudioReceiveStream : public ReceiveStream { | 34 class AudioReceiveStream { |
35 public: | 35 public: |
36 struct Stats { | 36 struct Stats { |
37 uint32_t remote_ssrc = 0; | 37 uint32_t remote_ssrc = 0; |
38 int64_t bytes_rcvd = 0; | 38 int64_t bytes_rcvd = 0; |
39 uint32_t packets_rcvd = 0; | 39 uint32_t packets_rcvd = 0; |
40 uint32_t packets_lost = 0; | 40 uint32_t packets_lost = 0; |
41 float fraction_lost = 0.0f; | 41 float fraction_lost = 0.0f; |
42 std::string codec_name; | 42 std::string codec_name; |
43 uint32_t ext_seqnum = 0; | 43 uint32_t ext_seqnum = 0; |
44 uint32_t jitter_ms = 0; | 44 uint32_t jitter_ms = 0; |
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96 // stream to one audio stream. Tracked by issue webrtc:4762. | 96 // stream to one audio stream. Tracked by issue webrtc:4762. |
97 std::string sync_group; | 97 std::string sync_group; |
98 | 98 |
99 // Decoders for every payload that we can receive. Call owns the | 99 // Decoders for every payload that we can receive. Call owns the |
100 // AudioDecoder instances once the Config is submitted to | 100 // AudioDecoder instances once the Config is submitted to |
101 // Call::CreateReceiveStream(). | 101 // Call::CreateReceiveStream(). |
102 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. | 102 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. |
103 std::map<uint8_t, AudioDecoder*> decoder_map; | 103 std::map<uint8_t, AudioDecoder*> decoder_map; |
104 }; | 104 }; |
105 | 105 |
| 106 // Starts stream activity. |
| 107 // When a stream is active, it can receive, process and deliver packets. |
| 108 virtual void Start() = 0; |
| 109 // Stops stream activity. |
| 110 // When a stream is stopped, it can't receive, process or deliver packets. |
| 111 virtual void Stop() = 0; |
| 112 |
106 virtual Stats GetStats() const = 0; | 113 virtual Stats GetStats() const = 0; |
107 | 114 |
108 // Sets an audio sink that receives unmixed audio from the receive stream. | 115 // Sets an audio sink that receives unmixed audio from the receive stream. |
109 // Ownership of the sink is passed to the stream and can be used by the | 116 // Ownership of the sink is passed to the stream and can be used by the |
110 // caller to do lifetime management (i.e. when the sink's dtor is called). | 117 // caller to do lifetime management (i.e. when the sink's dtor is called). |
111 // Only one sink can be set and passing a null sink clears an existing one. | 118 // Only one sink can be set and passing a null sink clears an existing one. |
112 // NOTE: Audio must still somehow be pulled through AudioTransport for audio | 119 // NOTE: Audio must still somehow be pulled through AudioTransport for audio |
113 // to stream through this sink. In practice, this happens if mixed audio | 120 // to stream through this sink. In practice, this happens if mixed audio |
114 // is being pulled+rendered and/or if audio is being pulled for the purposes | 121 // is being pulled+rendered and/or if audio is being pulled for the purposes |
115 // of feeding to the AEC. | 122 // of feeding to the AEC. |
116 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; | 123 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; |
| 124 |
| 125 protected: |
| 126 virtual ~AudioReceiveStream() {} |
117 }; | 127 }; |
118 } // namespace webrtc | 128 } // namespace webrtc |
119 | 129 |
120 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ | 130 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ |
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