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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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116 SdpAudioFormat format; | 116 SdpAudioFormat format; |
117 bool nack_enabled = false; | 117 bool nack_enabled = false; |
118 bool transport_cc_enabled = false; | 118 bool transport_cc_enabled = false; |
119 rtc::Optional<int> cng_payload_type; | 119 rtc::Optional<int> cng_payload_type; |
120 // If unset, use the encoder's default target bitrate. | 120 // If unset, use the encoder's default target bitrate. |
121 rtc::Optional<int> target_bitrate_bps; | 121 rtc::Optional<int> target_bitrate_bps; |
122 }; | 122 }; |
123 | 123 |
124 rtc::Optional<SendCodecSpec> send_codec_spec; | 124 rtc::Optional<SendCodecSpec> send_codec_spec; |
125 rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; | 125 rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; |
126 | |
127 // Track ID as specified during track creation. | |
128 std::string track_id; | |
nisse-webrtc
2017/09/18 08:44:28
Is the id of a track constant, or can it be change
alexnarest
2017/09/29 12:13:57
I do not think track_id can be changed directly bu
| |
126 }; | 129 }; |
127 | 130 |
128 // Reconfigure the stream according to the Configuration. | 131 // Reconfigure the stream according to the Configuration. |
129 virtual void Reconfigure(const Config& config) = 0; | 132 virtual void Reconfigure(const Config& config) = 0; |
130 | 133 |
131 // Starts stream activity. | 134 // Starts stream activity. |
132 // When a stream is active, it can receive, process and deliver packets. | 135 // When a stream is active, it can receive, process and deliver packets. |
133 virtual void Start() = 0; | 136 virtual void Start() = 0; |
134 // Stops stream activity. | 137 // Stops stream activity. |
135 // When a stream is stopped, it can't receive, process or deliver packets. | 138 // When a stream is stopped, it can't receive, process or deliver packets. |
136 virtual void Stop() = 0; | 139 virtual void Stop() = 0; |
137 | 140 |
138 // TODO(solenberg): Make payload_type a config property instead. | 141 // TODO(solenberg): Make payload_type a config property instead. |
139 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, | 142 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, |
140 int event, int duration_ms) = 0; | 143 int event, int duration_ms) = 0; |
141 | 144 |
142 virtual void SetMuted(bool muted) = 0; | 145 virtual void SetMuted(bool muted) = 0; |
143 | 146 |
144 virtual Stats GetStats() const = 0; | 147 virtual Stats GetStats() const = 0; |
145 | 148 |
146 protected: | 149 protected: |
147 virtual ~AudioSendStream() {} | 150 virtual ~AudioSendStream() {} |
148 }; | 151 }; |
149 } // namespace webrtc | 152 } // namespace webrtc |
150 | 153 |
151 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 154 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
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