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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 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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142 const AudioOptions& options, | 142 const AudioOptions& options, |
143 webrtc::Call* call); | 143 webrtc::Call* call); |
144 ~WebRtcVoiceMediaChannel() override; | 144 ~WebRtcVoiceMediaChannel() override; |
145 | 145 |
146 const AudioOptions& options() const { return options_; } | 146 const AudioOptions& options() const { return options_; } |
147 | 147 |
148 rtc::DiffServCodePoint PreferredDscp() const override; | 148 rtc::DiffServCodePoint PreferredDscp() const override; |
149 | 149 |
150 bool SetSendParameters(const AudioSendParameters& params) override; | 150 bool SetSendParameters(const AudioSendParameters& params) override; |
151 bool SetRecvParameters(const AudioRecvParameters& params) override; | 151 bool SetRecvParameters(const AudioRecvParameters& params) override; |
152 webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const override; | 152 webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override; |
153 bool SetRtpParameters(uint32_t ssrc, | 153 bool SetRtpSendParameters(uint32_t ssrc, |
154 const webrtc::RtpParameters& parameters) override; | 154 const webrtc::RtpParameters& parameters) override; |
| 155 webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override; |
| 156 bool SetRtpReceiveParameters( |
| 157 uint32_t ssrc, |
| 158 const webrtc::RtpParameters& parameters) override; |
155 | 159 |
156 bool SetPlayout(bool playout) override; | 160 bool SetPlayout(bool playout) override; |
157 bool PausePlayout(); | 161 bool PausePlayout(); |
158 bool ResumePlayout(); | 162 bool ResumePlayout(); |
159 void SetSend(bool send) override; | 163 void SetSend(bool send) override; |
160 bool SetAudioSend(uint32_t ssrc, | 164 bool SetAudioSend(uint32_t ssrc, |
161 bool enable, | 165 bool enable, |
162 const AudioOptions* options, | 166 const AudioOptions* options, |
163 AudioSource* source) override; | 167 AudioSource* source) override; |
164 bool AddSendStream(const StreamParams& sp) override; | 168 bool AddSendStream(const StreamParams& sp) override; |
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223 int GetLastEngineError() { return engine()->GetLastEngineError(); } | 227 int GetLastEngineError() { return engine()->GetLastEngineError(); } |
224 int GetOutputLevel(int channel); | 228 int GetOutputLevel(int channel); |
225 bool SetPlayout(int channel, bool playout); | 229 bool SetPlayout(int channel, bool playout); |
226 bool ChangePlayout(bool playout); | 230 bool ChangePlayout(bool playout); |
227 int CreateVoEChannel(); | 231 int CreateVoEChannel(); |
228 bool DeleteVoEChannel(int channel); | 232 bool DeleteVoEChannel(int channel); |
229 bool IsDefaultRecvStream(uint32_t ssrc) { | 233 bool IsDefaultRecvStream(uint32_t ssrc) { |
230 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); | 234 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); |
231 } | 235 } |
232 bool SetMaxSendBitrate(int bps); | 236 bool SetMaxSendBitrate(int bps); |
233 bool SetChannelParameters(int channel, | 237 bool SetChannelSendParameters(int channel, |
234 const webrtc::RtpParameters& parameters); | 238 const webrtc::RtpParameters& parameters); |
235 bool SetMaxSendBitrate(int channel, int bps); | 239 bool SetMaxSendBitrate(int channel, int bps); |
236 bool HasSendCodec() const { | 240 bool HasSendCodec() const { |
237 return send_codec_spec_.codec_inst.pltype != -1; | 241 return send_codec_spec_.codec_inst.pltype != -1; |
238 } | 242 } |
239 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); | 243 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); |
240 void SetupRecording(); | 244 void SetupRecording(); |
241 | 245 |
242 rtc::ThreadChecker worker_thread_checker_; | 246 rtc::ThreadChecker worker_thread_checker_; |
243 | 247 |
244 WebRtcVoiceEngine* const engine_ = nullptr; | 248 WebRtcVoiceEngine* const engine_ = nullptr; |
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287 int cng_payload_type = -1; | 291 int cng_payload_type = -1; |
288 int cng_plfreq = -1; | 292 int cng_plfreq = -1; |
289 webrtc::CodecInst codec_inst; | 293 webrtc::CodecInst codec_inst; |
290 } send_codec_spec_; | 294 } send_codec_spec_; |
291 | 295 |
292 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 296 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
293 }; | 297 }; |
294 } // namespace cricket | 298 } // namespace cricket |
295 | 299 |
296 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 300 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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