OLD | NEW |
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 |
(...skipping 130 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
141 const AudioOptions& options, | 141 const AudioOptions& options, |
142 webrtc::Call* call); | 142 webrtc::Call* call); |
143 ~WebRtcVoiceMediaChannel() override; | 143 ~WebRtcVoiceMediaChannel() override; |
144 | 144 |
145 const AudioOptions& options() const { return options_; } | 145 const AudioOptions& options() const { return options_; } |
146 | 146 |
147 rtc::DiffServCodePoint PreferredDscp() const override; | 147 rtc::DiffServCodePoint PreferredDscp() const override; |
148 | 148 |
149 bool SetSendParameters(const AudioSendParameters& params) override; | 149 bool SetSendParameters(const AudioSendParameters& params) override; |
150 bool SetRecvParameters(const AudioRecvParameters& params) override; | 150 bool SetRecvParameters(const AudioRecvParameters& params) override; |
151 webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const override; | 151 webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override; |
152 bool SetRtpParameters(uint32_t ssrc, | 152 bool SetRtpSendParameters(uint32_t ssrc, |
153 const webrtc::RtpParameters& parameters) override; | 153 const webrtc::RtpParameters& parameters) override; |
| 154 webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override; |
| 155 bool SetRtpReceiveParameters( |
| 156 uint32_t ssrc, |
| 157 const webrtc::RtpParameters& parameters) override; |
154 | 158 |
155 bool SetPlayout(bool playout) override; | 159 bool SetPlayout(bool playout) override; |
156 bool PausePlayout(); | 160 bool PausePlayout(); |
157 bool ResumePlayout(); | 161 bool ResumePlayout(); |
158 void SetSend(bool send) override; | 162 void SetSend(bool send) override; |
159 bool SetAudioSend(uint32_t ssrc, | 163 bool SetAudioSend(uint32_t ssrc, |
160 bool enable, | 164 bool enable, |
161 const AudioOptions* options, | 165 const AudioOptions* options, |
162 AudioSource* source) override; | 166 AudioSource* source) override; |
163 bool AddSendStream(const StreamParams& sp) override; | 167 bool AddSendStream(const StreamParams& sp) override; |
(...skipping 58 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
222 int GetLastEngineError() { return engine()->GetLastEngineError(); } | 226 int GetLastEngineError() { return engine()->GetLastEngineError(); } |
223 int GetOutputLevel(int channel); | 227 int GetOutputLevel(int channel); |
224 bool SetPlayout(int channel, bool playout); | 228 bool SetPlayout(int channel, bool playout); |
225 bool ChangePlayout(bool playout); | 229 bool ChangePlayout(bool playout); |
226 int CreateVoEChannel(); | 230 int CreateVoEChannel(); |
227 bool DeleteVoEChannel(int channel); | 231 bool DeleteVoEChannel(int channel); |
228 bool IsDefaultRecvStream(uint32_t ssrc) { | 232 bool IsDefaultRecvStream(uint32_t ssrc) { |
229 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); | 233 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); |
230 } | 234 } |
231 bool SetMaxSendBitrate(int bps); | 235 bool SetMaxSendBitrate(int bps); |
232 bool SetChannelParameters(int channel, | 236 bool SetChannelSendParameters(int channel, |
233 const webrtc::RtpParameters& parameters); | 237 const webrtc::RtpParameters& parameters); |
234 bool SetMaxSendBitrate(int channel, int bps); | 238 bool SetMaxSendBitrate(int channel, int bps); |
235 bool HasSendCodec() const { | 239 bool HasSendCodec() const { |
236 return send_codec_spec_.codec_inst.pltype != -1; | 240 return send_codec_spec_.codec_inst.pltype != -1; |
237 } | 241 } |
238 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); | 242 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); |
239 void SetupRecording(); | 243 void SetupRecording(); |
240 | 244 |
241 rtc::ThreadChecker worker_thread_checker_; | 245 rtc::ThreadChecker worker_thread_checker_; |
242 | 246 |
243 WebRtcVoiceEngine* const engine_ = nullptr; | 247 WebRtcVoiceEngine* const engine_ = nullptr; |
(...skipping 42 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
286 int cng_payload_type = -1; | 290 int cng_payload_type = -1; |
287 int cng_plfreq = -1; | 291 int cng_plfreq = -1; |
288 webrtc::CodecInst codec_inst; | 292 webrtc::CodecInst codec_inst; |
289 } send_codec_spec_; | 293 } send_codec_spec_; |
290 | 294 |
291 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 295 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
292 }; | 296 }; |
293 } // namespace cricket | 297 } // namespace cricket |
294 | 298 |
295 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 299 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
OLD | NEW |