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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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