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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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170 bool SetSendParameters(const AudioSendParameters& params) override; | 170 bool SetSendParameters(const AudioSendParameters& params) override; |
171 bool SetRecvParameters(const AudioRecvParameters& params) override; | 171 bool SetRecvParameters(const AudioRecvParameters& params) override; |
172 webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override; | 172 webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override; |
173 bool SetRtpSendParameters(uint32_t ssrc, | 173 bool SetRtpSendParameters(uint32_t ssrc, |
174 const webrtc::RtpParameters& parameters) override; | 174 const webrtc::RtpParameters& parameters) override; |
175 webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override; | 175 webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override; |
176 bool SetRtpReceiveParameters( | 176 bool SetRtpReceiveParameters( |
177 uint32_t ssrc, | 177 uint32_t ssrc, |
178 const webrtc::RtpParameters& parameters) override; | 178 const webrtc::RtpParameters& parameters) override; |
179 | 179 |
180 bool SetPlayout(bool playout) override; | 180 void SetPlayout(bool playout) override; |
181 bool PausePlayout(); | |
182 bool ResumePlayout(); | |
183 void SetSend(bool send) override; | 181 void SetSend(bool send) override; |
184 bool SetAudioSend(uint32_t ssrc, | 182 bool SetAudioSend(uint32_t ssrc, |
185 bool enable, | 183 bool enable, |
186 const AudioOptions* options, | 184 const AudioOptions* options, |
187 AudioSource* source) override; | 185 AudioSource* source) override; |
188 bool AddSendStream(const StreamParams& sp) override; | 186 bool AddSendStream(const StreamParams& sp) override; |
189 bool RemoveSendStream(uint32_t ssrc) override; | 187 bool RemoveSendStream(uint32_t ssrc) override; |
190 bool AddRecvStream(const StreamParams& sp) override; | 188 bool AddRecvStream(const StreamParams& sp) override; |
191 bool RemoveRecvStream(uint32_t ssrc) override; | 189 bool RemoveRecvStream(uint32_t ssrc) override; |
192 bool GetActiveStreams(AudioInfo::StreamList* actives) override; | 190 bool GetActiveStreams(AudioInfo::StreamList* actives) override; |
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238 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); | 236 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); |
239 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); | 237 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); |
240 bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters); | 238 bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters); |
241 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); | 239 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); |
242 bool SetLocalSource(uint32_t ssrc, AudioSource* source); | 240 bool SetLocalSource(uint32_t ssrc, AudioSource* source); |
243 bool MuteStream(uint32_t ssrc, bool mute); | 241 bool MuteStream(uint32_t ssrc, bool mute); |
244 | 242 |
245 WebRtcVoiceEngine* engine() { return engine_; } | 243 WebRtcVoiceEngine* engine() { return engine_; } |
246 int GetLastEngineError() { return engine()->GetLastEngineError(); } | 244 int GetLastEngineError() { return engine()->GetLastEngineError(); } |
247 int GetOutputLevel(int channel); | 245 int GetOutputLevel(int channel); |
248 bool SetPlayout(int channel, bool playout); | 246 void ChangePlayout(bool playout); |
249 bool ChangePlayout(bool playout); | |
250 int CreateVoEChannel(); | 247 int CreateVoEChannel(); |
251 bool DeleteVoEChannel(int channel); | 248 bool DeleteVoEChannel(int channel); |
252 bool IsDefaultRecvStream(uint32_t ssrc) { | 249 bool IsDefaultRecvStream(uint32_t ssrc) { |
253 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); | 250 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); |
254 } | 251 } |
255 bool SetMaxSendBitrate(int bps); | 252 bool SetMaxSendBitrate(int bps); |
256 bool SetChannelSendParameters(int channel, | 253 bool SetChannelSendParameters(int channel, |
257 const webrtc::RtpParameters& parameters); | 254 const webrtc::RtpParameters& parameters); |
258 bool SetMaxSendBitrate(int channel, int bps); | 255 bool SetMaxSendBitrate(int channel, int bps); |
259 bool HasSendCodec() const { | 256 bool HasSendCodec() const { |
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296 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 293 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
297 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 294 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
298 | 295 |
299 SendCodecSpec send_codec_spec_; | 296 SendCodecSpec send_codec_spec_; |
300 | 297 |
301 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 298 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
302 }; | 299 }; |
303 } // namespace cricket | 300 } // namespace cricket |
304 | 301 |
305 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 302 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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