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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 2206223002: Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated comments in test. Created 4 years, 4 months ago
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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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