Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(179)

Side by Side Diff: webrtc/media/base/fakemediaengine.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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/audio/audio_receive_stream.cc ('k') | webrtc/media/base/mediachannel.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 301 matching lines...) Expand 10 before | Expand all | Expand 10 after
312 SetSendRtpHeaderExtensions(params.extensions) && 312 SetSendRtpHeaderExtensions(params.extensions) &&
313 SetMaxSendBandwidth(params.max_bandwidth_bps) && 313 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
314 SetOptions(params.options)); 314 SetOptions(params.options));
315 } 315 }
316 316
317 virtual bool SetRecvParameters(const AudioRecvParameters& params) { 317 virtual bool SetRecvParameters(const AudioRecvParameters& params) {
318 return (SetRecvCodecs(params.codecs) && 318 return (SetRecvCodecs(params.codecs) &&
319 SetRecvRtpHeaderExtensions(params.extensions)); 319 SetRecvRtpHeaderExtensions(params.extensions));
320 } 320 }
321 321
322 virtual bool SetPlayout(bool playout) { 322 virtual void SetPlayout(bool playout) { set_playout(playout); }
323 set_playout(playout);
324 return true;
325 }
326 virtual void SetSend(bool send) { set_sending(send); } 323 virtual void SetSend(bool send) { set_sending(send); }
327 virtual bool SetAudioSend(uint32_t ssrc, 324 virtual bool SetAudioSend(uint32_t ssrc,
328 bool enable, 325 bool enable,
329 const AudioOptions* options, 326 const AudioOptions* options,
330 AudioSource* source) { 327 AudioSource* source) {
331 if (!SetLocalSource(ssrc, source)) { 328 if (!SetLocalSource(ssrc, source)) {
332 return false; 329 return false;
333 } 330 }
334 if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) { 331 if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) {
335 return false; 332 return false;
(...skipping 626 matching lines...) Expand 10 before | Expand all | Expand 10 after
962 959
963 private: 960 private:
964 std::vector<FakeDataMediaChannel*> channels_; 961 std::vector<FakeDataMediaChannel*> channels_;
965 std::vector<DataCodec> data_codecs_; 962 std::vector<DataCodec> data_codecs_;
966 DataChannelType last_channel_type_; 963 DataChannelType last_channel_type_;
967 }; 964 };
968 965
969 } // namespace cricket 966 } // namespace cricket
970 967
971 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ 968 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
OLDNEW
« no previous file with comments | « webrtc/audio/audio_receive_stream.cc ('k') | webrtc/media/base/mediachannel.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698