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Side by Side Diff: talk/media/webrtc/webrtcvoiceengine.cc

Issue 1311533009: - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@lj_remove_typingmonitor_files
Patch Set: rebase+one more test Created 5 years, 3 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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2367 DCHECK(send == SEND_NOTHING); 2367 DCHECK(send == SEND_NOTHING);
2368 if (engine()->voe()->base()->StopSend(channel) == -1) { 2368 if (engine()->voe()->base()->StopSend(channel) == -1) {
2369 LOG_RTCERR1(StopSend, channel); 2369 LOG_RTCERR1(StopSend, channel);
2370 return false; 2370 return false;
2371 } 2371 }
2372 } 2372 }
2373 2373
2374 return true; 2374 return true;
2375 } 2375 }
2376 2376
2377 bool WebRtcVoiceMediaChannel::SetAudioSend(uint32 ssrc, bool mute,
2378 const AudioOptions* options,
2379 AudioRenderer* renderer) {
2380 if (!SetLocalRenderer(ssrc, renderer)) {
2381 return false;
2382 }
2383 if (!MuteStream(ssrc, mute)) {
2384 return false;
pthatcher1 2015/09/09 14:25:14 Should we rollback the renderer if MuteStream fail
the sun 2015/09/09 14:52:11 See discussion in WebRtcVideoEngine2.cc.
2385 }
2386 if (!mute && options) {
2387 return SetOptions(*options);
pthatcher1 2015/09/09 14:25:14 Should we rollback the mute state and renderer if
the sun 2015/09/09 14:52:11 See discussion in WebRtcVideoEngine2.cc.
2388 } else {
pthatcher1 2015/09/09 14:25:14 Same here as before: could remove the "else".
the sun 2015/09/09 14:52:11 Done.
2389 return true;
2390 }
2391 }
2392
2377 // TODO(ronghuawu): Change this method to return bool. 2393 // TODO(ronghuawu): Change this method to return bool.
2378 void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) { 2394 void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
2379 if (engine()->voe()->network()->RegisterExternalTransport( 2395 if (engine()->voe()->network()->RegisterExternalTransport(
2380 channel, *this) == -1) { 2396 channel, *this) == -1) {
2381 LOG_RTCERR2(RegisterExternalTransport, channel, this); 2397 LOG_RTCERR2(RegisterExternalTransport, channel, this);
2382 } 2398 }
2383 2399
2384 // Enable RTCP (for quality stats and feedback messages) 2400 // Enable RTCP (for quality stats and feedback messages)
2385 EnableRtcp(channel); 2401 EnableRtcp(channel);
2386 2402
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3622 3638
3623 int WebRtcSoundclipStream::Rewind() { 3639 int WebRtcSoundclipStream::Rewind() {
3624 mem_.Rewind(); 3640 mem_.Rewind();
3625 // Return -1 to keep VoiceEngine from looping. 3641 // Return -1 to keep VoiceEngine from looping.
3626 return (loop_) ? 0 : -1; 3642 return (loop_) ? 0 : -1;
3627 } 3643 }
3628 3644
3629 } // namespace cricket 3645 } // namespace cricket
3630 3646
3631 #endif // HAVE_WEBRTC_VOICE 3647 #endif // HAVE_WEBRTC_VOICE
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