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Issue 1378513003: Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 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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1251 } 1251 }
1252 1252
1253 void WebRtcSession::SetAudioSend(uint32 ssrc, bool enable, 1253 void WebRtcSession::SetAudioSend(uint32 ssrc, bool enable,
1254 const cricket::AudioOptions& options, 1254 const cricket::AudioOptions& options,
1255 cricket::AudioRenderer* renderer) { 1255 cricket::AudioRenderer* renderer) {
1256 ASSERT(signaling_thread()->IsCurrent()); 1256 ASSERT(signaling_thread()->IsCurrent());
1257 if (!voice_channel_) { 1257 if (!voice_channel_) {
1258 LOG(LS_ERROR) << "SetAudioSend: No audio channel exists."; 1258 LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
1259 return; 1259 return;
1260 } 1260 }
1261 if (!voice_channel_->SetAudioSend(ssrc, !enable, &options, renderer)) { 1261 if (!voice_channel_->SetAudioSend(ssrc, enable, &options, renderer)) {
1262 LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc; 1262 LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc;
1263 } 1263 }
1264 } 1264 }
1265 1265
1266 void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) { 1266 void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) {
1267 ASSERT(signaling_thread()->IsCurrent()); 1267 ASSERT(signaling_thread()->IsCurrent());
1268 ASSERT(volume >= 0 && volume <= 10); 1268 ASSERT(volume >= 0 && volume <= 10);
1269 if (!voice_channel_) { 1269 if (!voice_channel_) {
1270 LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists."; 1270 LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
1271 return; 1271 return;
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1312 } 1312 }
1313 } 1313 }
1314 1314
1315 void WebRtcSession::SetVideoSend(uint32 ssrc, bool enable, 1315 void WebRtcSession::SetVideoSend(uint32 ssrc, bool enable,
1316 const cricket::VideoOptions* options) { 1316 const cricket::VideoOptions* options) {
1317 ASSERT(signaling_thread()->IsCurrent()); 1317 ASSERT(signaling_thread()->IsCurrent());
1318 if (!video_channel_) { 1318 if (!video_channel_) {
1319 LOG(LS_WARNING) << "SetVideoSend: No video channel exists."; 1319 LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
1320 return; 1320 return;
1321 } 1321 }
1322 if (!video_channel_->SetVideoSend(ssrc, !enable, options)) { 1322 if (!video_channel_->SetVideoSend(ssrc, enable, options)) {
1323 // Allow that MuteStream fail if |enable| is false but assert otherwise. 1323 // Allow that MuteStream fail if |enable| is false but assert otherwise.
1324 // This in the normal case when the underlying media channel has already 1324 // This in the normal case when the underlying media channel has already
1325 // been deleted. 1325 // been deleted.
1326 ASSERT(enable == false); 1326 ASSERT(enable == false);
1327 } 1327 }
1328 } 1328 }
1329 1329
1330 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) { 1330 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
1331 ASSERT(signaling_thread()->IsCurrent()); 1331 ASSERT(signaling_thread()->IsCurrent());
1332 if (!voice_channel_) { 1332 if (!voice_channel_) {
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2173 if (!srtp_cipher.empty()) { 2173 if (!srtp_cipher.empty()) {
2174 metrics_observer_->IncrementSparseEnumCounter( 2174 metrics_observer_->IncrementSparseEnumCounter(
2175 srtp_counter_type, rtc::GetSrtpCryptoSuiteFromName(srtp_cipher)); 2175 srtp_counter_type, rtc::GetSrtpCryptoSuiteFromName(srtp_cipher));
2176 } 2176 }
2177 if (ssl_cipher) { 2177 if (ssl_cipher) {
2178 metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type, ssl_cipher); 2178 metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type, ssl_cipher);
2179 } 2179 }
2180 } 2180 }
2181 2181
2182 } // namespace webrtc 2182 } // namespace webrtc
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