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

Issue 1393343003: Revert of Remove AudioTrackRenderer (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 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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1298 // callback. 1298 // callback.
1299 // This method is called on the libjingle worker thread. 1299 // This method is called on the libjingle worker thread.
1300 // TODO(xians): Make sure Start() is called only once. 1300 // TODO(xians): Make sure Start() is called only once.
1301 void Start(AudioRenderer* renderer) { 1301 void Start(AudioRenderer* renderer) {
1302 rtc::CritScope lock(&lock_); 1302 rtc::CritScope lock(&lock_);
1303 RTC_DCHECK(renderer != NULL); 1303 RTC_DCHECK(renderer != NULL);
1304 if (renderer_ != NULL) { 1304 if (renderer_ != NULL) {
1305 RTC_DCHECK(renderer_ == renderer); 1305 RTC_DCHECK(renderer_ == renderer);
1306 return; 1306 return;
1307 } 1307 }
1308
1309 // TODO(xians): Remove AddChannel() call after Chrome turns on APM
1310 // in getUserMedia by default.
1311 renderer->AddChannel(channel_);
1308 renderer->SetSink(this); 1312 renderer->SetSink(this);
1309 renderer_ = renderer; 1313 renderer_ = renderer;
1310 } 1314 }
1311 1315
1312 // Stops rendering by setting the sink of the renderer to NULL. No data 1316 // Stops rendering by setting the sink of the renderer to NULL. No data
1313 // callback will be received after this method. 1317 // callback will be received after this method.
1314 // This method is called on the libjingle worker thread. 1318 // This method is called on the libjingle worker thread.
1315 void Stop() { 1319 void Stop() {
1316 rtc::CritScope lock(&lock_); 1320 rtc::CritScope lock(&lock_);
1317 if (renderer_ != NULL) { 1321 if (renderer_ == NULL)
1318 renderer_->SetSink(NULL); 1322 return;
1319 renderer_ = NULL; 1323
1320 } 1324 renderer_->RemoveChannel(channel_);
1325 renderer_->SetSink(NULL);
1326 renderer_ = NULL;
1321 } 1327 }
1322 1328
1323 // AudioRenderer::Sink implementation. 1329 // AudioRenderer::Sink implementation.
1324 // This method is called on the audio thread. 1330 // This method is called on the audio thread.
1325 void OnData(const void* audio_data, 1331 void OnData(const void* audio_data,
1326 int bits_per_sample, 1332 int bits_per_sample,
1327 int sample_rate, 1333 int sample_rate,
1328 int number_of_channels, 1334 int number_of_channels,
1329 size_t number_of_frames) override { 1335 size_t number_of_frames) override {
1330 voe_audio_transport_->OnData(channel_, 1336 voe_audio_transport_->OnData(channel_,
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3214 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); 3220 LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
3215 return false; 3221 return false;
3216 } 3222 }
3217 } 3223 }
3218 return true; 3224 return true;
3219 } 3225 }
3220 3226
3221 } // namespace cricket 3227 } // namespace cricket
3222 3228
3223 #endif // HAVE_WEBRTC_VOICE 3229 #endif // HAVE_WEBRTC_VOICE
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