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Issue 1399553003: Reland: Remove AudioTrackRenderer (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 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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1323 // callback. 1323 // callback.
1324 // This method is called on the libjingle worker thread. 1324 // This method is called on the libjingle worker thread.
1325 // TODO(xians): Make sure Start() is called only once. 1325 // TODO(xians): Make sure Start() is called only once.
1326 void Start(AudioRenderer* renderer) { 1326 void Start(AudioRenderer* renderer) {
1327 rtc::CritScope lock(&lock_); 1327 rtc::CritScope lock(&lock_);
1328 RTC_DCHECK(renderer != NULL); 1328 RTC_DCHECK(renderer != NULL);
1329 if (renderer_ != NULL) { 1329 if (renderer_ != NULL) {
1330 RTC_DCHECK(renderer_ == renderer); 1330 RTC_DCHECK(renderer_ == renderer);
1331 return; 1331 return;
1332 } 1332 }
1333
1334 // TODO(xians): Remove AddChannel() call after Chrome turns on APM
1335 // in getUserMedia by default.
1336 renderer->AddChannel(channel_);
1337 renderer->SetSink(this); 1333 renderer->SetSink(this);
1338 renderer_ = renderer; 1334 renderer_ = renderer;
1339 } 1335 }
1340 1336
1341 // Stops rendering by setting the sink of the renderer to NULL. No data 1337 // Stops rendering by setting the sink of the renderer to NULL. No data
1342 // callback will be received after this method. 1338 // callback will be received after this method.
1343 // This method is called on the libjingle worker thread. 1339 // This method is called on the libjingle worker thread.
1344 void Stop() { 1340 void Stop() {
1345 rtc::CritScope lock(&lock_); 1341 rtc::CritScope lock(&lock_);
1346 if (renderer_ == NULL) 1342 if (renderer_ != NULL) {
1347 return; 1343 renderer_->SetSink(NULL);
1348 1344 renderer_ = NULL;
1349 renderer_->RemoveChannel(channel_); 1345 }
1350 renderer_->SetSink(NULL);
1351 renderer_ = NULL;
1352 } 1346 }
1353 1347
1354 // AudioRenderer::Sink implementation. 1348 // AudioRenderer::Sink implementation.
1355 // This method is called on the audio thread. 1349 // This method is called on the audio thread.
1356 void OnData(const void* audio_data, 1350 void OnData(const void* audio_data,
1357 int bits_per_sample, 1351 int bits_per_sample,
1358 int sample_rate, 1352 int sample_rate,
1359 int number_of_channels, 1353 int number_of_channels,
1360 size_t number_of_frames) override { 1354 size_t number_of_frames) override {
1361 voe_audio_transport_->OnData(channel_, 1355 voe_audio_transport_->OnData(channel_,
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3184 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); 3178 LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
3185 return false; 3179 return false;
3186 } 3180 }
3187 } 3181 }
3188 return true; 3182 return true;
3189 } 3183 }
3190 3184
3191 } // namespace cricket 3185 } // namespace cricket
3192 3186
3193 #endif // HAVE_WEBRTC_VOICE 3187 #endif // HAVE_WEBRTC_VOICE
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