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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/test/webrtc_audio_device_test.h" | 5 #include "content/test/webrtc_audio_device_test.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/bind_helpers.h" | 8 #include "base/bind_helpers.h" |
9 #include "base/compiler_specific.h" | 9 #include "base/compiler_specific.h" |
10 #include "base/file_util.h" | 10 #include "base/file_util.h" |
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217 has_input_devices_ = audio_manager_->HasAudioInputDevices(); | 217 has_input_devices_ = audio_manager_->HasAudioInputDevices(); |
218 has_output_devices_ = audio_manager_->HasAudioOutputDevices(); | 218 has_output_devices_ = audio_manager_->HasAudioOutputDevices(); |
219 | 219 |
220 // Create an IPC channel that handles incoming messages on the IO thread. | 220 // Create an IPC channel that handles incoming messages on the IO thread. |
221 CreateChannel(thread_name); | 221 CreateChannel(thread_name); |
222 } | 222 } |
223 | 223 |
224 void WebRTCAudioDeviceTest::UninitializeIOThread() { | 224 void WebRTCAudioDeviceTest::UninitializeIOThread() { |
225 resource_context_.reset(); | 225 resource_context_.reset(); |
226 | 226 |
227 media_stream_manager_.reset(); | |
tommi (sloooow) - chröme
2012/07/17 15:36:58
hmm... should we reset this object and the audio_m
no longer working on chromium
2012/07/17 17:18:25
Good idea, it fits the current design more.
Anothe
| |
228 audio_manager_.reset(); | 227 audio_manager_.reset(); |
229 test_request_context_.reset(); | 228 test_request_context_.reset(); |
230 initialize_com_.reset(); | 229 initialize_com_.reset(); |
231 } | 230 } |
232 | 231 |
233 void WebRTCAudioDeviceTest::CreateChannel(const char* name) { | 232 void WebRTCAudioDeviceTest::CreateChannel(const char* name) { |
234 DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO)); | 233 DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO)); |
235 audio_render_host_ = new AudioRendererHost( | 234 audio_render_host_ = new AudioRendererHost( |
236 audio_manager_.get(), media_observer_.get()); | 235 audio_manager_.get(), media_observer_.get()); |
237 audio_render_host_->OnChannelConnected(base::GetCurrentProcId()); | 236 audio_render_host_->OnChannelConnected(base::GetCurrentProcId()); |
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366 WebRTCTransportImpl::~WebRTCTransportImpl() {} | 365 WebRTCTransportImpl::~WebRTCTransportImpl() {} |
367 | 366 |
368 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 367 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
369 return network_->ReceivedRTPPacket(channel, data, len); | 368 return network_->ReceivedRTPPacket(channel, data, len); |
370 } | 369 } |
371 | 370 |
372 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 371 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
373 int len) { | 372 int len) { |
374 return network_->ReceivedRTCPPacket(channel, data, len); | 373 return network_->ReceivedRTCPPacket(channel, data, len); |
375 } | 374 } |
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