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Issue 9221010: Adds support for 16kHz input sample rate and mono channel config. in WebRTC. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Removed more size_t Created 8 years, 11 months ago
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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"
(...skipping 217 matching lines...) Expand 10 before | Expand all | Expand 10 after
228 *sample_rate = audio_util_callback_ ? 228 *sample_rate = audio_util_callback_ ?
229 audio_util_callback_->GetAudioHardwareSampleRate() : 0.0; 229 audio_util_callback_->GetAudioHardwareSampleRate() : 0.0;
230 } 230 }
231 231
232 void WebRTCAudioDeviceTest::OnGetHardwareInputSampleRate(double* sample_rate) { 232 void WebRTCAudioDeviceTest::OnGetHardwareInputSampleRate(double* sample_rate) {
233 EXPECT_TRUE(audio_util_callback_); 233 EXPECT_TRUE(audio_util_callback_);
234 *sample_rate = audio_util_callback_ ? 234 *sample_rate = audio_util_callback_ ?
235 audio_util_callback_->GetAudioInputHardwareSampleRate() : 0.0; 235 audio_util_callback_->GetAudioInputHardwareSampleRate() : 0.0;
236 } 236 }
237 237
238 void WebRTCAudioDeviceTest::OnGetHardwareInputChannelCount(uint32* channels) {
239 EXPECT_TRUE(audio_util_callback_);
240 *channels = audio_util_callback_ ?
241 audio_util_callback_->GetAudioInputHardwareChannelCount() : 0;
242 }
243
238 // IPC::Channel::Listener implementation. 244 // IPC::Channel::Listener implementation.
239 bool WebRTCAudioDeviceTest::OnMessageReceived(const IPC::Message& message) { 245 bool WebRTCAudioDeviceTest::OnMessageReceived(const IPC::Message& message) {
240 if (render_thread_) { 246 if (render_thread_) {
241 IPC::ChannelProxy::MessageFilter* filter = 247 IPC::ChannelProxy::MessageFilter* filter =
242 render_thread_->audio_input_message_filter(); 248 render_thread_->audio_input_message_filter();
243 if (filter->OnMessageReceived(message)) 249 if (filter->OnMessageReceived(message))
244 return true; 250 return true;
245 251
246 filter = render_thread_->audio_message_filter(); 252 filter = render_thread_->audio_message_filter();
247 if (filter->OnMessageReceived(message)) 253 if (filter->OnMessageReceived(message))
(...skipping 12 matching lines...) Expand all
260 return true; 266 return true;
261 } 267 }
262 268
263 bool handled ALLOW_UNUSED = true; 269 bool handled ALLOW_UNUSED = true;
264 bool message_is_ok = true; 270 bool message_is_ok = true;
265 IPC_BEGIN_MESSAGE_MAP_EX(WebRTCAudioDeviceTest, message, message_is_ok) 271 IPC_BEGIN_MESSAGE_MAP_EX(WebRTCAudioDeviceTest, message, message_is_ok)
266 IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareSampleRate, 272 IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareSampleRate,
267 OnGetHardwareSampleRate) 273 OnGetHardwareSampleRate)
268 IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputSampleRate, 274 IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputSampleRate,
269 OnGetHardwareInputSampleRate) 275 OnGetHardwareInputSampleRate)
276 IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputChannelCount,
277 OnGetHardwareInputChannelCount)
270 IPC_MESSAGE_UNHANDLED(handled = false) 278 IPC_MESSAGE_UNHANDLED(handled = false)
271 IPC_END_MESSAGE_MAP_EX() 279 IPC_END_MESSAGE_MAP_EX()
272 280
273 EXPECT_TRUE(message_is_ok); 281 EXPECT_TRUE(message_is_ok);
274 282
275 return true; 283 return true;
276 } 284 }
277 285
278 // Posts a final task to the IO message loop and waits for completion. 286 // Posts a final task to the IO message loop and waits for completion.
279 void WebRTCAudioDeviceTest::WaitForIOThreadCompletion() { 287 void WebRTCAudioDeviceTest::WaitForIOThreadCompletion() {
(...skipping 41 matching lines...) Expand 10 before | Expand all | Expand 10 after
321 WebRTCTransportImpl::~WebRTCTransportImpl() {} 329 WebRTCTransportImpl::~WebRTCTransportImpl() {}
322 330
323 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { 331 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
324 return network_->ReceivedRTPPacket(channel, data, len); 332 return network_->ReceivedRTPPacket(channel, data, len);
325 } 333 }
326 334
327 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, 335 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
328 int len) { 336 int len) {
329 return network_->ReceivedRTCPPacket(channel, data, len); 337 return network_->ReceivedRTCPPacket(channel, data, len);
330 } 338 }
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