OLD | NEW |
1 // Copyright (c) 2011 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2011 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/renderer/media/webrtc_audio_device_impl.h" | 5 #include "content/renderer/media/webrtc_audio_device_impl.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/string_util.h" | 8 #include "base/string_util.h" |
| 9 #include "base/win/windows_version.h" |
9 #include "content/common/view_messages.h" | 10 #include "content/common/view_messages.h" |
10 #include "content/renderer/render_thread_impl.h" | 11 #include "content/renderer/render_thread_impl.h" |
11 #include "media/audio/audio_util.h" | 12 #include "media/audio/audio_util.h" |
12 | 13 |
13 static const int64 kMillisecondsBetweenProcessCalls = 5000; | 14 static const int64 kMillisecondsBetweenProcessCalls = 5000; |
14 static const char kVersion[] = "WebRTC AudioDevice 1.0.0.Chrome"; | 15 static const char kVersion[] = "WebRTC AudioDevice 1.0.0.Chrome"; |
15 | 16 |
16 static int GetAudioInputHardwareSampleRate() { | 17 static int GetAudioInputHardwareSampleRate() { |
17 static double input_sample_rate = 0; | 18 static double input_sample_rate = 0; |
18 if (!input_sample_rate) { | 19 if (!input_sample_rate) { |
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33 input_sample_rate_(0), | 34 input_sample_rate_(0), |
34 output_sample_rate_(0), | 35 output_sample_rate_(0), |
35 input_delay_ms_(0), | 36 input_delay_ms_(0), |
36 output_delay_ms_(0), | 37 output_delay_ms_(0), |
37 last_error_(AudioDeviceModule::kAdmErrNone), | 38 last_error_(AudioDeviceModule::kAdmErrNone), |
38 last_process_time_(base::TimeTicks::Now()), | 39 last_process_time_(base::TimeTicks::Now()), |
39 session_id_(0), | 40 session_id_(0), |
40 initialized_(false), | 41 initialized_(false), |
41 playing_(false), | 42 playing_(false), |
42 recording_(false) { | 43 recording_(false) { |
43 VLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()"; | 44 DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()"; |
44 DCHECK(RenderThreadImpl::current()) << | 45 DCHECK(RenderThreadImpl::current()) << |
45 "WebRtcAudioDeviceImpl must be constructed on the render thread"; | 46 "WebRtcAudioDeviceImpl must be constructed on the render thread"; |
46 } | 47 } |
47 | 48 |
48 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() { | 49 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() { |
49 VLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()"; | 50 DVLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()"; |
50 if (playing_) | 51 if (playing_) |
51 StopPlayout(); | 52 StopPlayout(); |
52 if (recording_) | 53 if (recording_) |
53 StopRecording(); | 54 StopRecording(); |
54 if (initialized_) | 55 if (initialized_) |
55 Terminate(); | 56 Terminate(); |
56 } | 57 } |
57 | 58 |
58 int32_t WebRtcAudioDeviceImpl::AddRef() { | 59 int32_t WebRtcAudioDeviceImpl::AddRef() { |
59 return base::subtle::Barrier_AtomicIncrement(&ref_count_, 1); | 60 return base::subtle::Barrier_AtomicIncrement(&ref_count_, 1); |
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167 input_delay_ms_ + output_delay_ms_, | 168 input_delay_ms_ + output_delay_ms_, |
168 0, // clock_drift | 169 0, // clock_drift |
169 0, // current_mic_level | 170 0, // current_mic_level |
170 new_mic_level); // not used | 171 new_mic_level); // not used |
171 accumulated_audio_samples += samples_per_10_msec; | 172 accumulated_audio_samples += samples_per_10_msec; |
172 audio_byte_buffer += bytes_per_10_msec; | 173 audio_byte_buffer += bytes_per_10_msec; |
173 } | 174 } |
174 } | 175 } |
175 | 176 |
176 void WebRtcAudioDeviceImpl::OnDeviceStarted(int device_index) { | 177 void WebRtcAudioDeviceImpl::OnDeviceStarted(int device_index) { |
177 VLOG(1) << "OnDeviceStarted (device_index=" << device_index << ")"; | 178 DVLOG(1) << "OnDeviceStarted (device_index=" << device_index << ")"; |
178 // -1 is an invalid device index. Do nothing if a valid device has | 179 // -1 is an invalid device index. Do nothing if a valid device has |
179 // been started. Otherwise update the |recording_| state to false. | 180 // been started. Otherwise update the |recording_| state to false. |
180 if (device_index != -1) | 181 if (device_index != -1) |
181 return; | 182 return; |
182 | 183 |
183 base::AutoLock auto_lock(lock_); | 184 base::AutoLock auto_lock(lock_); |
184 if (recording_) | 185 if (recording_) |
185 recording_ = false; | 186 recording_ = false; |
186 } | 187 } |
187 | 188 |
188 void WebRtcAudioDeviceImpl::OnDeviceStopped() { | 189 void WebRtcAudioDeviceImpl::OnDeviceStopped() { |
189 VLOG(1) << "OnDeviceStopped"; | 190 DVLOG(1) << "OnDeviceStopped"; |
190 base::AutoLock auto_lock(lock_); | 191 base::AutoLock auto_lock(lock_); |
191 if (recording_) | 192 if (recording_) |
192 recording_ = false; | 193 recording_ = false; |
193 } | 194 } |
194 | 195 |
195 int32_t WebRtcAudioDeviceImpl::Version(char* version, | 196 int32_t WebRtcAudioDeviceImpl::Version(char* version, |
196 uint32_t& remaining_buffer_in_bytes, | 197 uint32_t& remaining_buffer_in_bytes, |
197 uint32_t& position) const { | 198 uint32_t& position) const { |
198 VLOG(1) << "Version()"; | 199 DVLOG(1) << "Version()"; |
199 DCHECK(version); | 200 DCHECK(version); |
200 if (version == NULL) | 201 if (version == NULL) |
201 return -1; | 202 return -1; |
202 size_t arr_size = arraysize(kVersion); | 203 size_t arr_size = arraysize(kVersion); |
203 if (remaining_buffer_in_bytes < arr_size) { | 204 if (remaining_buffer_in_bytes < arr_size) { |
204 DLOG(WARNING) << "version string requires " << arr_size << " bytes"; | 205 DLOG(WARNING) << "version string requires " << arr_size << " bytes"; |
205 return -1; | 206 return -1; |
206 } | 207 } |
207 base::strlcpy(&version[position], kVersion, arr_size - 1); | 208 base::strlcpy(&version[position], kVersion, arr_size - 1); |
208 remaining_buffer_in_bytes -= arr_size; | 209 remaining_buffer_in_bytes -= arr_size; |
209 position += arr_size; | 210 position += arr_size; |
210 VLOG(1) << "version: " << version; | 211 DVLOG(1) << "version: " << version; |
211 return 0; | 212 return 0; |
212 } | 213 } |
213 | 214 |
214 int32_t WebRtcAudioDeviceImpl::ChangeUniqueId(const int32_t id) { | 215 int32_t WebRtcAudioDeviceImpl::ChangeUniqueId(const int32_t id) { |
215 NOTIMPLEMENTED(); | 216 NOTIMPLEMENTED(); |
216 return -1; | 217 return -1; |
217 } | 218 } |
218 | 219 |
219 int32_t WebRtcAudioDeviceImpl::TimeUntilNextProcess() { | 220 int32_t WebRtcAudioDeviceImpl::TimeUntilNextProcess() { |
220 // Calculate the number of milliseconds until this module wants its | 221 // Calculate the number of milliseconds until this module wants its |
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238 NOTIMPLEMENTED(); | 239 NOTIMPLEMENTED(); |
239 return -1; | 240 return -1; |
240 } | 241 } |
241 | 242 |
242 webrtc::AudioDeviceModule::ErrorCode WebRtcAudioDeviceImpl::LastError() const { | 243 webrtc::AudioDeviceModule::ErrorCode WebRtcAudioDeviceImpl::LastError() const { |
243 return last_error_; | 244 return last_error_; |
244 } | 245 } |
245 | 246 |
246 int32_t WebRtcAudioDeviceImpl::RegisterEventObserver( | 247 int32_t WebRtcAudioDeviceImpl::RegisterEventObserver( |
247 webrtc::AudioDeviceObserver* event_callback) { | 248 webrtc::AudioDeviceObserver* event_callback) { |
248 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::RegisterEventObserver() " | 249 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::RegisterEventObserver() " |
249 << "NOT IMPLEMENTED"; | 250 << "NOT IMPLEMENTED"; |
250 return -1; | 251 return -1; |
251 } | 252 } |
252 | 253 |
253 int32_t WebRtcAudioDeviceImpl::RegisterAudioCallback( | 254 int32_t WebRtcAudioDeviceImpl::RegisterAudioCallback( |
254 webrtc::AudioTransport* audio_callback) { | 255 webrtc::AudioTransport* audio_callback) { |
255 VLOG(1) << "RegisterAudioCallback()"; | 256 DVLOG(1) << "RegisterAudioCallback()"; |
256 if (playing_ || recording_) { | 257 if (playing_ || recording_) { |
257 LOG(ERROR) << "Unable to (de)register transport during active media"; | 258 LOG(ERROR) << "Unable to (de)register transport during active media"; |
258 return -1; | 259 return -1; |
259 } | 260 } |
260 audio_transport_callback_ = audio_callback; | 261 audio_transport_callback_ = audio_callback; |
261 return 0; | 262 return 0; |
262 } | 263 } |
263 | 264 |
264 int32_t WebRtcAudioDeviceImpl::Init() { | 265 int32_t WebRtcAudioDeviceImpl::Init() { |
265 VLOG(1) << "Init()"; | 266 DVLOG(1) << "Init()"; |
266 | 267 |
267 if (!render_loop_->BelongsToCurrentThread()) { | 268 if (!render_loop_->BelongsToCurrentThread()) { |
268 int32_t error = 0; | 269 int32_t error = 0; |
269 base::WaitableEvent event(false, false); | 270 base::WaitableEvent event(false, false); |
270 // Ensure that we call Init() from the main render thread since | 271 // Ensure that we call Init() from the main render thread since |
271 // the audio clients can only be created on this thread. | 272 // the audio clients can only be created on this thread. |
272 render_loop_->PostTask( | 273 render_loop_->PostTask( |
273 FROM_HERE, | 274 FROM_HERE, |
274 base::Bind(&WebRtcAudioDeviceImpl::InitOnRenderThread, | 275 base::Bind(&WebRtcAudioDeviceImpl::InitOnRenderThread, |
275 this, &error, &event)); | 276 this, &error, &event)); |
276 event.Wait(); | 277 event.Wait(); |
277 return error; | 278 return error; |
278 } | 279 } |
279 | 280 |
280 // Calling Init() multiple times in a row is OK. | 281 // Calling Init() multiple times in a row is OK. |
281 if (initialized_) | 282 if (initialized_) |
282 return 0; | 283 return 0; |
283 | 284 |
284 DCHECK(!audio_input_device_); | 285 DCHECK(!audio_input_device_); |
285 DCHECK(!audio_output_device_); | 286 DCHECK(!audio_output_device_); |
286 DCHECK(!input_buffer_.get()); | 287 DCHECK(!input_buffer_.get()); |
287 DCHECK(!output_buffer_.get()); | 288 DCHECK(!output_buffer_.get()); |
288 | 289 |
289 // Ask the browser for the default audio output hardware sample-rate. | 290 // Ask the browser for the default audio output hardware sample-rate. |
290 // This request is based on a synchronous IPC message. | 291 // This request is based on a synchronous IPC message. |
291 int output_sample_rate = | 292 int output_sample_rate = |
292 static_cast<int>(AudioDevice::GetAudioHardwareSampleRate()); | 293 static_cast<int>(AudioDevice::GetAudioHardwareSampleRate()); |
293 VLOG(1) << "Audio output hardware sample rate: " << output_sample_rate; | 294 DVLOG(1) << "Audio output hardware sample rate: " << output_sample_rate; |
294 | 295 |
295 // Ask the browser for the default audio input hardware sample-rate. | 296 // Ask the browser for the default audio input hardware sample-rate. |
296 // This request is based on a synchronous IPC message. | 297 // This request is based on a synchronous IPC message. |
297 int input_sample_rate = GetAudioInputHardwareSampleRate(); | 298 int input_sample_rate = GetAudioInputHardwareSampleRate(); |
298 VLOG(1) << "Audio input hardware sample rate: " << input_sample_rate; | 299 DVLOG(1) << "Audio input hardware sample rate: " << input_sample_rate; |
299 | 300 |
300 int input_channels = 0; | 301 int input_channels = 0; |
301 int output_channels = 0; | 302 int output_channels = 0; |
302 | 303 |
303 size_t input_buffer_size = 0; | 304 size_t input_buffer_size = 0; |
304 size_t output_buffer_size = 0; | 305 size_t output_buffer_size = 0; |
305 | 306 |
306 // For real-time audio (in combination with the webrtc::VoiceEngine) it | 307 // Windows |
307 // is convenient to use audio buffers of size N*10ms. | |
308 | |
309 #if defined(OS_WIN) | 308 #if defined(OS_WIN) |
310 if (output_sample_rate != 48000) { | 309 if (input_sample_rate != 48000 && input_sample_rate != 44100) { |
311 DLOG(ERROR) << "Only 48kHz sample rate is supported on Windows."; | 310 DLOG(ERROR) << "Only 48 and 44.1kHz input rates are supported on Windows."; |
| 311 return -1; |
| 312 } |
| 313 if (output_sample_rate != 48000 && output_sample_rate != 44100) { |
| 314 DLOG(ERROR) << "Only 48 and 44.1kHz output rates are supported on Windows."; |
312 return -1; | 315 return -1; |
313 } | 316 } |
314 | 317 |
315 // Use stereo recording on Windows since low-latency Core Audio (WASAPI) | 318 // Use stereo recording on Windows since low-latency Core Audio (WASAPI) |
316 // does not support mono. | 319 // does not support mono. |
317 input_channels = 2; | 320 input_channels = 2; |
318 output_channels = 1; | 321 |
| 322 // Use stereo rendering on Windows to make input and output sides |
| 323 // symmetric. WASAPI supports both stereo and mono. |
| 324 output_channels = 2; |
319 | 325 |
320 // Capture side: AUDIO_PCM_LOW_LATENCY is based on the Core Audio (WASAPI) | 326 // Capture side: AUDIO_PCM_LOW_LATENCY is based on the Core Audio (WASAPI) |
321 // API which was introduced in Windows Vista. For lower Windows versions, | 327 // API which was introduced in Windows Vista. For lower Windows versions, |
322 // a callback-driven Wave implementation is used instead. An input buffer | 328 // a callback-driven Wave implementation is used instead. An input buffer |
323 // size of 10ms works well for both these implementations. | 329 // size of 10ms works well for both these implementations. |
324 | 330 |
325 // Use different buffer sizes depending on the current hardware sample rate. | 331 // Use different buffer sizes depending on the current hardware sample rate. |
326 if (input_sample_rate == 48000) { | 332 if (input_sample_rate == 48000) { |
327 input_buffer_size = 480; | 333 input_buffer_size = 480; |
328 } else { | 334 } else { |
329 // We do run at 44.1kHz at the actual audio layer, but ask for frames | 335 // We do run at 44.1kHz at the actual audio layer, but ask for frames |
330 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. | 336 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. |
331 input_buffer_size = 440; | 337 input_buffer_size = 440; |
332 } | 338 } |
333 | 339 |
334 // Rendering side: AUDIO_PCM_LOW_LATENCY on Windows is based on a callback- | 340 // Render side: AUDIO_PCM_LOW_LATENCY is based on the Core Audio (WASAPI) |
335 // driven Wave implementation where 2 buffers are fed to the audio driver | 341 // API which was introduced in Windows Vista. For lower Windows versions, |
336 // before actual rendering starts. Initial real-time tests have shown that | 342 // a callback-driven Wave implementation is used instead. An output buffer |
337 // 20ms buffer size (corresponds to ~40ms total delay) is not enough but | 343 // size of 10ms works well for WASAPI but 30ms is needed for Wave. |
338 // can lead to buffer underruns. The next even multiple of 10ms is 30ms | |
339 // (<=> ~60ms total delay) and it works fine also under high load. | |
340 output_buffer_size = 3 * 480; | |
341 #elif defined(OS_MACOSX) | |
342 if (output_sample_rate != 48000 && output_sample_rate != 44100) { | |
343 DLOG(ERROR) << "Only 48 and 44.1kHz sample rates are supported on Mac OSX."; | |
344 return -1; | |
345 } | |
346 input_channels = 1; | |
347 output_channels = 1; | |
348 | |
349 // Rendering side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback- | |
350 // driven Core Audio implementation. Tests have shown that 10ms is a suitable | |
351 // frame size to use, both for 48kHz and 44.1kHz. | |
352 // Capturing side: AUDIO_PCM_LINEAR on Mac OS X uses the Audio Queue Services | |
353 // API which is not well suited for real-time applications since the delay | |
354 // is very high. We set buffer sizes to 10ms for the input side here as well | |
355 // but none of them will work. | |
356 // TODO(henrika): add support for AUDIO_PCM_LOW_LATENCY on the capture side | |
357 // based on the Mac OS X Core Audio API. | |
358 | 344 |
359 // Use different buffer sizes depending on the current hardware sample rate. | 345 // Use different buffer sizes depending on the current hardware sample rate. |
360 if (output_sample_rate == 48000) { | 346 if (output_sample_rate == 48000) { |
361 input_buffer_size = 480; | |
362 output_buffer_size = 480; | 347 output_buffer_size = 480; |
363 } else { | 348 } else { |
364 // We do run at 44.1kHz at the actual audio layer, but ask for frames | 349 // We do run at 44.1kHz at the actual audio layer, but ask for frames |
365 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. | 350 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. |
| 351 // TODO(henrika): figure out why we seem to need 20ms here for glitch- |
| 352 // free audio. |
| 353 output_buffer_size = 2 * 440; |
| 354 } |
| 355 |
| 356 // Windows XP and lower can't cope with 10 ms output buffer size. |
| 357 // It must be extended to 30 ms (60 ms will be used internally by WaveOut). |
| 358 if (base::win::GetVersion() <= base::win::VERSION_XP) { |
| 359 output_buffer_size = 3 * output_buffer_size; |
| 360 DLOG(WARNING) << "Extending the output buffer size by a factor of three " |
| 361 << "since Windows XP has been detected."; |
| 362 } |
| 363 |
| 364 // Mac OS X |
| 365 #elif defined(OS_MACOSX) |
| 366 if (input_sample_rate != 48000 && input_sample_rate != 44100) { |
| 367 DLOG(ERROR) << "Only 48 and 44.1kHz input rates are supported on Mac OSX."; |
| 368 return -1; |
| 369 } |
| 370 if (output_sample_rate != 48000 && output_sample_rate != 44100) { |
| 371 DLOG(ERROR) << "Only 48 and 44.1kHz output rates are supported on Mac OSX."; |
| 372 return -1; |
| 373 } |
| 374 |
| 375 input_channels = 1; |
| 376 output_channels = 1; |
| 377 |
| 378 // Capture side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback- |
| 379 // driven Core Audio implementation. Tests have shown that 10ms is a suitable |
| 380 // frame size to use, both for 48kHz and 44.1kHz. |
| 381 |
| 382 // Use different buffer sizes depending on the current hardware sample rate. |
| 383 if (input_sample_rate == 48000) { |
| 384 input_buffer_size = 480; |
| 385 } else { |
| 386 // We do run at 44.1kHz at the actual audio layer, but ask for frames |
| 387 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. |
366 input_buffer_size = 440; | 388 input_buffer_size = 440; |
| 389 } |
| 390 |
| 391 // Render side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback- |
| 392 // driven Core Audio implementation. Tests have shown that 10ms is a suitable |
| 393 // frame size to use, both for 48kHz and 44.1kHz. |
| 394 |
| 395 // Use different buffer sizes depending on the current hardware sample rate. |
| 396 if (output_sample_rate == 48000) { |
| 397 output_buffer_size = 480; |
| 398 } else { |
| 399 // We do run at 44.1kHz at the actual audio layer, but ask for frames |
| 400 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. |
367 output_buffer_size = 440; | 401 output_buffer_size = 440; |
368 } | 402 } |
| 403 |
| 404 // Linux |
369 #elif defined(OS_LINUX) | 405 #elif defined(OS_LINUX) |
370 if (output_sample_rate != 48000) { | 406 if (output_sample_rate != 48000) { |
371 DLOG(ERROR) << "Only 48kHz sample rate is supported on Linux."; | 407 DLOG(ERROR) << "Only 48kHz sample rate is supported on Linux."; |
372 return -1; | 408 return -1; |
373 } | 409 } |
374 input_channels = 1; | 410 input_channels = 1; |
375 output_channels = 1; | 411 output_channels = 1; |
376 | 412 |
377 // Based on tests using the current ALSA implementation in Chrome, we have | 413 // Based on tests using the current ALSA implementation in Chrome, we have |
378 // found that the best combination is 20ms on the input side and 10ms on the | 414 // found that the best combination is 20ms on the input side and 10ms on the |
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414 input_buffer_.reset(new int16[input_buffer_size * input_channels]); | 450 input_buffer_.reset(new int16[input_buffer_size * input_channels]); |
415 output_buffer_.reset(new int16[output_buffer_size * output_channels]); | 451 output_buffer_.reset(new int16[output_buffer_size * output_channels]); |
416 | 452 |
417 DCHECK(input_buffer_.get()); | 453 DCHECK(input_buffer_.get()); |
418 DCHECK(output_buffer_.get()); | 454 DCHECK(output_buffer_.get()); |
419 | 455 |
420 bytes_per_sample_ = sizeof(*input_buffer_.get()); | 456 bytes_per_sample_ = sizeof(*input_buffer_.get()); |
421 | 457 |
422 initialized_ = true; | 458 initialized_ = true; |
423 | 459 |
424 VLOG(1) << "Capture parameters (size/channels/rate): (" | 460 DVLOG(1) << "Capture parameters (size/channels/rate): (" |
425 << input_buffer_size_ << "/" << input_channels_ << "/" | 461 << input_buffer_size_ << "/" << input_channels_ << "/" |
426 << input_sample_rate_ << ")"; | 462 << input_sample_rate_ << ")"; |
427 VLOG(1) << "Render parameters (size/channels/rate): (" | 463 DVLOG(1) << "Render parameters (size/channels/rate): (" |
428 << output_buffer_size_ << "/" << output_channels_ << "/" | 464 << output_buffer_size_ << "/" << output_channels_ << "/" |
429 << output_sample_rate_ << ")"; | 465 << output_sample_rate_ << ")"; |
430 return 0; | 466 return 0; |
431 } | 467 } |
432 | 468 |
433 void WebRtcAudioDeviceImpl::InitOnRenderThread(int32_t* error, | 469 void WebRtcAudioDeviceImpl::InitOnRenderThread(int32_t* error, |
434 base::WaitableEvent* event) { | 470 base::WaitableEvent* event) { |
435 DCHECK(render_loop_->BelongsToCurrentThread()); | 471 DCHECK(render_loop_->BelongsToCurrentThread()); |
436 *error = Init(); | 472 *error = Init(); |
437 event->Signal(); | 473 event->Signal(); |
438 } | 474 } |
439 | 475 |
440 int32_t WebRtcAudioDeviceImpl::Terminate() { | 476 int32_t WebRtcAudioDeviceImpl::Terminate() { |
441 VLOG(1) << "Terminate()"; | 477 DVLOG(1) << "Terminate()"; |
442 | 478 |
443 // Calling Terminate() multiple times in a row is OK. | 479 // Calling Terminate() multiple times in a row is OK. |
444 if (!initialized_) | 480 if (!initialized_) |
445 return 0; | 481 return 0; |
446 | 482 |
447 DCHECK(audio_input_device_); | 483 DCHECK(audio_input_device_); |
448 DCHECK(audio_output_device_); | 484 DCHECK(audio_output_device_); |
449 DCHECK(input_buffer_.get()); | 485 DCHECK(input_buffer_.get()); |
450 DCHECK(output_buffer_.get()); | 486 DCHECK(output_buffer_.get()); |
451 | 487 |
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483 | 519 |
484 int32_t WebRtcAudioDeviceImpl::RecordingDeviceName( | 520 int32_t WebRtcAudioDeviceImpl::RecordingDeviceName( |
485 uint16_t index, | 521 uint16_t index, |
486 char name[webrtc::kAdmMaxDeviceNameSize], | 522 char name[webrtc::kAdmMaxDeviceNameSize], |
487 char guid[webrtc::kAdmMaxGuidSize]) { | 523 char guid[webrtc::kAdmMaxGuidSize]) { |
488 NOTIMPLEMENTED(); | 524 NOTIMPLEMENTED(); |
489 return -1; | 525 return -1; |
490 } | 526 } |
491 | 527 |
492 int32_t WebRtcAudioDeviceImpl::SetPlayoutDevice(uint16_t index) { | 528 int32_t WebRtcAudioDeviceImpl::SetPlayoutDevice(uint16_t index) { |
493 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetPlayoutDevice() " | 529 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetPlayoutDevice() " |
494 << "NOT IMPLEMENTED"; | 530 << "NOT IMPLEMENTED"; |
495 return 0; | 531 return 0; |
496 } | 532 } |
497 | 533 |
498 int32_t WebRtcAudioDeviceImpl::SetPlayoutDevice(WindowsDeviceType device) { | 534 int32_t WebRtcAudioDeviceImpl::SetPlayoutDevice(WindowsDeviceType device) { |
499 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetPlayoutDevice() " | 535 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetPlayoutDevice() " |
500 << "NOT IMPLEMENTED"; | 536 << "NOT IMPLEMENTED"; |
501 return 0; | 537 return 0; |
502 } | 538 } |
503 | 539 |
504 int32_t WebRtcAudioDeviceImpl::SetRecordingDevice(uint16_t index) { | 540 int32_t WebRtcAudioDeviceImpl::SetRecordingDevice(uint16_t index) { |
505 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetRecordingDevice() " | 541 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetRecordingDevice() " |
506 << "NOT IMPLEMENTED"; | 542 << "NOT IMPLEMENTED"; |
507 return 0; | 543 return 0; |
508 } | 544 } |
509 | 545 |
510 int32_t WebRtcAudioDeviceImpl::SetRecordingDevice(WindowsDeviceType device) { | 546 int32_t WebRtcAudioDeviceImpl::SetRecordingDevice(WindowsDeviceType device) { |
511 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetRecordingDevice() " | 547 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetRecordingDevice() " |
512 << "NOT IMPLEMENTED"; | 548 << "NOT IMPLEMENTED"; |
513 return 0; | 549 return 0; |
514 } | 550 } |
515 | 551 |
516 int32_t WebRtcAudioDeviceImpl::PlayoutIsAvailable(bool* available) { | 552 int32_t WebRtcAudioDeviceImpl::PlayoutIsAvailable(bool* available) { |
517 VLOG(1) << "PlayoutIsAvailable()"; | 553 DVLOG(1) << "PlayoutIsAvailable()"; |
518 *available = (audio_output_device_ != NULL); | 554 *available = (audio_output_device_ != NULL); |
519 return 0; | 555 return 0; |
520 } | 556 } |
521 | 557 |
522 int32_t WebRtcAudioDeviceImpl::InitPlayout() { | 558 int32_t WebRtcAudioDeviceImpl::InitPlayout() { |
523 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitPlayout() " | 559 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitPlayout() " |
524 << "NOT IMPLEMENTED"; | 560 << "NOT IMPLEMENTED"; |
525 return 0; | 561 return 0; |
526 } | 562 } |
527 | 563 |
528 bool WebRtcAudioDeviceImpl::PlayoutIsInitialized() const { | 564 bool WebRtcAudioDeviceImpl::PlayoutIsInitialized() const { |
529 VLOG(1) << "PlayoutIsInitialized()"; | 565 DVLOG(1) << "PlayoutIsInitialized()"; |
530 return (audio_output_device_ != NULL); | 566 return (audio_output_device_ != NULL); |
531 } | 567 } |
532 | 568 |
533 int32_t WebRtcAudioDeviceImpl::RecordingIsAvailable(bool* available) { | 569 int32_t WebRtcAudioDeviceImpl::RecordingIsAvailable(bool* available) { |
534 VLOG(1) << "RecordingIsAvailable()"; | 570 DVLOG(1) << "RecordingIsAvailable()"; |
535 *available = (audio_input_device_ != NULL); | 571 *available = (audio_input_device_ != NULL); |
536 return 0; | 572 return 0; |
537 } | 573 } |
538 | 574 |
539 int32_t WebRtcAudioDeviceImpl::InitRecording() { | 575 int32_t WebRtcAudioDeviceImpl::InitRecording() { |
540 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitRecording() " | 576 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitRecording() " |
541 << "NOT IMPLEMENTED"; | 577 << "NOT IMPLEMENTED"; |
542 return 0; | 578 return 0; |
543 } | 579 } |
544 | 580 |
545 bool WebRtcAudioDeviceImpl::RecordingIsInitialized() const { | 581 bool WebRtcAudioDeviceImpl::RecordingIsInitialized() const { |
546 VLOG(1) << "RecordingIsInitialized()"; | 582 DVLOG(1) << "RecordingIsInitialized()"; |
547 return (audio_input_device_ != NULL); | 583 return (audio_input_device_ != NULL); |
548 } | 584 } |
549 | 585 |
550 int32_t WebRtcAudioDeviceImpl::StartPlayout() { | 586 int32_t WebRtcAudioDeviceImpl::StartPlayout() { |
551 VLOG(1) << "StartPlayout()"; | 587 DVLOG(1) << "StartPlayout()"; |
552 if (!audio_transport_callback_) { | 588 if (!audio_transport_callback_) { |
553 LOG(ERROR) << "Audio transport is missing"; | 589 LOG(ERROR) << "Audio transport is missing"; |
554 return -1; | 590 return -1; |
555 } | 591 } |
556 if (playing_) { | 592 if (playing_) { |
557 // webrtc::VoiceEngine assumes that it is OK to call Start() twice and | 593 // webrtc::VoiceEngine assumes that it is OK to call Start() twice and |
558 // that the call is ignored the second time. | 594 // that the call is ignored the second time. |
559 LOG(WARNING) << "Playout is already active"; | 595 LOG(WARNING) << "Playout is already active"; |
560 return 0; | 596 return 0; |
561 } | 597 } |
562 audio_output_device_->Start(); | 598 audio_output_device_->Start(); |
563 playing_ = true; | 599 playing_ = true; |
564 return 0; | 600 return 0; |
565 } | 601 } |
566 | 602 |
567 int32_t WebRtcAudioDeviceImpl::StopPlayout() { | 603 int32_t WebRtcAudioDeviceImpl::StopPlayout() { |
568 VLOG(1) << "StopPlayout()"; | 604 DVLOG(1) << "StopPlayout()"; |
569 DCHECK(audio_output_device_); | 605 DCHECK(audio_output_device_); |
570 if (!playing_) { | 606 if (!playing_) { |
571 // webrtc::VoiceEngine assumes that it is OK to call Stop() just in case. | 607 // webrtc::VoiceEngine assumes that it is OK to call Stop() just in case. |
572 LOG(WARNING) << "Playout was already stopped"; | 608 LOG(WARNING) << "Playout was already stopped"; |
573 return 0; | 609 return 0; |
574 } | 610 } |
575 playing_ = !audio_output_device_->Stop(); | 611 playing_ = !audio_output_device_->Stop(); |
576 return (!playing_ ? 0 : -1); | 612 return (!playing_ ? 0 : -1); |
577 } | 613 } |
578 | 614 |
579 bool WebRtcAudioDeviceImpl::Playing() const { | 615 bool WebRtcAudioDeviceImpl::Playing() const { |
580 return playing_; | 616 return playing_; |
581 } | 617 } |
582 | 618 |
583 int32_t WebRtcAudioDeviceImpl::StartRecording() { | 619 int32_t WebRtcAudioDeviceImpl::StartRecording() { |
584 VLOG(1) << "StartRecording()"; | 620 DVLOG(1) << "StartRecording()"; |
585 #if defined(OS_MACOSX) | 621 #if defined(OS_MACOSX) |
586 DLOG(WARNING) << "Real-time recording is not yet fully supported on Mac OS X"; | 622 DLOG(WARNING) << "Real-time recording is not yet fully supported on Mac OS X"; |
587 #endif | 623 #endif |
588 LOG_IF(ERROR, !audio_transport_callback_) << "Audio transport is missing"; | 624 LOG_IF(ERROR, !audio_transport_callback_) << "Audio transport is missing"; |
589 if (!audio_transport_callback_) { | 625 if (!audio_transport_callback_) { |
590 LOG(ERROR) << "Audio transport is missing"; | 626 LOG(ERROR) << "Audio transport is missing"; |
591 return -1; | 627 return -1; |
592 } | 628 } |
593 | 629 |
594 if (session_id_ <= 0) { | 630 if (session_id_ <= 0) { |
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607 } | 643 } |
608 | 644 |
609 // Specify the session_id which is mapped to a certain device. | 645 // Specify the session_id which is mapped to a certain device. |
610 audio_input_device_->SetDevice(session_id_); | 646 audio_input_device_->SetDevice(session_id_); |
611 audio_input_device_->Start(); | 647 audio_input_device_->Start(); |
612 recording_ = true; | 648 recording_ = true; |
613 return 0; | 649 return 0; |
614 } | 650 } |
615 | 651 |
616 int32_t WebRtcAudioDeviceImpl::StopRecording() { | 652 int32_t WebRtcAudioDeviceImpl::StopRecording() { |
617 VLOG(1) << "StopRecording()"; | 653 DVLOG(1) << "StopRecording()"; |
618 DCHECK(audio_input_device_); | 654 DCHECK(audio_input_device_); |
619 | 655 |
620 base::AutoLock auto_lock(lock_); | 656 base::AutoLock auto_lock(lock_); |
621 if (!recording_) { | 657 if (!recording_) { |
622 // webrtc::VoiceEngine assumes that it is OK to call Stop() just in case. | 658 // webrtc::VoiceEngine assumes that it is OK to call Stop() just in case. |
623 LOG(WARNING) << "Recording was already stopped"; | 659 LOG(WARNING) << "Recording was already stopped"; |
624 return 0; | 660 return 0; |
625 } | 661 } |
626 recording_ = !audio_input_device_->Stop(); | 662 recording_ = !audio_input_device_->Stop(); |
627 return (!recording_ ? 0 : -1); | 663 return (!recording_ ? 0 : -1); |
628 } | 664 } |
629 | 665 |
630 bool WebRtcAudioDeviceImpl::Recording() const { | 666 bool WebRtcAudioDeviceImpl::Recording() const { |
631 return recording_; | 667 return recording_; |
632 } | 668 } |
633 | 669 |
634 int32_t WebRtcAudioDeviceImpl::SetAGC(bool enable) { | 670 int32_t WebRtcAudioDeviceImpl::SetAGC(bool enable) { |
635 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetAGC() " | 671 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetAGC() " << "NOT IMPLEMENTED"; |
636 << "NOT IMPLEMENTED"; | |
637 return -1; | 672 return -1; |
638 } | 673 } |
639 | 674 |
640 bool WebRtcAudioDeviceImpl::AGC() const { | 675 bool WebRtcAudioDeviceImpl::AGC() const { |
641 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::AGC() " | 676 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::AGC() " << "NOT IMPLEMENTED"; |
642 << "NOT IMPLEMENTED"; | |
643 return false; | 677 return false; |
644 } | 678 } |
645 | 679 |
646 int32_t WebRtcAudioDeviceImpl::SetWaveOutVolume(uint16_t volume_left, | 680 int32_t WebRtcAudioDeviceImpl::SetWaveOutVolume(uint16_t volume_left, |
647 uint16_t volume_right) { | 681 uint16_t volume_right) { |
648 NOTIMPLEMENTED(); | 682 NOTIMPLEMENTED(); |
649 return -1; | 683 return -1; |
650 } | 684 } |
651 int32_t WebRtcAudioDeviceImpl::WaveOutVolume( | 685 int32_t WebRtcAudioDeviceImpl::WaveOutVolume( |
652 uint16_t* volume_left, | 686 uint16_t* volume_left, |
653 uint16_t* volume_right) const { | 687 uint16_t* volume_right) const { |
654 NOTIMPLEMENTED(); | 688 NOTIMPLEMENTED(); |
655 return -1; | 689 return -1; |
656 } | 690 } |
657 | 691 |
658 int32_t WebRtcAudioDeviceImpl::SpeakerIsAvailable(bool* available) { | 692 int32_t WebRtcAudioDeviceImpl::SpeakerIsAvailable(bool* available) { |
659 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SpeakerIsAvailable() " | 693 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SpeakerIsAvailable() " |
660 << "NOT IMPLEMENTED"; | 694 << "NOT IMPLEMENTED"; |
661 *available = true; | 695 *available = true; |
662 return 0; | 696 return 0; |
663 } | 697 } |
664 | 698 |
665 int32_t WebRtcAudioDeviceImpl::InitSpeaker() { | 699 int32_t WebRtcAudioDeviceImpl::InitSpeaker() { |
666 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitSpeaker() " | 700 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitSpeaker() " |
667 << "NOT IMPLEMENTED"; | 701 << "NOT IMPLEMENTED"; |
668 return 0; | 702 return 0; |
669 } | 703 } |
670 | 704 |
671 bool WebRtcAudioDeviceImpl::SpeakerIsInitialized() const { | 705 bool WebRtcAudioDeviceImpl::SpeakerIsInitialized() const { |
672 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SpeakerIsInitialized() " | 706 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SpeakerIsInitialized() " |
673 << "NOT IMPLEMENTED"; | 707 << "NOT IMPLEMENTED"; |
674 return true; | 708 return true; |
675 } | 709 } |
676 | 710 |
677 int32_t WebRtcAudioDeviceImpl::MicrophoneIsAvailable(bool* available) { | 711 int32_t WebRtcAudioDeviceImpl::MicrophoneIsAvailable(bool* available) { |
678 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::MicrophoneIsAvailable() " | 712 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::MicrophoneIsAvailable() " |
679 << "NOT IMPLEMENTED"; | 713 << "NOT IMPLEMENTED"; |
680 *available = true; | 714 *available = true; |
681 return 0; | 715 return 0; |
682 } | 716 } |
683 | 717 |
684 int32_t WebRtcAudioDeviceImpl::InitMicrophone() { | 718 int32_t WebRtcAudioDeviceImpl::InitMicrophone() { |
685 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitMicrophone() " | 719 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::InitMicrophone() " |
686 << "NOT IMPLEMENTED"; | 720 << "NOT IMPLEMENTED"; |
687 return 0; | 721 return 0; |
688 } | 722 } |
689 | 723 |
690 bool WebRtcAudioDeviceImpl::MicrophoneIsInitialized() const { | 724 bool WebRtcAudioDeviceImpl::MicrophoneIsInitialized() const { |
691 NOTIMPLEMENTED(); | 725 NOTIMPLEMENTED(); |
692 return true; | 726 return true; |
693 } | 727 } |
694 | 728 |
695 int32_t WebRtcAudioDeviceImpl::SpeakerVolumeIsAvailable( | 729 int32_t WebRtcAudioDeviceImpl::SpeakerVolumeIsAvailable( |
696 bool* available) { | 730 bool* available) { |
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735 return -1; | 769 return -1; |
736 } | 770 } |
737 | 771 |
738 int32_t WebRtcAudioDeviceImpl::MicrophoneVolume(uint32_t* volume) const { | 772 int32_t WebRtcAudioDeviceImpl::MicrophoneVolume(uint32_t* volume) const { |
739 NOTIMPLEMENTED(); | 773 NOTIMPLEMENTED(); |
740 return -1; | 774 return -1; |
741 } | 775 } |
742 | 776 |
743 int32_t WebRtcAudioDeviceImpl::MaxMicrophoneVolume( | 777 int32_t WebRtcAudioDeviceImpl::MaxMicrophoneVolume( |
744 uint32_t* max_volume) const { | 778 uint32_t* max_volume) const { |
745 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::MaxMicrophoneVolume() " | 779 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::MaxMicrophoneVolume() " |
746 << "NOT IMPLEMENTED"; | 780 << "NOT IMPLEMENTED"; |
747 return -1; | 781 return -1; |
748 } | 782 } |
749 | 783 |
750 int32_t WebRtcAudioDeviceImpl::MinMicrophoneVolume( | 784 int32_t WebRtcAudioDeviceImpl::MinMicrophoneVolume( |
751 uint32_t* min_volume) const { | 785 uint32_t* min_volume) const { |
752 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::MinMicrophoneVolume() " | 786 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::MinMicrophoneVolume() " |
753 << "NOT IMPLEMENTED"; | 787 << "NOT IMPLEMENTED"; |
754 return -1; | 788 return -1; |
755 } | 789 } |
756 | 790 |
757 int32_t WebRtcAudioDeviceImpl::MicrophoneVolumeStepSize( | 791 int32_t WebRtcAudioDeviceImpl::MicrophoneVolumeStepSize( |
758 uint16_t* step_size) const { | 792 uint16_t* step_size) const { |
759 NOTIMPLEMENTED(); | 793 NOTIMPLEMENTED(); |
760 return -1; | 794 return -1; |
761 } | 795 } |
762 | 796 |
763 int32_t WebRtcAudioDeviceImpl::SpeakerMuteIsAvailable(bool* available) { | 797 int32_t WebRtcAudioDeviceImpl::SpeakerMuteIsAvailable(bool* available) { |
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800 NOTIMPLEMENTED(); | 834 NOTIMPLEMENTED(); |
801 return -1; | 835 return -1; |
802 } | 836 } |
803 | 837 |
804 int32_t WebRtcAudioDeviceImpl::MicrophoneBoost(bool* enabled) const { | 838 int32_t WebRtcAudioDeviceImpl::MicrophoneBoost(bool* enabled) const { |
805 NOTIMPLEMENTED(); | 839 NOTIMPLEMENTED(); |
806 return -1; | 840 return -1; |
807 } | 841 } |
808 | 842 |
809 int32_t WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable(bool* available) const { | 843 int32_t WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable(bool* available) const { |
810 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable() " | 844 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable() " |
811 << "NOT IMPLEMENTED"; | 845 << "NOT IMPLEMENTED"; |
812 *available = false; | 846 *available = false; |
813 return 0; | 847 return 0; |
814 } | 848 } |
815 | 849 |
816 int32_t WebRtcAudioDeviceImpl::SetStereoPlayout(bool enable) { | 850 int32_t WebRtcAudioDeviceImpl::SetStereoPlayout(bool enable) { |
817 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetStereoPlayout() " | 851 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetStereoPlayout() " |
818 << "NOT IMPLEMENTED"; | 852 << "NOT IMPLEMENTED"; |
819 return 0; | 853 return 0; |
820 } | 854 } |
821 | 855 |
822 int32_t WebRtcAudioDeviceImpl::StereoPlayout(bool* enabled) const { | 856 int32_t WebRtcAudioDeviceImpl::StereoPlayout(bool* enabled) const { |
823 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoPlayout() " | 857 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoPlayout() " |
824 << "NOT IMPLEMENTED"; | 858 << "NOT IMPLEMENTED"; |
825 return 0; | 859 return 0; |
826 } | 860 } |
827 | 861 |
828 int32_t WebRtcAudioDeviceImpl::StereoRecordingIsAvailable( | 862 int32_t WebRtcAudioDeviceImpl::StereoRecordingIsAvailable( |
829 bool* available) const { | 863 bool* available) const { |
830 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoRecordingIsAvailable() " | 864 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoRecordingIsAvailable() " |
831 << "NOT IMPLEMENTED"; | 865 << "NOT IMPLEMENTED"; |
832 return 0; | 866 return 0; |
833 } | 867 } |
834 | 868 |
835 int32_t WebRtcAudioDeviceImpl::SetStereoRecording(bool enable) { | 869 int32_t WebRtcAudioDeviceImpl::SetStereoRecording(bool enable) { |
836 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetStereoRecording() " | 870 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::SetStereoRecording() " |
837 << "NOT IMPLEMENTED"; | 871 << "NOT IMPLEMENTED"; |
838 return -1; | 872 return -1; |
839 } | 873 } |
840 | 874 |
841 int32_t WebRtcAudioDeviceImpl::StereoRecording(bool* enabled) const { | 875 int32_t WebRtcAudioDeviceImpl::StereoRecording(bool* enabled) const { |
842 VLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoRecording() " | 876 DVLOG(2) << "WARNING: WebRtcAudioDeviceImpl::StereoRecording() " |
843 << "NOT IMPLEMENTED"; | 877 << "NOT IMPLEMENTED"; |
844 return -1; | 878 return -1; |
845 } | 879 } |
846 | 880 |
847 int32_t WebRtcAudioDeviceImpl::SetRecordingChannel(const ChannelType channel) { | 881 int32_t WebRtcAudioDeviceImpl::SetRecordingChannel(const ChannelType channel) { |
848 NOTIMPLEMENTED(); | 882 NOTIMPLEMENTED(); |
849 return -1; | 883 return -1; |
850 } | 884 } |
851 | 885 |
852 int32_t WebRtcAudioDeviceImpl::RecordingChannel(ChannelType* channel) const { | 886 int32_t WebRtcAudioDeviceImpl::RecordingChannel(ChannelType* channel) const { |
853 NOTIMPLEMENTED(); | 887 NOTIMPLEMENTED(); |
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944 } | 978 } |
945 | 979 |
946 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const { | 980 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const { |
947 NOTIMPLEMENTED(); | 981 NOTIMPLEMENTED(); |
948 return -1; | 982 return -1; |
949 } | 983 } |
950 | 984 |
951 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) { | 985 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) { |
952 session_id_ = session_id; | 986 session_id_ = session_id; |
953 } | 987 } |
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