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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 | |
3 // found in the LICENSE file. | |
4 | |
5 #include "media/audio/win/audio_low_latency_output_win.h" | |
6 | |
7 #include "base/logging.h" | |
8 #include "base/memory/scoped_ptr.h" | |
9 #include "base/utf_string_conversions.h" | |
10 #include "media/audio/audio_util.h" | |
11 #include "media/audio/win/audio_manager_win.h" | |
12 #include "media/audio/win/avrt_wrapper_win.h" | |
13 | |
14 using base::win::ScopedComPtr; | |
15 using base::win::ScopedCOMInitializer; | |
16 | |
17 WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager, | |
18 const AudioParameters& params, | |
19 ERole device_role) | |
20 : com_init_(ScopedCOMInitializer::kMTA), | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
do you want to add a check to the constructor (as
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
21 manager_(manager), | |
22 render_thread_(NULL), | |
23 opened_(false), | |
24 started_(false), | |
25 volume_(1.0), | |
26 endpoint_buffer_size_frames_(0), | |
27 device_role_(device_role), | |
28 num_written_frames_(0), | |
29 source_(NULL) { | |
30 DCHECK(manager_); | |
31 | |
32 // Load the Avrt DLL if not already loaded. Required to support MMCSS. | |
33 bool avrt_init = avrt::Initialize(); | |
34 DCHECK(avrt_init) << "Failed to load the Avrt.dll"; | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
nit: "Failed to load avrt.dll"
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
35 | |
36 // Set up the desired render format specified by the client. | |
37 format_.nSamplesPerSec = params.sample_rate; | |
38 format_.wFormatTag = WAVE_FORMAT_PCM; | |
39 format_.wBitsPerSample = params.bits_per_sample; | |
40 format_.nChannels = params.channels; | |
41 format_.nBlockAlign = (format_.wBitsPerSample / 8) * format_.nChannels; | |
42 format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign; | |
43 format_.cbSize = 0; | |
44 | |
45 // Size in bytes of each audio frame. | |
46 frame_size_ = format_.nBlockAlign; | |
47 | |
48 // Store size (in different units) of audio packets which we expect to | |
49 // get from the audio endpoint device in each render event. | |
50 packet_size_frames_ = params.GetPacketSize() / format_.nBlockAlign; | |
Raymond Toy (Google)
2011/11/03 22:49:56
Is it possible for GetPacketSize() to return somet
henrika (OOO until Aug 14)
2011/11/04 11:26:15
The result will always be equal to params.samples_
Raymond Toy (Google)
2011/11/04 16:58:02
No, this is fine.
| |
51 packet_size_bytes_ = params.GetPacketSize(); | |
52 packet_size_ms_ = (1000.0 * packet_size_frames_) / params.sample_rate; | |
53 DVLOG(1) << "Number of bytes per audio frame : " << frame_size_; | |
54 DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_; | |
55 DVLOG(1) << "Number of milliseconds per packet: " << packet_size_ms_; | |
56 | |
57 // All events are auto-reset events and non-signaled initially. | |
58 | |
59 // Create the event which the audio engine will signal each time | |
60 // a buffer becomes ready to be processed by the client. | |
61 audio_samples_render_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL)); | |
62 DCHECK(audio_samples_render_event_.IsValid()); | |
63 | |
64 // Create the event which will be set in Stop() when capturing shall stop. | |
65 stop_render_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL)); | |
66 DCHECK(stop_render_event_.IsValid()); | |
67 } | |
68 | |
69 WASAPIAudioOutputStream::~WASAPIAudioOutputStream() {} | |
70 | |
71 bool WASAPIAudioOutputStream::Open() { | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
ponder: Should we DCHECK in these methods (Open, S
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Good point. Makes sense here since we need a certa
| |
72 // Verify that we are not already opened. | |
73 if (opened_) | |
74 return false; | |
75 | |
76 // Obtain a reference to the IMMDevice interface of the default rendering | |
77 // device with the specified role. | |
78 HRESULT hr = SetRenderDevice(device_role_); | |
79 if (FAILED(hr)) { | |
80 HandleError(hr); | |
81 return false; | |
82 } | |
83 | |
84 // Obtain an IAudioClient interface which enables us to create and initialize | |
85 // an audio stream between an audio application and the audio engine. | |
86 hr = ActivateRenderDevice(); | |
87 if (FAILED(hr)) { | |
88 HandleError(hr); | |
89 return false; | |
90 } | |
91 | |
92 // Retrieve the stream format which the audio engine uses for its internal | |
93 // processing/mixing of shared-mode streams. | |
94 hr = GetAudioEngineStreamFormat(); | |
95 if (FAILED(hr)) { | |
96 HandleError(hr); | |
97 return false; | |
98 } | |
99 | |
100 // Verify that the selected audio endpoint supports the specified format | |
101 // set during construction. | |
102 if (!DesiredFormatIsSupported()) { | |
103 hr = E_INVALIDARG; | |
104 HandleError(hr); | |
105 return false; | |
106 } | |
107 | |
108 // Initialize the audio stream between the client and the device using | |
109 // shared mode and a lowest possible glitch-free latency. | |
110 hr = InitializeAudioEngine(); | |
111 if (FAILED(hr)) { | |
112 HandleError(hr); | |
113 return false; | |
114 } | |
115 | |
116 opened_ = true; | |
117 | |
118 return true; | |
119 } | |
120 | |
121 void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) { | |
122 DCHECK(callback); | |
123 DCHECK(opened_); | |
124 | |
125 if (!opened_) | |
126 return; | |
127 | |
128 if (started_) | |
129 return; | |
130 | |
131 source_ = callback; | |
132 | |
133 // Create and start the thread that will drive the rendering by waiting for | |
134 // render events. | |
135 render_thread_ = new base::DelegateSimpleThread(this, "wasapi_render_thread"); | |
136 render_thread_->Start(); | |
Niklas Enbom
2011/11/03 15:32:09
Why start the thread already here? What if followi
henrika (OOO until Aug 14)
2011/11/04 11:26:15
Good point, will modify.
| |
137 | |
138 // Avoid start-up glitches by filling up the endpoint buffer with "silence" | |
139 // before starting the stream. | |
140 BYTE* data_ptr = NULL; | |
141 HRESULT hr = audio_render_client_->GetBuffer(endpoint_buffer_size_frames_, | |
142 &data_ptr); | |
143 if (SUCCEEDED(hr)) { | |
144 // Using the AUDCLNT_BUFFERFLAGS_SILENT flag eliminates the need to | |
145 // explicitly write silence data to the rendering buffer. | |
146 audio_render_client_->ReleaseBuffer(endpoint_buffer_size_frames_, | |
147 AUDCLNT_BUFFERFLAGS_SILENT); | |
148 num_written_frames_ = endpoint_buffer_size_frames_; | |
149 | |
150 // Sanity check: verify that the endpoint buffer is filled with silence. | |
151 UINT32 num_queued_frames = 0; | |
152 audio_client_->GetCurrentPadding(&num_queued_frames); | |
153 DCHECK(num_queued_frames == num_written_frames_); | |
154 } | |
155 | |
156 // Start streaming data between the endpoint buffer and the audio engine. | |
157 hr = audio_client_->Start(); | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
If this fails, should we call HandleError()?
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Improved:
if (FAILED(hr)) {
SetEvent(stop_re
| |
158 DLOG_IF(ERROR, FAILED(hr)) << "Failed to start output streaming: " | |
159 << std::hex << hr; | |
160 | |
161 started_ = SUCCEEDED(hr); | |
162 } | |
163 | |
164 void WASAPIAudioOutputStream::Stop() { | |
165 if (!started_) | |
166 return; | |
167 | |
168 // Shut down the render thread. | |
169 if (stop_render_event_.IsValid()) { | |
170 SetEvent(stop_render_event_.Get()); | |
171 } | |
172 | |
173 // Stop output audio streaming. | |
174 HRESULT hr = audio_client_->Stop(); | |
175 DLOG_IF(ERROR, FAILED(hr)) << "Failed to stop output streaming: " | |
176 << std::hex << hr; | |
177 | |
178 // Wait until the thread completes and perform cleanup. | |
179 if (render_thread_) { | |
180 SetEvent(stop_render_event_.Get()); | |
181 render_thread_->Join(); | |
182 render_thread_ = NULL; | |
183 } | |
184 | |
185 started_ = false; | |
186 } | |
187 | |
188 void WASAPIAudioOutputStream::Close() { | |
189 // It is valid to call Close() before calling open or Start(). | |
190 // It is also valid to call Close() after Start() has been called. | |
191 Stop(); | |
192 | |
193 // Inform the audio manager that we have been closed. This will cause our | |
194 // destruction. | |
195 manager_->ReleaseOutputStream(this); | |
196 } | |
197 | |
198 void WASAPIAudioOutputStream::SetVolume(double volume) { | |
199 if (volume < 0.0f || volume > 1.0f) | |
200 return; | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
is this something we expect to happen? If this sh
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
201 volume_ = static_cast<float>(volume); | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
nit: casting from double to float isn't exact, so
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Fixed ;-)
| |
202 } | |
203 | |
204 void WASAPIAudioOutputStream::GetVolume(double* volume) { | |
205 *volume = static_cast<double>(volume_); | |
206 } | |
207 | |
208 // static | |
209 double WASAPIAudioOutputStream::HardwareSampleRate(ERole device_role) { | |
210 // It is assumed that this static method is called from a COM thread, i.e., | |
211 // CoInitializeEx() is not called here again to avoid STA/MTA conflicts. | |
212 ScopedComPtr<IMMDeviceEnumerator> enumerator; | |
213 HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), | |
214 NULL, | |
215 CLSCTX_INPROC_SERVER, | |
216 __uuidof(IMMDeviceEnumerator), | |
217 enumerator.ReceiveVoid()); | |
218 if (FAILED(hr)) { | |
219 NOTREACHED() << "error code: " << std::hex << hr; | |
220 } | |
Raymond Toy (Google)
2011/11/03 22:49:56
Should we return 0.0 here, like we do in other NOT
henrika (OOO until Aug 14)
2011/11/04 11:26:15
Fixed.
| |
221 | |
222 ScopedComPtr<IMMDevice> endpoint_device; | |
223 hr = enumerator->GetDefaultAudioEndpoint(eRender, | |
224 device_role, | |
225 endpoint_device.Receive()); | |
226 if (FAILED(hr)) { | |
227 // This will happen if there's no audio output device found or available | |
228 // (e.g. some audio cards that have outputs will still report them as | |
229 // "not found" when no speaker is plugged into the output jack). | |
230 LOG(WARNING) << "No audio end point: " << std::hex << hr; | |
231 return 0.0; | |
232 } | |
233 | |
234 ScopedComPtr<IAudioClient> audio_client; | |
235 hr = endpoint_device->Activate(__uuidof(IAudioClient), | |
236 CLSCTX_INPROC_SERVER, | |
237 NULL, | |
238 audio_client.ReceiveVoid()); | |
239 if (FAILED(hr)) { | |
240 NOTREACHED() << "error code: " << std::hex << hr; | |
241 return 0.0; | |
242 } | |
243 | |
244 base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format; | |
245 hr = audio_client->GetMixFormat(&audio_engine_mix_format); | |
246 if (FAILED(hr)) { | |
247 NOTREACHED() << "error code: " << std::hex << hr; | |
248 return 0.0; | |
249 } | |
250 | |
251 return static_cast<double>(audio_engine_mix_format->nSamplesPerSec); | |
252 } | |
253 | |
254 void WASAPIAudioOutputStream::Run() { | |
255 ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA); | |
256 | |
257 // Increase the thread priority. | |
258 render_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio); | |
259 | |
260 // Enable MMCSS to ensure that this thread receives prioritized access to | |
261 // CPU resources. | |
262 DWORD task_index = 0; | |
263 HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio", | |
264 &task_index); | |
265 bool mmcss_is_ok = | |
266 (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL)); | |
267 if (!mmcss_is_ok) { | |
268 // Failed to enable MMCSS on this thread. It is not fatal but can lead | |
269 // to reduced QoS at high load. | |
270 DWORD err = GetLastError(); | |
271 LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ")."; | |
272 } | |
273 | |
274 HRESULT hr = S_FALSE; | |
275 | |
276 bool playing = true; | |
277 bool error = false; | |
278 HANDLE wait_array[2] = {stop_render_event_, audio_samples_render_event_}; | |
279 UINT64 device_frequency = 0; | |
280 | |
281 // The IAudioClock interface enables us to monitor a stream's data | |
282 // rate and the current position in the stream. Allocate it before we | |
283 // start spinning. | |
284 ScopedComPtr<IAudioClock> audio_clock; | |
285 hr = audio_client_->GetService(__uuidof(IAudioClock), | |
286 audio_clock.ReceiveVoid()); | |
287 if (SUCCEEDED(hr)) { | |
288 // The device frequency is the frequency generated by the hardware clock in | |
289 // the audio device. The GetFrequency() method reports a constant frequency. | |
290 hr = audio_clock->GetFrequency(&device_frequency); | |
291 } | |
292 error = FAILED(hr); | |
Raymond Toy (Google)
2011/11/03 22:49:56
It's a little confusing here that hr can be the re
henrika (OOO until Aug 14)
2011/11/04 11:26:15
If GetService() fails, error is true and the threa
| |
293 PLOG_IF(ERROR, error) << "Failed to acquire IAudioClock interface: " | |
294 << std::hex << hr; | |
295 | |
296 while (playing && !error) { | |
297 // Wait for a close-down event or a new render event. | |
298 DWORD wait_result = WaitForMultipleObjects(2, wait_array, FALSE, INFINITE); | |
299 | |
300 switch (wait_result) { | |
301 case WAIT_OBJECT_0 + 0: | |
302 // |stop_render_event_| has been set. | |
303 playing = false; | |
304 break; | |
305 case WAIT_OBJECT_0 + 1: | |
306 { | |
307 // |audio_samples_render_event_| has been set. | |
308 UINT32 num_queued_frames = 0; | |
309 uint8* audio_data = NULL; | |
310 | |
311 // Get the padding value which represents the amount of rendering | |
312 // data that is queued up to play in the endpoint buffer. | |
313 hr = audio_client_->GetCurrentPadding(&num_queued_frames); | |
314 | |
315 // Determine how much new data we can write to the buffer without | |
316 // the risk of overwriting previously written data that the audio | |
317 // engine has not yet read from the buffer. | |
318 size_t num_available_frames = | |
319 endpoint_buffer_size_frames_ - num_queued_frames; | |
320 | |
321 // Check if there is enough available space to fit the packet size | |
322 // specified by the client. | |
323 if (num_available_frames < packet_size_frames_) { | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
nit: remove {}
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
324 continue; | |
325 } | |
326 | |
327 // Derive the number of packets we need get from the client to | |
328 // fill up the available area in the endpoint buffer. | |
329 size_t num_packets = (num_available_frames / packet_size_frames_); | |
330 | |
331 // Get data from the client/source. | |
332 for (size_t n = 0; n < num_packets; n++) { | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
++n
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
333 // Grab all available space in the rendering endpoint buffer | |
334 // into which the client can write a data packet. | |
335 hr = audio_render_client_->GetBuffer(packet_size_frames_, | |
336 &audio_data); | |
337 | |
338 // Derive the audio delay which corresponds to the delay between | |
339 // a render event and the time when the first audio sample in a | |
340 // packet is played out through the speaker. This delay value | |
341 // can typically be utilized by an acoustic echo-control (AEC) | |
342 // unit at the render side. | |
343 if (SUCCEEDED(hr) && audio_data) { | |
Raymond Toy (Google)
2011/11/03 22:49:56
What happens if GetBuffer doesn't succeed? Is the
henrika (OOO until Aug 14)
2011/11/04 11:26:15
It is an extremely rare event and I have not been
tommi (sloooow) - chröme
2011/11/07 11:47:03
if it does fail though, should error be set to tru
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
344 UINT64 position = 0; | |
345 int audio_delay_bytes = 0; | |
346 hr = audio_clock->GetPosition(&position, NULL); | |
347 if (SUCCEEDED(hr)) { | |
Raymond Toy (Google)
2011/11/03 22:49:56
Same thing. What happens if GetPosition doesn't s
henrika (OOO until Aug 14)
2011/11/04 11:26:15
The user will receive 0 as audio delay and can see
tommi (sloooow) - chröme
2011/11/07 11:47:03
and set error to true?
henrika (OOO until Aug 14)
2011/11/08 12:18:37
I don't want to stop rendering even if delay estim
| |
348 // Stream position of the sample that is currently playing | |
349 // through the speaker. | |
350 double pos_sample_playing_frames = format_.nSamplesPerSec * | |
351 (static_cast<double>(position) / device_frequency); | |
352 | |
353 // Stream position of the last sample written to the endpoint | |
354 // buffer. Note that, the packet we are about to receive in | |
355 // the upcoming callback is also included. | |
356 size_t pos_last_sample_written_frames = | |
357 num_written_frames_ + packet_size_frames_; | |
358 | |
359 // Derive the actual delay value which will be fed to the | |
360 // render client using the OnMoreData() callback. | |
361 audio_delay_bytes = (pos_last_sample_written_frames - | |
362 pos_sample_playing_frames) * frame_size_; | |
363 } | |
364 | |
365 // Read a data packet from the registered client source and | |
366 // deliver a delay estimate in the same callback to the client. | |
367 // A time stamp is also stored in the AudioBuffersState. This | |
368 // time stamp can be used at the client side to compensate for | |
369 // the delay between the usage of the delay value and the time | |
370 // of generation. | |
371 uint32 num_filled_bytes = source_->OnMoreData( | |
372 this, audio_data, packet_size_bytes_, | |
373 AudioBuffersState(0, audio_delay_bytes)); | |
374 | |
375 // Perform in-place, software-volume adjustments. | |
376 media::AdjustVolume(audio_data, | |
377 num_filled_bytes, | |
378 format_.nChannels, | |
379 format_.wBitsPerSample >> 3, | |
380 volume_); | |
381 | |
382 // Zero out the part of the packet which has not been filled by | |
383 // the client. | |
384 if (num_filled_bytes < packet_size_bytes_) { | |
385 memset(&audio_data[num_filled_bytes], | |
386 0, | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
move to the line above
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
387 (packet_size_bytes_ - num_filled_bytes)); | |
Niklas Enbom
2011/11/03 15:46:47
Why, since silence is better than repeat in case o
henrika (OOO until Aug 14)
2011/11/04 11:26:15
Do you mean that you would like me to add better c
henrika (OOO until Aug 14)
2011/11/07 12:37:40
Modified the comment.
| |
388 } | |
389 } | |
390 | |
391 // Release the buffer space acquired in the GetBuffer() call. | |
392 DWORD flags(0); | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
flags = 0
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
393 hr = audio_render_client_->ReleaseBuffer(packet_size_frames_, | |
394 flags); | |
395 | |
396 num_written_frames_ += packet_size_frames_; | |
397 } | |
398 } | |
399 break; | |
400 default: | |
401 error = true; | |
402 break; | |
403 } | |
404 } | |
405 | |
406 if (playing && error) { | |
407 // TODO(henrika): perhaps it worth improving the cleanup here by e.g. | |
408 // stopping the audio client, joining the thread etc.? | |
409 NOTREACHED() << "WASAPI rendering failed with error code " | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
PLOG(ERROR)?
NOTREACHED is a debug-only assert for
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Thanks. Improved this section.
| |
410 << GetLastError(); | |
411 } | |
412 | |
413 // Disable MMCSS. | |
414 if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) { | |
415 PLOG(WARNING) << "Failed to disable MMCSS"; | |
416 } | |
417 } | |
418 | |
419 void WASAPIAudioOutputStream::HandleError(HRESULT err) { | |
420 NOTREACHED() << "Error code: " << std::hex << err; | |
421 if (source_) | |
422 source_->OnError(this, static_cast<int>(err)); | |
423 } | |
424 | |
425 HRESULT WASAPIAudioOutputStream::SetRenderDevice(ERole device_role) { | |
426 ScopedComPtr<IMMDeviceEnumerator> enumerator; | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
Assert called on correct thread?
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Not needed. Called from Open(), DCHECK() is done t
| |
427 HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), | |
428 NULL, | |
429 CLSCTX_INPROC_SERVER, | |
430 __uuidof(IMMDeviceEnumerator), | |
431 enumerator.ReceiveVoid()); | |
432 if (SUCCEEDED(hr)) { | |
433 // Retrieve the default render audio endpoint for the specified role. | |
434 // Note that, in Windows Vista, the MMDevice API supports device roles | |
435 // but the system-supplied user interface programs do not. | |
436 hr = enumerator->GetDefaultAudioEndpoint(eRender, | |
437 device_role, | |
438 endpoint_device_.Receive()); | |
439 | |
440 // Verify that the audio endpoint device is active. That is, the audio | |
441 // adapter that connects to the endpoint device is present and enabled. | |
442 DWORD state = DEVICE_STATE_DISABLED; | |
443 hr = endpoint_device_->GetState(&state); | |
444 if (SUCCEEDED(hr)) { | |
445 if (!(state & DEVICE_STATE_ACTIVE)) { | |
446 DLOG(ERROR) << "Selected render device is not active."; | |
447 hr = E_ACCESSDENIED; | |
448 } | |
449 } | |
450 } | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
else log error + hr?
henrika (OOO until Aug 14)
2011/11/08 12:18:37
The called does this as a result of hr. Should be
| |
451 | |
452 return hr; | |
Raymond Toy (Google)
2011/11/03 22:49:56
hr can contain the error code from up to 3 differe
henrika (OOO until Aug 14)
2011/11/07 12:37:40
The Open() call consist of five sub functions wher
| |
453 } | |
454 | |
455 HRESULT WASAPIAudioOutputStream::ActivateRenderDevice() { | |
456 // Creates and activates an IAudioClient COM object given the selected | |
457 // render endpoint device. | |
458 HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient), | |
459 CLSCTX_INPROC_SERVER, | |
460 NULL, | |
461 audio_client_.ReceiveVoid()); | |
462 return hr; | |
463 } | |
464 | |
465 HRESULT WASAPIAudioOutputStream::GetAudioEngineStreamFormat() { | |
466 // Retrieve the stream format that the audio engine uses for its internal | |
467 // processing/mixing of shared-mode streams. | |
468 return audio_client_->GetMixFormat(&audio_engine_mix_format_); | |
469 } | |
470 | |
471 bool WASAPIAudioOutputStream::DesiredFormatIsSupported() { | |
472 // In shared mode, the audio engine always supports the mix format, | |
473 // which is stored in the |audio_engine_mix_format_| member. In addition, | |
474 // the audio engine *might* support similar formats that have the same | |
475 // sample rate and number of channels as the mix format but differ in | |
476 // the representation of audio sample values. | |
477 base::win::ScopedCoMem<WAVEFORMATEX> closest_match; | |
478 HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, | |
479 &format_, | |
480 &closest_match); | |
481 DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported " | |
482 << "but a closest match exists."; | |
483 return (hr == S_OK); | |
484 } | |
485 | |
486 HRESULT WASAPIAudioOutputStream::InitializeAudioEngine() { | |
487 // TODO(henrika): this buffer scheme is still under development. | |
488 // The exact details are yet to be determined based on tests with different | |
489 // audio clients. | |
490 int glitch_free_buffer_size_ms = static_cast<int>(packet_size_ms_ + 0.5); | |
491 if (audio_engine_mix_format_->nSamplesPerSec == 48000) { | |
492 // Initial tests have shown that we have to add 10 ms extra to | |
493 // ensure that we don't run empty for any packet size. | |
494 glitch_free_buffer_size_ms += 10; | |
495 } else if (audio_engine_mix_format_->nSamplesPerSec == 44100) { | |
496 // Initial tests have shown that we have to add 20 ms extra to | |
497 // ensure that we don't run empty for any packet size. | |
498 glitch_free_buffer_size_ms += 20; | |
499 } else { | |
500 glitch_free_buffer_size_ms += 20; | |
501 } | |
502 DVLOG(1) << "glitch_free_buffer_size_ms: " << glitch_free_buffer_size_ms; | |
503 REFERENCE_TIME requested_buffer_duration_hns = | |
504 static_cast<REFERENCE_TIME>(glitch_free_buffer_size_ms * 10000); | |
tommi (sloooow) - chröme
2011/11/07 11:47:03
indent
henrika (OOO until Aug 14)
2011/11/08 12:18:37
Done.
| |
505 | |
506 // Initialize the audio stream between the client and the device. | |
507 // We connect indirectly through the audio engine by using shared mode | |
508 // and WASAPI is initialized in an event driven mode. | |
509 // Note that this API ensures that the buffer is never smaller than the | |
510 // minimum buffer size needed to ensure glitch-free rendering. | |
511 // If we requests a buffer size that is smaller than the audio engine's | |
512 // minimum required buffer size, the method sets the buffer size to this | |
513 // minimum buffer size rather than to the buffer size requested. | |
514 HRESULT hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED, | |
515 AUDCLNT_STREAMFLAGS_EVENTCALLBACK | | |
516 AUDCLNT_STREAMFLAGS_NOPERSIST, | |
517 requested_buffer_duration_hns, | |
518 0, | |
519 &format_, | |
520 NULL); | |
521 if (FAILED(hr)) | |
522 return hr; | |
523 | |
524 // Retrieve the length of the endpoint buffer shared between the client | |
525 // and the audio engine. The buffer length the buffer length determines | |
526 // the maximum amount of rendering data that the client can write to | |
527 // the endpoint buffer during a single processing pass. | |
528 // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate. | |
529 hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_); | |
530 if (FAILED(hr)) | |
531 return hr; | |
532 DVLOG(1) << "endpoint buffer size: " << endpoint_buffer_size_frames_ | |
533 << " [frames]"; | |
534 #ifndef NDEBUG | |
535 // The period between processing passes by the audio engine is fixed for a | |
536 // particular audio endpoint device and represents the smallest processing | |
537 // quantum for the audio engine. This period plus the stream latency between | |
538 // the buffer and endpoint device represents the minimum possible latency | |
539 // that an audio application can achieve. | |
540 REFERENCE_TIME device_period_shared_mode = 0; | |
541 REFERENCE_TIME device_period_exclusive_mode = 0; | |
542 HRESULT hr_dbg = audio_client_->GetDevicePeriod( | |
543 &device_period_shared_mode, &device_period_exclusive_mode); | |
544 if (SUCCEEDED(hr_dbg)) { | |
545 DVLOG(1) << "device period: " | |
546 << static_cast<double>(device_period_shared_mode / 10000.0) | |
547 << " [ms]"; | |
548 } | |
549 | |
550 REFERENCE_TIME latency = 0; | |
551 hr_dbg = audio_client_->GetStreamLatency(&latency); | |
552 if (SUCCEEDED(hr_dbg)) { | |
553 DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0) | |
554 << " [ms]"; | |
555 } | |
556 #endif | |
557 | |
558 // Set the event handle that the audio engine will signal each time | |
559 // a buffer becomes ready to be processed by the client. | |
560 hr = audio_client_->SetEventHandle(audio_samples_render_event_.Get()); | |
561 if (FAILED(hr)) | |
562 return hr; | |
563 | |
564 // Get access to the IAudioRenderClient interface. This interface | |
565 // enables us to write output data to a rendering endpoint buffer. | |
566 // The methods in this interface manage the movement of data packets | |
567 // that contain audio-rendering data. | |
568 hr = audio_client_->GetService(__uuidof(IAudioRenderClient), | |
569 audio_render_client_.ReceiveVoid()); | |
570 return hr; | |
Raymond Toy (Google)
2011/11/03 22:49:56
As above, hr can be the result of several differen
henrika (OOO until Aug 14)
2011/11/07 12:37:40
Failure is enough. The HRESULT code will contain t
| |
571 } | |
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