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Issue 14244005: Resolves "Huge increase in audio latency on Windows following r192046" (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: rebased Created 7 years, 8 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 "media/audio/audio_io.h" 5 #include "media/audio/audio_io.h"
6 6
7 #include <windows.h> 7 #include <windows.h>
8 #include <objbase.h> // This has to be before initguid.h 8 #include <objbase.h> // This has to be before initguid.h
9 #include <initguid.h> 9 #include <initguid.h>
10 #include <mmsystem.h> 10 #include <mmsystem.h>
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360 int input_channels = 0; 360 int input_channels = 0;
361 bool use_input_params = !CoreAudioUtil::IsSupported(); 361 bool use_input_params = !CoreAudioUtil::IsSupported();
362 if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio)) { 362 if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio)) {
363 // TODO(crogers): tune these values for best possible WebAudio performance. 363 // TODO(crogers): tune these values for best possible WebAudio performance.
364 // WebRTC works well at 48kHz and a buffer size of 480 samples will be used 364 // WebRTC works well at 48kHz and a buffer size of 480 samples will be used
365 // for this case. Note that exclusive mode is experimental. 365 // for this case. Note that exclusive mode is experimental.
366 // This sample rate will be combined with a buffer size of 256 samples, 366 // This sample rate will be combined with a buffer size of 256 samples,
367 // which corresponds to an output delay of ~5.33ms. 367 // which corresponds to an output delay of ~5.33ms.
368 sample_rate = 48000; 368 sample_rate = 48000;
369 buffer_size = 256; 369 buffer_size = 256;
370 if (input_params.IsValid())
371 channel_layout = input_params.channel_layout();
370 } else if (!use_input_params) { 372 } else if (!use_input_params) {
371 // Hardware sample-rate on Windows can be configured, so we must query. 373 // Hardware sample-rate on Windows can be configured, so we must query.
372 // TODO(henrika): improve possibility to specify an audio endpoint. 374 // TODO(henrika): improve possibility to specify an audio endpoint.
373 // Use the default device (same as for Wave) for now to be compatible. 375 // Use the default device (same as for Wave) for now to be compatible.
374 int hw_sample_rate = WASAPIAudioOutputStream::HardwareSampleRate(); 376 int hw_sample_rate = WASAPIAudioOutputStream::HardwareSampleRate();
375 377
376 AudioParameters params; 378 AudioParameters params;
377 HRESULT hr = CoreAudioUtil::GetPreferredAudioParameters(eRender, eConsole, 379 HRESULT hr = CoreAudioUtil::GetPreferredAudioParameters(eRender, eConsole,
378 &params); 380 &params);
379 int hw_buffer_size = 381 int hw_buffer_size =
380 FAILED(hr) ? kFallbackBufferSize : params.frames_per_buffer(); 382 FAILED(hr) ? kFallbackBufferSize : params.frames_per_buffer();
381 channel_layout = WASAPIAudioOutputStream::HardwareChannelLayout(); 383 channel_layout = WASAPIAudioOutputStream::HardwareChannelLayout();
382 384
383 // TODO(henrika): Figure out the right thing to do here. 385 // TODO(henrika): Figure out the right thing to do here.
384 if (hw_sample_rate && hw_buffer_size) { 386 if (hw_sample_rate && hw_buffer_size) {
385 sample_rate = hw_sample_rate; 387 sample_rate = hw_sample_rate;
386 buffer_size = hw_buffer_size; 388 buffer_size = hw_buffer_size;
387 } else { 389 } else {
388 use_input_params = true; 390 use_input_params = true;
389 } 391 }
390 } 392 }
391 393
392 if (input_params.IsValid()) { 394 if (input_params.IsValid()) {
393 if (CoreAudioUtil::IsSupported() && 395 if (CoreAudioUtil::IsSupported()) {
394 CoreAudioUtil::IsChannelLayoutSupported(eRender, eConsole, 396 // Check if it is possible to open up at the specified input channel
395 input_params.channel_layout())) { 397 // layout but avoid checking if the specified layout is the same as the
396 // Open up using the same channel layout as the source if it is 398 // hardware (preferred) layout. We do this extra check to avoid the
397 // supported by the hardware. 399 // CoreAudioUtil::IsChannelLayoutSupported() overhead in most cases.
398 channel_layout = input_params.channel_layout(); 400 if (input_params.channel_layout() != channel_layout) {
399 VLOG(1) << "Hardware channel layout is not used; using same " 401 if (CoreAudioUtil::IsChannelLayoutSupported(
400 << "layout as the source instead (" << channel_layout << ")"; 402 eRender, eConsole, input_params.channel_layout())) {
403 // Open up using the same channel layout as the source if it is
404 // supported by the hardware.
405 channel_layout = input_params.channel_layout();
406 VLOG(1) << "Hardware channel layout is not used; using same layout"
407 << " as the source instead (" << channel_layout << ")";
408 }
409 }
401 } 410 }
402 input_channels = input_params.input_channels(); 411 input_channels = input_params.input_channels();
403 if (use_input_params) { 412 if (use_input_params) {
404 // If WASAPI isn't supported we'll fallback to WaveOut, which will take 413 // If WASAPI isn't supported we'll fallback to WaveOut, which will take
405 // care of resampling and bits per sample changes. By setting these 414 // care of resampling and bits per sample changes. By setting these
406 // equal to the input values, AudioOutputResampler will skip resampling 415 // equal to the input values, AudioOutputResampler will skip resampling
407 // and bit per sample differences (since the input parameters will match 416 // and bit per sample differences (since the input parameters will match
408 // the output parameters). 417 // the output parameters).
409 sample_rate = input_params.sample_rate(); 418 sample_rate = input_params.sample_rate();
410 bits_per_sample = input_params.bits_per_sample(); 419 bits_per_sample = input_params.bits_per_sample();
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439 return new PCMWaveInAudioInputStream(this, params, kNumInputBuffers, 448 return new PCMWaveInAudioInputStream(this, params, kNumInputBuffers,
440 xp_device_id); 449 xp_device_id);
441 } 450 }
442 451
443 /// static 452 /// static
444 AudioManager* CreateAudioManager() { 453 AudioManager* CreateAudioManager() {
445 return new AudioManagerWin(); 454 return new AudioManagerWin();
446 } 455 }
447 456
448 } // namespace media 457 } // namespace media
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