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Issue 10991019: audio_manager_linux: Allow CRAS to decide if sample rate conversion is needed. (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: Created 8 years, 2 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/linux/audio_manager_linux.h" 5 #include "media/audio/linux/audio_manager_linux.h"
6 6
7 #include "base/command_line.h" 7 #include "base/command_line.h"
8 #include "base/environment.h" 8 #include "base/environment.h"
9 #include "base/logging.h" 9 #include "base/logging.h"
10 #include "base/nix/xdg_util.h" 10 #include "base/nix/xdg_util.h"
(...skipping 333 matching lines...) Expand 10 before | Expand all | Expand 10 after
344 } 344 }
345 345
346 AudioParameters AudioManagerLinux::GetPreferredLowLatencyOutputStreamParameters( 346 AudioParameters AudioManagerLinux::GetPreferredLowLatencyOutputStreamParameters(
347 const AudioParameters& input_params) { 347 const AudioParameters& input_params) {
348 // Since Linux doesn't actually have a low latency path the hardware buffer 348 // Since Linux doesn't actually have a low latency path the hardware buffer
349 // size is quite large in order to prevent glitches with general usage. Some 349 // size is quite large in order to prevent glitches with general usage. Some
350 // clients, such as WebRTC, have a more limited use case and work acceptably 350 // clients, such as WebRTC, have a more limited use case and work acceptably
351 // with a smaller buffer size. The check below allows clients which want to 351 // with a smaller buffer size. The check below allows clients which want to
352 // try a smaller buffer size on Linux to do so. 352 // try a smaller buffer size on Linux to do so.
353 int buffer_size = 0; 353 int buffer_size = 0;
354 if (static_cast<size_t>(input_params.frames_per_buffer()) < 354 if (static_cast<size_t>(input_params.frames_per_buffer()) <
DaleCurtis 2012/09/25 19:56:55 Mind collapsing this to: int buffer_size = GetAud
355 GetAudioHardwareBufferSize()) { 355 GetAudioHardwareBufferSize()) {
356 buffer_size = input_params.frames_per_buffer(); 356 buffer_size = input_params.frames_per_buffer();
357 } else { 357 } else {
358 buffer_size = GetAudioHardwareBufferSize(); 358 buffer_size = GetAudioHardwareBufferSize();
359 } 359 }
360 360
361 #if defined(USE_CRAS)
362 // CRAS will sample rate convert if it is needed, pass the desired frame rate.
DaleCurtis 2012/09/25 19:56:55 To keep style with the above code, why not: int s
363 if (UseCras())
364 return AudioParameters(
365 AudioParameters::AUDIO_PCM_LOW_LATENCY, input_params.channel_layout(),
366 input_params.sample_rate(), 16, buffer_size);
367 #endif
368
361 // TODO(dalecurtis): This should include bits per channel and channel layout 369 // TODO(dalecurtis): This should include bits per channel and channel layout
362 // eventually. 370 // eventually.
363 return AudioParameters( 371 return AudioParameters(
364 AudioParameters::AUDIO_PCM_LOW_LATENCY, input_params.channel_layout(), 372 AudioParameters::AUDIO_PCM_LOW_LATENCY, input_params.channel_layout(),
365 GetAudioHardwareSampleRate(), 16, buffer_size); 373 GetAudioHardwareSampleRate(), 16, buffer_size);
366 } 374 }
367 375
368 } // namespace media 376 } // namespace media
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