| Index: media/base/audio_latency.cc
|
| diff --git a/media/base/audio_latency.cc b/media/base/audio_latency.cc
|
| index 61b3724efa48cc36fc8ff7c3a545c95d7ba5bb56..477b4bfeadcf842cbbb6504a269a4b035ad8c9f6 100644
|
| --- a/media/base/audio_latency.cc
|
| +++ b/media/base/audio_latency.cc
|
| @@ -9,6 +9,7 @@
|
| #include <algorithm>
|
|
|
| #include "base/logging.h"
|
| +#include "base/sys_info.h"
|
| #include "base/time/time.h"
|
| #include "build/build_config.h"
|
|
|
| @@ -142,4 +143,75 @@ int AudioLatency::GetExactBufferSize(base::TimeDelta duration,
|
| return std::max(buffer_size, hardware_buffer_size);
|
| }
|
|
|
| +#if defined(OS_MACOSX)
|
| +namespace {
|
| +// Define bounds for for low-latency input and output streams.
|
| +static const int kMinimumInputOutputBufferSize = 128;
|
| +static const int kMaximumInputOutputBufferSize = 4096;
|
| +
|
| +static int GetDefaultBufferSize(bool is_input, int sample_rate) {
|
| + // kMinimumInputOutputBufferSize is too small for the output side because
|
| + // CoreAudio can get into under-run if the renderer fails delivering data
|
| + // to the browser within the allowed time by the OS. The workaround is to
|
| + // use 256 samples as the default output buffer size for sample rates
|
| + // smaller than 96KHz.
|
| + // TODO(xians): Remove this workaround after WebAudio supports user defined
|
| + // buffer size. See https://github.com/WebAudio/web-audio-api/issues/348
|
| + // for details.
|
| + int buffer_size = is_input ? kMinimumInputOutputBufferSize
|
| + : 2 * kMinimumInputOutputBufferSize;
|
| + if (sample_rate > 48000) {
|
| + // The default buffer size is too small for higher sample rates and may lead
|
| + // to glitching. Adjust upwards by multiples of the default size.
|
| + if (sample_rate <= 96000)
|
| + buffer_size = 2 * kMinimumInputOutputBufferSize;
|
| + else if (sample_rate <= 192000)
|
| + buffer_size = 4 * kMinimumInputOutputBufferSize;
|
| + }
|
| + return buffer_size;
|
| +}
|
| +} // namespace
|
| +
|
| +int AudioLatency::GetMinimumAudioBufferSize(int sample_rate) {
|
| + return GetDefaultBufferSize(false, sample_rate);
|
| +}
|
| +
|
| +int AudioLatency::GetMaximumAudioBufferSize(int sample_rate) {
|
| + return kMaximumInputOutputBufferSize;
|
| +}
|
| +
|
| +int AudioLatency::GetDefaultAudioBufferSize(bool is_input, int sample_rate) {
|
| + return GetDefaultBufferSize(is_input, sample_rate);
|
| +}
|
| +#elif defined(USE_CRAS)
|
| +namespace {
|
| +// Define bounds for the output buffer size.
|
| +const int kMinimumOutputBufferSize = 512;
|
| +const int kMaximumOutputBufferSize = 8192;
|
| +
|
| +static int GetMinimumOutputBufferSizePerBoard() {
|
| + // On faster boards we can use smaller buffer size for lower latency.
|
| + // On slower boards we should use larger buffer size to prevent underrun.
|
| + std::string board = base::SysInfo::GetLsbReleaseBoard();
|
| + if (board == "kevin")
|
| + return 768;
|
| + else if (board == "samus")
|
| + return 256;
|
| + return kMinimumOutputBufferSize;
|
| +}
|
| +} // namespace
|
| +
|
| +int AudioLatency::GetMinimumAudioBufferSize(int sample_rate) {
|
| + return GetMinimumOutputBufferSizePerBoard();
|
| +}
|
| +
|
| +int AudioLatency::GetMaximumAudioBufferSize(int sample_rate) {
|
| + return kMaximumOutputBufferSize;
|
| +}
|
| +
|
| +int AudioLatency::GetDefaultAudioBufferSize(bool is_input, int sample_rate) {
|
| + return GetMinimumOutputBufferSizePerBoard();
|
| +}
|
| +#endif
|
| +
|
| } // namespace media
|
|
|