Mixing audio with different latency requirements

media/base/audio_hardware_config.cc

 // Copyright (c) 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/base/audio_hardware_config.h"
 
 #include <stdint.h>
 
 #include <algorithm>
 #include <cmath>
 
 #include "base/logging.h"
 #include "build/build_config.h"
 
 using base::AutoLock;
 using media::AudioParameters;
 
 namespace media {
 
-#if !defined(OS_WIN)
-// Taken from "Bit Twiddling Hacks"
-// http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2
-static uint32_t RoundUpToPowerOfTwo(uint32_t v) {
-  v--;
-  v |= v >> 1;
-  v |= v >> 2;
-  v |= v >> 4;
-  v |= v >> 8;
-  v |= v >> 16;
-  v++;
-  return v;
-}
-#endif
-
 AudioHardwareConfig::AudioHardwareConfig(
     const AudioParameters& input_params,
     const AudioParameters& output_params)
     : input_params_(input_params),
       output_params_(output_params) {}
 
 AudioHardwareConfig::~AudioHardwareConfig() {}
 
 int AudioHardwareConfig::GetOutputBufferSize() const {
   AutoLock auto_lock(config_lock_);
@@ skipping 47 matching lines @@
   AutoLock auto_lock(config_lock_);
   input_params_ = input_params;
 }
 
 void AudioHardwareConfig::UpdateOutputConfig(
     const AudioParameters& output_params) {
   AutoLock auto_lock(config_lock_);
   output_params_ = output_params;
 }
 
-// static
-int AudioHardwareConfig::GetHighLatencyBufferSize(int sample_rate,
-                                                  int buffer_size) {
-  // Empirically, we consider 20ms of samples to be high latency.
-  const double twenty_ms_size = 2.0 * sample_rate / 100;
-
-#if defined(OS_WIN)
-  buffer_size = std::max(buffer_size, 1);
-
-  // Windows doesn't use power of two buffer sizes, so we should always round up
-  // to the nearest multiple of the output buffer size.
-  const int high_latency_buffer_size =
-      std::ceil(twenty_ms_size / buffer_size) * buffer_size;
-#else
-  // On other platforms use the nearest higher power of two buffer size.  For a
-  // given sample rate, this works out to:
-  //
-  //     <= 3200   : 64
-  //     <= 6400   : 128
-  //     <= 12800  : 256
-  //     <= 25600  : 512
-  //     <= 51200  : 1024
-  //     <= 102400 : 2048
-  //     <= 204800 : 4096
-  //
-  // On Linux, the minimum hardware buffer size is 512, so the lower calculated
-  // values are unused.  OSX may have a value as low as 128.
-  const int high_latency_buffer_size = RoundUpToPowerOfTwo(twenty_ms_size);
-#endif  // defined(OS_WIN)
-
-  return std::max(buffer_size, high_latency_buffer_size);
-}
-
-int AudioHardwareConfig::GetHighLatencyBufferSize() const {
-  AutoLock auto_lock(config_lock_);
-  return GetHighLatencyBufferSize(output_params_.sample_rate(),
-                                  output_params_.frames_per_buffer());
-}
-
 }  // namespace media