Select OSX buffer size based on sample rate.

Similar to what we do on Windows.  The default buffer size is too
small for higher sample rates.

BUG=179058
TEST=audio playback sounds correct.

Review URL: https://codereview.chromium.org/12662043

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@189965 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 28 insertions, 11 deletions

diff --git a/media/audio/mac/audio_low_latency_input_mac.cc b/media/audio/mac/audio_low_latency_input_mac.cc
index 972ae39..c1d9866 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -497,17 +497,13 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
   if (!audio_data)
     return kAudioUnitErr_InvalidElement;
 
-  // See http://crbug.com/154352 for details.
-  CHECK_EQ(number_of_frames, static_cast<UInt32>(number_of_frames_));
-
   // Accumulate captured audio in FIFO until we can match the output size
   // requested by the client.
-  DCHECK_LE(fifo_->forward_bytes(), requested_size_bytes_);
   fifo_->Append(audio_data, buffer.mDataByteSize);
 
   // Deliver recorded data to the client as soon as the FIFO contains a
   // sufficient amount.
-  if (fifo_->forward_bytes() >= requested_size_bytes_) {
+  while (fifo_->forward_bytes() >= requested_size_bytes_) {
     // Read from FIFO into temporary data buffer.
     fifo_->Read(data_->GetWritableData(), requested_size_bytes_);
 
diff --git a/media/audio/mac/audio_manager_mac.cc b/media/audio/mac/audio_manager_mac.cc
index 1520be5..93dbcc9 100644
--- a/media/audio/mac/audio_manager_mac.cc
+++ b/media/audio/mac/audio_manager_mac.cc
@@ -32,6 +32,23 @@ static const int kMaxOutputStreams = 50;
 // Default buffer size in samples for low-latency input and output streams.
 static const int kDefaultLowLatencyBufferSize = 128;
 
+static int ChooseBufferSize(int output_sample_rate) {
+  int buffer_size = kDefaultLowLatencyBufferSize;
+  const int user_buffer_size = GetUserBufferSize();
+  if (user_buffer_size) {
+    buffer_size = user_buffer_size;
+  } else if (output_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 (output_sample_rate <= 96000)
+      buffer_size = 2 * kDefaultLowLatencyBufferSize;
+    else if (output_sample_rate <= 192000)
+      buffer_size = 4 * kDefaultLowLatencyBufferSize;
+  }
+
+  return buffer_size;
+}
+
 static bool HasAudioHardware(AudioObjectPropertySelector selector) {
   AudioDeviceID output_device_id = kAudioObjectUnknown;
   const AudioObjectPropertyAddress property_address = {
@@ -370,11 +387,17 @@ void AudioManagerMac::GetAudioInputDeviceNames(
 
 AudioParameters AudioManagerMac::GetInputStreamParameters(
     const std::string& device_id) {
+  // Due to the sharing of the input and output buffer sizes, we need to choose
+  // the input buffer size based on the output sample rate.  See
+  // http://crbug.com/154352.
+  const int buffer_size = ChooseBufferSize(
+      AUAudioOutputStream::HardwareSampleRate());
+
   // TODO(xians): query the native channel layout for the specific device.
   return AudioParameters(
       AudioParameters::AUDIO_PCM_LOW_LATENCY, CHANNEL_LAYOUT_STEREO,
       AUAudioInputStream::HardwareSampleRate(), 16,
-      kDefaultLowLatencyBufferSize);
+      buffer_size);
 }
 
 AudioOutputStream* AudioManagerMac::MakeLinearOutputStream(
@@ -430,10 +453,8 @@ AudioParameters AudioManagerMac::GetPreferredOutputStreamParameters(
 
   ChannelLayout channel_layout = GuessChannelLayout(hardware_channels);
 
-  int buffer_size = kDefaultLowLatencyBufferSize;
-  int user_buffer_size = GetUserBufferSize();
-  if (user_buffer_size)
-    buffer_size = user_buffer_size;
+  const int hardware_sample_rate = AUAudioOutputStream::HardwareSampleRate();
+  const int buffer_size = ChooseBufferSize(hardware_sample_rate);
 
   int input_channels = 0;
   if (input_params.IsValid()) {
@@ -452,7 +473,7 @@ AudioParameters AudioManagerMac::GetPreferredOutputStreamParameters(
       AudioParameters::AUDIO_PCM_LOW_LATENCY,
       channel_layout,
       input_channels,
-      AUAudioOutputStream::HardwareSampleRate(),
+      hardware_sample_rate,
       16,
       buffer_size);