Flip AudioOutputResampler to on by default.

media/audio/mac/audio_low_latency_output_mac.cc

 // Copyright (c) 2012 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/audio/mac/audio_low_latency_output_mac.h"
 
 #include <CoreServices/CoreServices.h>
 
 #include "base/basictypes.h"
+#include "base/command_line.h"
 #include "base/logging.h"
 #include "base/mac/mac_logging.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/mac/audio_manager_mac.h"
+#include "media/base/media_switches.h"
 
 namespace media {
 
 // Reorder PCM from AAC layout to Core Audio 5.1 layout.
 // TODO(fbarchard): Switch layout when ffmpeg is updated.
 template<class Format>
 static void SwizzleCoreAudioLayout5_1(Format* b, uint32 filled) {
   static const int kNumSurroundChannels = 6;
   Format aac[kNumSurroundChannels];
   for (uint32 i = 0; i < filled; i += sizeof(aac), b += kNumSurroundChannels) {
@@ skipping 189 matching lines @@
     return;
   *volume = volume_;
 }
 
 // Pulls on our provider to get rendered audio stream.
 // Note to future hackers of this function: Do not add locks here because this
 // is running on a real-time thread (for low-latency).
 OSStatus AUAudioOutputStream::Render(UInt32 number_of_frames,
                                      AudioBufferList* io_data,
                                      const AudioTimeStamp* output_time_stamp) {
+  static const bool kDisableAudioOutputResampler =
+      CommandLine::ForCurrentProcess()->HasSwitch(
+          switches::kDisableAudioOutputResampler);
+
   // Update the playout latency.
   double playout_latency_frames = GetPlayoutLatency(output_time_stamp);
 
   AudioBuffer& buffer = io_data->mBuffers[0];
   uint8* audio_data = reinterpret_cast<uint8*>(buffer.mData);
   uint32 hardware_pending_bytes = static_cast<uint32>
       ((playout_latency_frames + 0.5) * format_.mBytesPerFrame);
 
-  DCHECK_EQ(number_of_frames, static_cast<UInt32>(audio_bus_->frames()));
-  int frames_filled = source_->OnMoreData(
-      audio_bus_.get(), AudioBuffersState(0, hardware_pending_bytes));
+  // If we specify a buffer size which is too low, the OS will ask for more data
+  // to fulfill the hardware request, so resize the AudioBus as appropriate.
+  // This change requires AudioOutputResampler to prevent buffer size mismatches
+  // downstream, so glitch if it's not enabled.
+  if (!kDisableAudioOutputResampler &&
+      static_cast<UInt32>(audio_bus_->frames()) != number_of_frames) {
+    audio_bus_ = AudioBus::Create(audio_bus_->channels(), number_of_frames);
+  }

[Chris Rogers, 2012/09/14 17:28:45]

I'm not crazy about this change long-term, and I'd like to discuss alternatives.
 Can you please add a TODO to revisit for potential cleanup?

+
+  int frames_filled = std::min(source_->OnMoreData(
+      audio_bus_.get(), AudioBuffersState(0, hardware_pending_bytes)),
+      static_cast<int>(number_of_frames));
   // Note: If this ever changes to output raw float the data must be clipped and
   // sanitized since it may come from an untrusted source such as NaCl.
   audio_bus_->ToInterleaved(
       frames_filled, format_.mBitsPerChannel / 8, audio_data);
   uint32 filled = frames_filled * format_.mBytesPerFrame;
 
   // Handle channel order for 5.1 audio.
   // TODO(dalecurtis): Channel downmixing, upmixing, should be done in mixer;
   // volume adjust should use SSE optimized vector_fmul() prior to interleave.
   if (format_.mChannelsPerFrame == 6) {
@@ skipping 112 matching lines @@
   UInt64 output_time_ns = AudioConvertHostTimeToNanos(
       output_time_stamp->mHostTime);
   UInt64 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
   double delay_frames = static_cast<double>
       (1e-9 * (output_time_ns - now_ns) * format_.mSampleRate);
 
   return (delay_frames + hardware_latency_frames_);
 }
 
 }  // namespace media