Switch OnMoreData() to use AudioBus.

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/logging.h"
@@ skipping 35 matching lines @@
 // and notify the audio manager, which likely will destroy this object.
 
 AUAudioOutputStream::AUAudioOutputStream(
     AudioManagerMac* manager, const AudioParameters& params)
     : manager_(manager),
       source_(NULL),
       output_unit_(0),
       output_device_id_(kAudioObjectUnknown),
       volume_(1),
       hardware_latency_frames_(0),
-      stopped_(false) {
+      stopped_(false),
+      audio_bus_(AudioBus::Create(params)) {
   // We must have a manager.
   DCHECK(manager_);
   // A frame is one sample across all channels. In interleaved audio the per
   // frame fields identify the set of n |channels|. In uncompressed audio, a
   // packet is always one frame.
   format_.mSampleRate = params.sample_rate();
   format_.mFormatID = kAudioFormatLinearPCM;
   format_.mFormatFlags = kLinearPCMFormatFlagIsPacked |
                          kLinearPCMFormatFlagIsSignedInteger;
   format_.mBitsPerChannel = params.bits_per_sample();
@@ skipping 152 matching lines @@
 OSStatus AUAudioOutputStream::Render(UInt32 number_of_frames,
                                      AudioBufferList* io_data,
                                      const AudioTimeStamp* output_time_stamp) {
   // 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);
-  uint32 filled = source_->OnMoreData(
-      audio_data, buffer.mDataByteSize,
-      AudioBuffersState(0, hardware_pending_bytes));
+
+  DCHECK_EQ(number_of_frames, static_cast<UInt32>(audio_bus_->frames()));
+  int frames_filled = source_->OnMoreData(
+      audio_bus_.get(), AudioBuffersState(0, hardware_pending_bytes));
+  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.

[Chris Rogers, 2012/08/24 20:20:26]

I think we're still going to need some platform-specific final channel ordering
adjustments like the Swizzle below.  We will have some notion of cross-platform
channel ordering for 5.1, etc.  But that may be different than the OS-level
channel ordering.

[DaleCurtis, 2012/08/24 23:53:12]

I'm hoping we can have each AOS return the matrix necessary for channel remixing
to the desired layout given a ChannelLayout.  We need to discuss this and
resampling in more detail next week.

   if (format_.mChannelsPerFrame == 6) {
     if (format_.mBitsPerChannel == 8) {
       SwizzleCoreAudioLayout5_1(reinterpret_cast<uint8*>(audio_data), filled);
     } else if (format_.mBitsPerChannel == 16) {
       SwizzleCoreAudioLayout5_1(reinterpret_cast<int16*>(audio_data), filled);
     } else if (format_.mBitsPerChannel == 32) {
       SwizzleCoreAudioLayout5_1(reinterpret_cast<int32*>(audio_data), filled);
     }
   }
 
@@ skipping 103 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