Switch OnMoreData() to use AudioBus.

media/audio/win/waveout_output_win.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/win/waveout_output_win.h"
 
 #include <windows.h>
 #include <mmsystem.h>
 #pragma comment(lib, "winmm.lib")
 
@@ skipping 72 matching lines @@
     : state_(PCMA_BRAND_NEW),
       manager_(manager),
       device_id_(device_id),
       waveout_(NULL),
       callback_(NULL),
       num_buffers_(num_buffers),
       buffer_size_(params.GetBytesPerBuffer()),
       volume_(1),
       channels_(params.channels()),
       pending_bytes_(0),
-      waiting_handle_(NULL) {
+      waiting_handle_(NULL),
+      audio_bus_(AudioBus::Create(params)) {
   format_.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
   format_.Format.nChannels = params.channels();
   format_.Format.nSamplesPerSec = params.sample_rate();
   format_.Format.wBitsPerSample = params.bits_per_sample();
   format_.Format.cbSize = sizeof(format_) - sizeof(WAVEFORMATEX);
   // The next are computed from above.
   format_.Format.nBlockAlign = (format_.Format.nChannels *
                                 format_.Format.wBitsPerSample) / 8;
   format_.Format.nAvgBytesPerSec = format_.Format.nBlockAlign *
                                    format_.Format.nSamplesPerSec;
@@ skipping 230 matching lines @@
 
 void PCMWaveOutAudioOutputStream::QueueNextPacket(WAVEHDR *buffer) {
   // Call the source which will fill our buffer with pleasant sounds and
   // return to us how many bytes were used.
   // If we are down sampling to a smaller number of channels, we need to
   // scale up the amount of pending bytes.
   // TODO(fbarchard): Handle used 0 by queueing more.
   uint32 scaled_pending_bytes = pending_bytes_ * channels_ /
                                 format_.Format.nChannels;
   // TODO(sergeyu): Specify correct hardware delay for AudioBuffersState.
-  uint32 used = callback_->OnMoreData(
-      reinterpret_cast<uint8*>(buffer->lpData), buffer_size_,
-      AudioBuffersState(scaled_pending_bytes, 0));
+  int frames_filled = callback_->OnMoreData(
+      audio_bus_.get(), AudioBuffersState(scaled_pending_bytes, 0));
+  uint32 used = frames_filled * audio_bus_->channels() *
+      format_.Format.wBitsPerSample / 8;
+
   if (used <= buffer_size_) {
+    // 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_.Format.wBitsPerSample / 8, buffer->lpData);
+
     buffer->dwBufferLength = used * format_.Format.nChannels / channels_;
     if (channels_ > 2 && format_.Format.nChannels == 2) {
       media::FoldChannels(buffer->lpData, used,
                           channels_, format_.Format.wBitsPerSample >> 3,
                           volume_);
     } else {
       media::AdjustVolume(buffer->lpData, used,
                           format_.Format.nChannels,
                           format_.Format.wBitsPerSample >> 3,
                           volume_);
@@ skipping 46 matching lines @@
                                        buffer,
                                        sizeof(WAVEHDR));
       if (result != MMSYSERR_NOERROR)
         stream->HandleError(result);
       stream->pending_bytes_ += buffer->dwBufferLength;
     }
   }
 }
 
 }  // namespace media