Switch OnMoreData() to use AudioBus.

media/audio/win/audio_low_latency_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/audio_low_latency_output_win.h"
 
 #include <Functiondiscoverykeys_devpkey.h>
 
 #include "base/command_line.h"
 #include "base/logging.h"
@@ skipping 320 matching lines @@
       manager_(manager),
       opened_(false),
       started_(false),
       restart_rendering_mode_(false),
       volume_(1.0),
       endpoint_buffer_size_frames_(0),
       device_role_(device_role),
       share_mode_(GetShareMode()),
       client_channel_count_(params.channels()),
       num_written_frames_(0),
-      source_(NULL) {
+      source_(NULL),
+      audio_bus_(AudioBus::Create(params)) {
   CHECK(com_init_.succeeded());
   DCHECK(manager_);
 
   // Load the Avrt DLL if not already loaded. Required to support MMCSS.
   bool avrt_init = avrt::Initialize();
   DCHECK(avrt_init) << "Failed to load the avrt.dll";
 
   if (share_mode_ == AUDCLNT_SHAREMODE_EXCLUSIVE) {
     VLOG(1) << ">> Note that EXCLUSIVE MODE is enabled <<";
   }
@@ skipping 462 matching lines @@
             // A time stamp is also stored in the AudioBuffersState. This
             // time stamp can be used at the client side to compensate for
             // the delay between the usage of the delay value and the time
             // of generation.
 
             uint32 num_filled_bytes = 0;
             const int bytes_per_sample = format_.Format.wBitsPerSample >> 3;
 
             if (channel_factor() == 1) {
               // Case I: no up-mixing.
-              num_filled_bytes = source_->OnMoreData(
-                  audio_data, packet_size_bytes_,
-                  AudioBuffersState(0, audio_delay_bytes));
+              int frames_filled = source_->OnMoreData(
+                  audio_bus_.get(), AudioBuffersState(0, audio_delay_bytes));
+              num_filled_bytes = frames_filled * frame_size_;
+              DCHECK_LE(num_filled_bytes, packet_size_bytes_);
+              audio_bus_->ToInterleaved(

[henrika (OOO until Aug 14), 2012/08/23 14:42:53]

Have you verified that WASAPIAudioOutputStreamTest::ReadFromStereoFile works
(run with --gtest_also_run_disabled_tests)? I guess you must go from interleaved
to planar (non-interleaved) in the test as well to match now; right?

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

It seems to work as is without modification. Is that what you expected? I'm a
little unclear on your comment.

+                  frames_filled, bytes_per_sample, audio_data);
             } else {
               // Case II: up-mixing.
               const int audio_source_size_bytes =
                   packet_size_bytes_ / channel_factor();
               scoped_array<uint8> buffer;
               buffer.reset(new uint8[audio_source_size_bytes]);
 
-              num_filled_bytes = source_->OnMoreData(
-                  buffer.get(), audio_source_size_bytes,
-                  AudioBuffersState(0, audio_delay_bytes));
+              int frames_filled = source_->OnMoreData(
+                  audio_bus_.get(), AudioBuffersState(0, audio_delay_bytes));
+              num_filled_bytes =
+                  frames_filled * bytes_per_sample * audio_bus_->channels();
+              DCHECK_LE(num_filled_bytes,
+                        static_cast<size_t>(audio_source_size_bytes));
+              audio_bus_->ToInterleaved(
+                  frames_filled, bytes_per_sample, buffer.get());
 
               // Do channel up-mixing on 16-bit PCM samples.
               num_filled_bytes = ChannelUpMix(buffer.get(),
                                               &audio_data[0],
                                               client_channel_count_,
                                               endpoint_channel_count(),
                                               num_filled_bytes,
                                               bytes_per_sample);
             }
 
@@ skipping 471 matching lines @@
   // are now re-initiated and it is now possible to re-start audio rendering.
 
   // Start rendering again using the new default audio endpoint.
   hr = audio_client_->Start();
 
   restart_rendering_mode_ = false;
   return SUCCEEDED(hr);
 }
 
 }  // namespace media