Make AudioParameters a class instead of a struct

content/renderer/media/audio_renderer_impl.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 "content/renderer/media/audio_renderer_impl.h"
 
 #include <math.h>
 
 #include <algorithm>
 
@@ skipping 35 matching lines @@
       predicted_play_time = base::TimeDelta::FromMicroseconds(
           static_cast<int64>(ceil(predicted_play_time.InMicroseconds() *
                                   playback_rate)));
     }
     earliest_end_time_ =
         std::max(earliest_end_time_,
                  time_now + request_delay + predicted_play_time);
   }
 }
 
 bool AudioRendererImpl::OnInitialize(int bits_per_channel,

[vrk (LEFT CHROMIUM), 2012/03/09 20:59:32]

This should also probably be changed to use AudioParameters, but I was going to
change this when I squished AudioRendererImpl and AudioRendererBase into one
(crbug.com/116645).

[scherkus (not reviewing), 2012/03/09 21:48:59]

SGTM

                                      ChannelLayout channel_layout,
                                      int sample_rate) {
   // We use the AUDIO_PCM_LINEAR flag because AUDIO_PCM_LOW_LATENCY
   // does not currently support all the sample-rates that we require.
   // Please see: http://code.google.com/p/chromium/issues/detail?id=103627
   // for more details.
-  audio_parameters_ = AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
-                                      channel_layout,
-                                      sample_rate,
-                                      bits_per_channel,
-                                      0);
+  audio_parameters_.Reset(
+      AudioParameters::AUDIO_PCM_LINEAR,
+      channel_layout, sample_rate, bits_per_channel,
+      audio_hardware::GetHighLatencyOutputBufferSize(sample_rate));
 
   bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
 
   DCHECK(sink_.get());
 
   if (!is_initialized_) {
-    sink_->Initialize(
-        audio_hardware::GetHighLatencyOutputBufferSize(sample_rate),
-        audio_parameters_.channels,
-        audio_parameters_.sample_rate,
-        audio_parameters_.format,
-        this);
+    sink_->Initialize(audio_parameters_, this);
 
     sink_->Start();
     is_initialized_ = true;
     return true;
   }
 
   return false;
 }
 
 void AudioRendererImpl::OnStop() {
@@ skipping 96 matching lines @@
   base::TimeDelta request_delay =
       base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
 
   // Finally we need to adjust the delay according to playback rate.
   if (GetPlaybackRate() != 1.0f) {
     request_delay = base::TimeDelta::FromMicroseconds(
         static_cast<int64>(ceil(request_delay.InMicroseconds() *
                                 GetPlaybackRate())));
   }
 
   uint32 bytes_per_frame =

[scherkus (not reviewing), 2012/03/09 21:48:59]

isn't there a function on AudioParameters for this?

[vrk (LEFT CHROMIUM), 2012/03/16 18:30:41]

There isn't, but while I'm here I can add one! Mostly done; I can use this
function in a few more other places, but after the
AudioRendererImpl/AudioRendererBase merge.

-      audio_parameters_.bits_per_sample * audio_parameters_.channels / 8;
+      audio_parameters_.bits_per_sample() * audio_parameters_.channels() / 8;
 
   const size_t buf_size = number_of_frames * bytes_per_frame;
   scoped_array<uint8> buf(new uint8[buf_size]);
 
   uint32 frames_filled = FillBuffer(buf.get(), number_of_frames, request_delay);
   uint32 bytes_filled = frames_filled * bytes_per_frame;
   DCHECK_LE(bytes_filled, buf_size);
   UpdateEarliestEndTime(bytes_filled, request_delay, base::Time::Now());
 
   // Deinterleave each audio channel.
@@ skipping 19 matching lines @@
 
 void AudioRendererImpl::OnRenderError() {
   host()->DisableAudioRenderer();
 }
 
 void AudioRendererImpl::OnRenderEndOfStream() {
   // TODO(enal): schedule callback instead of polling.
   if (base::Time::Now() >= earliest_end_time_)
     SignalEndOfStream();
 }