Cleanup some audio classes with C++11 shine!

Use newly allowed features to cleanup some audio code:
- auto for iterator based for loop.
- nullptr for NULL.

Additionally const& cleanup found while cleaning.

BUG=none
TEST=unittests pass.

Review URL: https://codereview.chromium.org/602193003

Cr-Commit-Position: refs/heads/master@{#297048}

5 files changed, 12 insertions, 20 deletions

diff --git a/media/audio/audio_output_resampler.cc b/media/audio/audio_output_resampler.cc
index cf966e0..5926996 100644
--- a/media/audio/audio_output_resampler.cc
+++ b/media/audio/audio_output_resampler.cc
@@ -40,7 +40,7 @@ class OnMoreDataConverter
   // Clears |source_callback_| and flushes the resampler.
   void Stop();
 
-  bool started() { return source_callback_ != NULL; }
+  bool started() { return source_callback_ != nullptr; }
 
  private:
   // AudioConverter::InputCallback implementation.
@@ -237,7 +237,7 @@ bool AudioOutputResampler::StartStream(
     AudioOutputProxy* stream_proxy) {
   DCHECK(task_runner_->BelongsToCurrentThread());
 
-  OnMoreDataConverter* resampler_callback = NULL;
+  OnMoreDataConverter* resampler_callback = nullptr;
   CallbackMap::iterator it = callbacks_.find(stream_proxy);
   if (it == callbacks_.end()) {
     resampler_callback = new OnMoreDataConverter(params_, output_params_);
@@ -299,7 +299,7 @@ OnMoreDataConverter::OnMoreDataConverter(const AudioParameters& input_params,
                                          const AudioParameters& output_params)
     : io_ratio_(static_cast<double>(input_params.GetBytesPerSecond()) /
                 output_params.GetBytesPerSecond()),
-      source_callback_(NULL),
+      source_callback_(nullptr),
       input_bytes_per_second_(input_params.GetBytesPerSecond()),
       audio_converter_(input_params, output_params, false) {}
 
@@ -322,7 +322,7 @@ void OnMoreDataConverter::Start(
 
 void OnMoreDataConverter::Stop() {
   CHECK(source_callback_);
-  source_callback_ = NULL;
+  source_callback_ = nullptr;
   audio_converter_.RemoveInput(this);
 }
 
diff --git a/media/base/audio_converter.cc b/media/base/audio_converter.cc
index aa0be4f..1941119 100644
--- a/media/base/audio_converter.cc
+++ b/media/base/audio_converter.cc
@@ -197,14 +197,11 @@ void AudioConverter::SourceCallback(int fifo_frame_delay, AudioBus* dest) {
       transform_inputs_.size() == 1 ? temp_dest : mixer_input_audio_bus_.get();
 
   // Have each mixer render its data into an output buffer then mix the result.
-  for (InputCallbackSet::iterator it = transform_inputs_.begin();
-       it != transform_inputs_.end(); ++it) {
-    InputCallback* input = *it;
-
+  for (auto* input : transform_inputs_) {
     const float volume = input->ProvideInput(provide_input_dest, buffer_delay);
 
     // Optimize the most common single input, full volume case.
-    if (it == transform_inputs_.begin()) {
+    if (input == transform_inputs_.front()) {
       if (volume == 1.0f) {
         if (temp_dest != provide_input_dest)
           provide_input_dest->CopyTo(temp_dest);
diff --git a/media/base/audio_renderer_mixer.cc b/media/base/audio_renderer_mixer.cc
index 2695651..112d7ef 100644
--- a/media/base/audio_renderer_mixer.cc
+++ b/media/base/audio_renderer_mixer.cc
@@ -94,11 +94,8 @@ int AudioRendererMixer::Render(AudioBus* audio_bus,
 void AudioRendererMixer::OnRenderError() {
   // Call each mixer input and signal an error.
   base::AutoLock auto_lock(lock_);
-  for (ErrorCallbackList::const_iterator it = error_callbacks_.begin();
-       it != error_callbacks_.end();
-       ++it) {
-    it->Run();
-  }
+  for (const auto& cb : error_callbacks_)
+    cb.Run();
 }
 
 }  // namespace media
diff --git a/media/base/audio_splicer.cc b/media/base/audio_splicer.cc
index c8e0d4d..9424b0b 100644
--- a/media/base/audio_splicer.cc
+++ b/media/base/audio_splicer.cc
@@ -214,10 +214,8 @@ void AudioStreamSanitizer::AddOutputBuffer(
 
 int AudioStreamSanitizer::GetFrameCount() const {
   int frame_count = 0;
-  for (BufferQueue::const_iterator it = output_buffers_.begin();
-       it != output_buffers_.end(); ++it) {
-    frame_count += (*it)->frame_count();
-  }
+  for (const auto& buffer : output_buffers_)
+    frame_count += buffer->frame_count();
   return frame_count;
 }
 
@@ -451,7 +449,7 @@ scoped_ptr<AudioBus> AudioSplicer::ExtractCrossfadeFromPreSplice(
 
 void AudioSplicer::CrossfadePostSplice(
     scoped_ptr<AudioBus> pre_splice_bus,
-    scoped_refptr<AudioBuffer> crossfade_buffer) {
+    const scoped_refptr<AudioBuffer>& crossfade_buffer) {
   // Use the calculated timestamp and duration to ensure there's no extra gaps
   // or overlaps to process when adding the buffer to |output_sanitizer_|.
   const AudioTimestampHelper& output_ts_helper =
diff --git a/media/base/audio_splicer.h b/media/base/audio_splicer.h
index 0db5d08..e32a9fd 100644
--- a/media/base/audio_splicer.h
+++ b/media/base/audio_splicer.h
@@ -85,7 +85,7 @@ class MEDIA_EXPORT AudioSplicer {
   //
   // |post_splice_sanitizer_| will be empty after this operation.
   void CrossfadePostSplice(scoped_ptr<AudioBus> pre_splice_bus,
-                           scoped_refptr<AudioBuffer> crossfade_buffer);
+                           const scoped_refptr<AudioBuffer>& crossfade_buffer);
 
   // Reset the splice and splice end timestamps.
   void reset_splice_timestamps() {