AudioBuffer: Eliminate the SplitChannelBuffer class

It's just a container for two IFChannelBuffers, and doesn't earn its
keep. The main problem is that the number of methods it needs that
just forward calls to either of its two IFChannelBuffers was already
large, and was about to grow.

R=aluebs@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6717 4adac7df-926f-26a2-2b94-8c16560cd09d

2 files changed, 18 insertions, 29 deletions

diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 7eac7ecf..1ceba383 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -125,24 +125,6 @@ class IFChannelBuffer {
   ChannelBuffer<float> fbuf_;
 };
 
-class SplitChannelBuffer {
- public:
-  SplitChannelBuffer(int samples_per_split_channel, int num_channels)
-      : low_(samples_per_split_channel, num_channels),
-        high_(samples_per_split_channel, num_channels) {
-  }
-  ~SplitChannelBuffer() {}
-
-  int16_t* low_channel(int i) { return low_.ibuf()->channel(i); }
-  int16_t* high_channel(int i) { return high_.ibuf()->channel(i); }
-  float* low_channel_f(int i) { return low_.fbuf()->channel(i); }
-  float* high_channel_f(int i) { return high_.fbuf()->channel(i); }
-
- private:
-  IFChannelBuffer low_;
-  IFChannelBuffer high_;
-};
-
 AudioBuffer::AudioBuffer(int input_samples_per_channel,
                          int num_input_channels,
                          int process_samples_per_channel,
@@ -198,8 +180,10 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
 
   if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) {
     samples_per_split_channel_ = kSamplesPer16kHzChannel;
-    split_channels_.reset(new SplitChannelBuffer(samples_per_split_channel_,
-                                                 num_proc_channels_));
+    split_channels_low_.reset(new IFChannelBuffer(samples_per_split_channel_,
+                                                  num_proc_channels_));
+    split_channels_high_.reset(new IFChannelBuffer(samples_per_split_channel_,
+                                                   num_proc_channels_));
     filter_states_.reset(new SplitFilterStates[num_proc_channels_]);
   }
 }
@@ -302,8 +286,9 @@ float* AudioBuffer::data_f(int channel) {
 }
 
 const int16_t* AudioBuffer::low_pass_split_data(int channel) const {
-  return split_channels_.get() ? split_channels_->low_channel(channel)
-                               : data(channel);
+  return split_channels_low_.get()
+      ? split_channels_low_->ibuf()->channel(channel)
+      : data(channel);
 }
 
 int16_t* AudioBuffer::low_pass_split_data(int channel) {
@@ -313,8 +298,9 @@ int16_t* AudioBuffer::low_pass_split_data(int channel) {
 }
 
 const float* AudioBuffer::low_pass_split_data_f(int channel) const {
-  return split_channels_.get() ? split_channels_->low_channel_f(channel)
-                               : data_f(channel);
+  return split_channels_low_.get()
+      ? split_channels_low_->fbuf()->channel(channel)
+      : data_f(channel);
 }
 
 float* AudioBuffer::low_pass_split_data_f(int channel) {
@@ -324,7 +310,9 @@ float* AudioBuffer::low_pass_split_data_f(int channel) {
 }
 
 const int16_t* AudioBuffer::high_pass_split_data(int channel) const {
-  return split_channels_.get() ? split_channels_->high_channel(channel) : NULL;
+  return split_channels_high_.get()
+      ? split_channels_high_->ibuf()->channel(channel)
+      : NULL;
 }
 
 int16_t* AudioBuffer::high_pass_split_data(int channel) {
@@ -333,8 +321,9 @@ int16_t* AudioBuffer::high_pass_split_data(int channel) {
 }
 
 const float* AudioBuffer::high_pass_split_data_f(int channel) const {
-  return split_channels_.get() ? split_channels_->high_channel_f(channel)
-                               : NULL;
+  return split_channels_high_.get()
+      ? split_channels_high_->fbuf()->channel(channel)
+      : NULL;
 }
 
 float* AudioBuffer::high_pass_split_data_f(int channel) {
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index 6b1a46f9..5c26ae29 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -23,7 +23,6 @@
 namespace webrtc {
 
 class PushSincResampler;
-class SplitChannelBuffer;
 class IFChannelBuffer;
 
 struct SplitFilterStates {
@@ -115,7 +114,8 @@ class AudioBuffer {
 
   const float* keyboard_data_;
   scoped_ptr<IFChannelBuffer> channels_;
-  scoped_ptr<SplitChannelBuffer> split_channels_;
+  scoped_ptr<IFChannelBuffer> split_channels_low_;
+  scoped_ptr<IFChannelBuffer> split_channels_high_;
   scoped_ptr<SplitFilterStates[]> filter_states_;
   scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
   scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;