Simplify AudioBuffer::mixed_low_pass_data API

R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/audio_buffer.cc

@@ -51,7 +51,6 @@ int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) {
   return -1;
 }
 
-
 void StereoToMono(const float* left, const float* right, float* out,
                   int samples_per_channel) {
   for (int i = 0; i < samples_per_channel; ++i) {
@@ -155,8 +154,7 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
     num_proc_channels_(num_process_channels),
     output_samples_per_channel_(output_samples_per_channel),
     samples_per_split_channel_(proc_samples_per_channel_),
-    num_mixed_channels_(0),
+    mixed_low_pass_valid_(false),
-    num_mixed_low_pass_channels_(0),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
     keyboard_data_(NULL),
@@ -278,8 +276,7 @@ void AudioBuffer::CopyTo(int samples_per_channel,
 
 void AudioBuffer::InitForNewData() {
   keyboard_data_ = NULL;
-  num_mixed_channels_ = 0;
+  mixed_low_pass_valid_ = false;
-  num_mixed_low_pass_channels_ = 0;
   reference_copied_ = false;
   activity_ = AudioFrame::kVadUnknown;
 }
@@ -289,6 +286,7 @@ const int16_t* AudioBuffer::data(int channel) const {
 }
 
 int16_t* AudioBuffer::data(int channel) {
+  mixed_low_pass_valid_ = false;
   const AudioBuffer* t = this;
   return const_cast<int16_t*>(t->data(channel));
 }
@@ -298,6 +296,7 @@ const float* AudioBuffer::data_f(int channel) const {
 }
 
 float* AudioBuffer::data_f(int channel) {
+  mixed_low_pass_valid_ = false;
   const AudioBuffer* t = this;
   return const_cast<float*>(t->data_f(channel));
 }
@@ -308,6 +307,7 @@ const int16_t* AudioBuffer::low_pass_split_data(int channel) const {
 }
 
 int16_t* AudioBuffer::low_pass_split_data(int channel) {
+  mixed_low_pass_valid_ = false;
   const AudioBuffer* t = this;
   return const_cast<int16_t*>(t->low_pass_split_data(channel));
 }
@@ -318,6 +318,7 @@ const float* AudioBuffer::low_pass_split_data_f(int channel) const {
 }
 
 float* AudioBuffer::low_pass_split_data_f(int channel) {
+  mixed_low_pass_valid_ = false;
   const AudioBuffer* t = this;
   return const_cast<float*>(t->low_pass_split_data_f(channel));
 }
@@ -341,12 +342,26 @@ float* AudioBuffer::high_pass_split_data_f(int channel) {
   return const_cast<float*>(t->high_pass_split_data_f(channel));
 }
 
-const int16_t* AudioBuffer::mixed_data(int channel) const {
+const int16_t* AudioBuffer::mixed_low_pass_data() {
-  return mixed_channels_->channel(channel);
+  // Currently only mixing stereo to mono is supported.
+  assert(num_proc_channels_ == 1 || num_proc_channels_ == 2);
+
+  if (num_proc_channels_ == 1) {
+    return low_pass_split_data(0);
   }
 
-const int16_t* AudioBuffer::mixed_low_pass_data(int channel) const {
+  if (!mixed_low_pass_valid_) {
-  return mixed_low_pass_channels_->channel(channel);
+    if (!mixed_low_pass_channels_.get()) {
+      mixed_low_pass_channels_.reset(
+          new ChannelBuffer<int16_t>(samples_per_split_channel_, 1));
+    }
+    StereoToMono(low_pass_split_data(0),
+                 low_pass_split_data(1),
+                 mixed_low_pass_channels_->data(),
+                 samples_per_split_channel_);
+    mixed_low_pass_valid_ = true;
+  }
+  return mixed_low_pass_channels_->data();
 }
 
 const int16_t* AudioBuffer::low_pass_reference(int channel) const {
@@ -433,42 +448,6 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
   }
 }
 
-void AudioBuffer::CopyAndMix(int num_mixed_channels) {
-  // We currently only support the stereo to mono case.
-  assert(num_proc_channels_ == 2);
-  assert(num_mixed_channels == 1);
-  if (!mixed_channels_.get()) {
-    mixed_channels_.reset(
-        new ChannelBuffer<int16_t>(proc_samples_per_channel_,
-                                   num_mixed_channels));
-  }
-
-  StereoToMono(channels_->ibuf()->channel(0),
-               channels_->ibuf()->channel(1),
-               mixed_channels_->channel(0),
-               proc_samples_per_channel_);
-
-  num_mixed_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) {
-  // We currently only support the stereo to mono case.
-  assert(num_proc_channels_ == 2);
-  assert(num_mixed_channels == 1);
-  if (!mixed_low_pass_channels_.get()) {
-    mixed_low_pass_channels_.reset(
-        new ChannelBuffer<int16_t>(samples_per_split_channel_,
-                                   num_mixed_channels));
-  }
-
-  StereoToMono(low_pass_split_data(0),
-               low_pass_split_data(1),
-               mixed_low_pass_channels_->channel(0),
-               samples_per_split_channel_);
-
-  num_mixed_low_pass_channels_ = num_mixed_channels;
-}
-
 void AudioBuffer::CopyLowPassToReference() {
   reference_copied_ = true;
   if (!low_pass_reference_channels_.get()) {

webrtc/modules/audio_processing/audio_buffer.h

@@ -63,8 +63,9 @@ class AudioBuffer {
   const int16_t* low_pass_split_data(int channel) const;
   int16_t* high_pass_split_data(int channel);
   const int16_t* high_pass_split_data(int channel) const;
-  const int16_t* mixed_data(int channel) const;
+  // Returns a pointer to the low-pass data downmixed to mono. If this data
-  const int16_t* mixed_low_pass_data(int channel) const;
+  // isn't already available it re-calculates it.
+  const int16_t* mixed_low_pass_data();
   const int16_t* low_pass_reference(int channel) const;
 
   // Float versions of the accessors, with automatic conversion back and forth
@@ -85,7 +86,6 @@ class AudioBuffer {
 
   // Use for int16 interleaved data.
   void DeinterleaveFrom(AudioFrame* audioFrame);
-  void InterleaveTo(AudioFrame* audioFrame) const;
   // If |data_changed| is false, only the non-audio data members will be copied
   // to |frame|.
   void InterleaveTo(AudioFrame* frame, bool data_changed) const;
@@ -97,9 +97,6 @@ class AudioBuffer {
   void CopyTo(int samples_per_channel,
               AudioProcessing::ChannelLayout layout,
               float* const* data);
-
-  void CopyAndMix(int num_mixed_channels);
-  void CopyAndMixLowPass(int num_mixed_channels);
   void CopyLowPassToReference();
 
  private:
@@ -112,8 +109,7 @@ class AudioBuffer {
   const int num_proc_channels_;
   const int output_samples_per_channel_;
   int samples_per_split_channel_;
-  int num_mixed_channels_;
+  bool mixed_low_pass_valid_;
-  int num_mixed_low_pass_channels_;
   bool reference_copied_;
   AudioFrame::VADActivity activity_;
 
@@ -121,7 +117,6 @@ class AudioBuffer {
   scoped_ptr<IFChannelBuffer> channels_;
   scoped_ptr<SplitChannelBuffer> split_channels_;
   scoped_ptr<SplitFilterStates[]> filter_states_;
-  scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_;
   scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
   scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
   scoped_ptr<ChannelBuffer<float> > input_buffer_;

webrtc/modules/audio_processing/gain_control_impl.cc

@@ -59,17 +59,11 @@ int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
 
   assert(audio->samples_per_split_channel() <= 160);
 
-  const int16_t* mixed_data = audio->low_pass_split_data(0);
-  if (audio->num_channels() > 1) {
-    audio->CopyAndMixLowPass(1);
-    mixed_data = audio->mixed_low_pass_data(0);
-  }
-
   for (int i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int err = WebRtcAgc_AddFarend(
         my_handle,
-        mixed_data,
+        audio->mixed_low_pass_data(),
         static_cast<int16_t>(audio->samples_per_split_channel()));
 
     if (err != apm_->kNoError) {

webrtc/modules/audio_processing/voice_detection_impl.cc

@@ -61,17 +61,11 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   }
   assert(audio->samples_per_split_channel() <= 160);
 
-  const int16_t* mixed_data = audio->low_pass_split_data(0);
-  if (audio->num_channels() > 1) {
-    audio->CopyAndMixLowPass(1);
-    mixed_data = audio->mixed_low_pass_data(0);
-  }
-
   // TODO(ajm): concatenate data in frame buffer here.
 
   int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
                                   apm_->proc_split_sample_rate_hz(),
-                                  mixed_data,
+                                  audio->mixed_low_pass_data(),
                                   frame_size_samples_);
   if (vad_ret == 0) {
     stream_has_voice_ = false;