Add empty 3 band splitting filter API

This is only an empty API that will never be used. For now is 48kHz not supported in AudioProcessing. For that it needs to be added in InitializeLocked. But before the 3 band filter bank needs to be populated.

BUG=webrtc:3146
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7715 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/audio_buffer.cc

@@ -20,7 +20,8 @@ namespace {
 enum {
   kSamplesPer8kHzChannel = 80,
   kSamplesPer16kHzChannel = 160,
-  kSamplesPer32kHzChannel = 320
+  kSamplesPer32kHzChannel = 320,
+  kSamplesPer48kHzChannel = 480
 };
 
 bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
@@ -171,13 +172,18 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
     }
   }
 
-  if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) {
+  if (proc_samples_per_channel_ == kSamplesPer32kHzChannel ||
+      proc_samples_per_channel_ == kSamplesPer48kHzChannel) {
     samples_per_split_channel_ = kSamplesPer16kHzChannel;
     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_));
     splitting_filter_.reset(new SplittingFilter(num_proc_channels_));
+    if (proc_samples_per_channel_ == kSamplesPer48kHzChannel) {
+      split_channels_super_high_.reset(
+          new IFChannelBuffer(samples_per_split_channel_, num_proc_channels_));
+    }
   }
 }
 
@@ -391,6 +397,18 @@ float* const* AudioBuffer::high_pass_split_channels_f() {
       : NULL;
 }
 
+const float* const* AudioBuffer::super_high_pass_split_channels_f() const {
+  return split_channels_super_high_.get()
+      ? split_channels_super_high_->fbuf_const()->channels()
+      : NULL;
+}
+
+float* const* AudioBuffer::super_high_pass_split_channels_f() {
+  return split_channels_super_high_.get()
+      ? split_channels_super_high_->fbuf()->channels()
+      : NULL;
+}
+
 const int16_t* AudioBuffer::mixed_low_pass_data() {
   // Currently only mixing stereo to mono is supported.
   assert(num_proc_channels_ == 1 || num_proc_channels_ == 2);
@@ -513,15 +531,29 @@ void AudioBuffer::CopyLowPassToReference() {
 }
 
 void AudioBuffer::SplitIntoFrequencyBands() {
-  splitting_filter_->TwoBandsAnalysis(
+  if (samples_per_channel() == kSamplesPer32kHzChannel) {
-      channels(), samples_per_channel(), num_proc_channels_,
+    splitting_filter_->TwoBandsAnalysis(
-      low_pass_split_channels(), high_pass_split_channels());
+        channels(), samples_per_channel(), num_proc_channels_,
+        low_pass_split_channels(), high_pass_split_channels());
+  } else if (samples_per_channel() == kSamplesPer48kHzChannel) {
+    splitting_filter_->ThreeBandsAnalysis(
+        channels_f(), samples_per_channel(), num_proc_channels_,
+        low_pass_split_channels_f(), high_pass_split_channels_f(),
+        super_high_pass_split_channels_f());
+  }
 }
 
 void AudioBuffer::MergeFrequencyBands() {
-  splitting_filter_->TwoBandsSynthesis(
+  if (samples_per_channel() == kSamplesPer32kHzChannel) {
-      low_pass_split_channels(), high_pass_split_channels(),
+    splitting_filter_->TwoBandsSynthesis(
-      samples_per_split_channel(), num_proc_channels_, channels());
+        low_pass_split_channels(), high_pass_split_channels(),
+        samples_per_split_channel(), num_proc_channels_, channels());
+  } else if (samples_per_channel() == kSamplesPer48kHzChannel) {
+    splitting_filter_->ThreeBandsSynthesis(
+        low_pass_split_channels_f(), high_pass_split_channels_f(),
+        super_high_pass_split_channels_f(), samples_per_split_channel(),
+        num_proc_channels_, channels_f());
+  }
 }
 
 }  // namespace webrtc

webrtc/modules/audio_processing/audio_buffer.h

@@ -78,6 +78,8 @@ class AudioBuffer {
   const float* const* low_pass_split_channels_f() const;
   float* const* high_pass_split_channels_f();
   const float* const* high_pass_split_channels_f() const;
+  float* const* super_high_pass_split_channels_f();
+  const float* const* super_high_pass_split_channels_f() const;
 
   const float* keyboard_data() const;
 
@@ -122,6 +124,7 @@ class AudioBuffer {
   scoped_ptr<IFChannelBuffer> channels_;
   scoped_ptr<IFChannelBuffer> split_channels_low_;
   scoped_ptr<IFChannelBuffer> split_channels_high_;
+  scoped_ptr<IFChannelBuffer> split_channels_super_high_;
   scoped_ptr<SplittingFilter> splitting_filter_;
   scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
   scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;

webrtc/modules/audio_processing/audio_processing_impl.cc

@@ -262,7 +262,8 @@ int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
   // demonstrated to work well for AEC in most practical scenarios.
   rev_proc_format_.set(rev_proc_rate, 1);
 
-  if (fwd_proc_format_.rate() == kSampleRate32kHz) {
+  if (fwd_proc_format_.rate() == kSampleRate32kHz ||
+      fwd_proc_format_.rate() == kSampleRate48kHz) {
     split_rate_ = kSampleRate16kHz;
   } else {
     split_rate_ = fwd_proc_format_.rate();
@@ -404,7 +405,8 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
-      frame->sample_rate_hz_ != kSampleRate32kHz) {
+      frame->sample_rate_hz_ != kSampleRate32kHz &&
+      frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
   if (echo_control_mobile_->is_enabled() &&
@@ -540,7 +542,8 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
-      frame->sample_rate_hz_ != kSampleRate32kHz) {
+      frame->sample_rate_hz_ != kSampleRate32kHz &&
+      frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
   // This interface does not tolerate different forward and reverse rates.
@@ -775,14 +778,16 @@ bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
 }
 
 bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
-  return (is_data_processed && fwd_proc_format_.rate() == kSampleRate32kHz);
+  return (is_data_processed && (fwd_proc_format_.rate() == kSampleRate32kHz ||
+          fwd_proc_format_.rate() == kSampleRate48kHz));
 }
 
 bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
   if (!is_data_processed && !voice_detection_->is_enabled()) {
     // Only level_estimator_ is enabled.
     return false;
-  } else if (fwd_proc_format_.rate() == kSampleRate32kHz) {
+  } else if (fwd_proc_format_.rate() == kSampleRate32kHz ||
+             fwd_proc_format_.rate() == kSampleRate48kHz) {
     // Something besides level_estimator_ is enabled, and we have super-wb.
     return true;
   }

webrtc/modules/audio_processing/include/audio_processing.h

@@ -380,7 +380,8 @@ class AudioProcessing {
   enum NativeRate {
     kSampleRate8kHz = 8000,
     kSampleRate16kHz = 16000,
-    kSampleRate32kHz = 32000
+    kSampleRate32kHz = 32000,
+    kSampleRate48kHz = 48000
   };
 
   static const int kChunkSizeMs = 10;

webrtc/modules/audio_processing/splitting_filter.cc

@@ -46,4 +46,20 @@ void SplittingFilter::TwoBandsSynthesis(const int16_t* const* low_band,
   }
 }
 
+void SplittingFilter::ThreeBandsAnalysis(const float* const* in_data,
+                                         int in_data_length,
+                                         int channels,
+                                         float* const* low_band,
+                                         float* const* high_band,
+                                         float* const* super_high_band) {
+}
+
+void SplittingFilter::ThreeBandsSynthesis(const float* const* low_band,
+                                          const float* const* high_band,
+                                          const float* const* super_high_band,
+                                          int band_length,
+                                          int channels,
+                                          float* const* out_data) {
+}
+
 }  // namespace webrtc

webrtc/modules/audio_processing/splitting_filter.h

@@ -47,6 +47,18 @@ class SplittingFilter {
                          int band_length,
                          int channels,
                          int16_t* const* out_data);
+  void ThreeBandsAnalysis(const float* const* in_data,
+                          int in_data_length,
+                          int channels,
+                          float* const* low_band,
+                          float* const* high_band,
+                          float* const* super_high_band);
+  void ThreeBandsSynthesis(const float* const* low_band,
+                           const float* const* high_band,
+                           const float* const* super_high_band,
+                           int band_length,
+                           int channels,
+                           float* const* out_data);
 
  private:
   int channels_;