Pull out the PostFilter to its own NonlinearBeamformer API

webrtc/common_audio/channel_buffer.h

Index: webrtc/common_audio/channel_buffer.h
diff --git a/webrtc/common_audio/channel_buffer.h b/webrtc/common_audio/channel_buffer.h
index faecec9194ceb86be1d37104e24e7ad0a6915048..60ab0f4e6d9882d89e4dd1e9bdb3a9f500c334e5 100644
--- a/webrtc/common_audio/channel_buffer.h
+++ b/webrtc/common_audio/channel_buffer.h
@@ -49,6 +49,7 @@ class ChannelBuffer {
         num_frames_(num_frames),
         num_frames_per_band_(num_frames / num_bands),
         num_channels_(num_channels),

[peah-webrtc, 2016/06/08 12:04:55]

I see that the changes in audio_buffer are needed for the changes in
lapped_transform.
What I cannot see though, is why these changes are needed for this CL?

As I see it, the difference with this in this CL is that the downmixing has been
moved from within the beamformer to being done inside APM (but outside the
beamformer).

I thought before this CL this downmixing was partly done by the call
ca->set_num_channels(1); done inside APM and partly by the linear beamforming
done inside the beamformer, right?

And the way I interpret it, these changes that are now done in the audio_buffer,
channel_buffer and splitting_filter are needed to perform all of the downmixing
at the call to ca->set_num_channels(1); 
Is that correct?

[aluebs-webrtc, 2016/06/09 02:11:46]

Please see my other comment about the bug this solves.

+        num_official_channels_(num_channels),
         num_bands_(num_bands) {
     for (size_t i = 0; i < num_channels_; ++i) {
       for (size_t j = 0; j < num_bands_; ++j) {
@@ -88,11 +89,11 @@ class ChannelBuffer {
   // Usage:
   // bands(channel)[band][sample].
   // Where:
-  // 0 <= channel < |num_channels_|
+  // 0 <= channel < |num_official_channels_|
   // 0 <= band < |num_bands_|
   // 0 <= sample < |num_frames_per_band_|
   const T* const* bands(size_t channel) const {
-    RTC_DCHECK_LT(channel, num_channels_);
+    RTC_DCHECK_LT(channel, num_official_channels_);
     RTC_DCHECK_GE(channel, 0u);
     return &bands_[channel * num_bands_];
   }
@@ -105,7 +106,7 @@ class ChannelBuffer {
   // Returns |slice| for convenience.
   const T* const* Slice(T** slice, size_t start_frame) const {
     RTC_DCHECK_LT(start_frame, num_frames_);
-    for (size_t i = 0; i < num_channels_; ++i)
+    for (size_t i = 0; i < num_official_channels_; ++i)
       slice[i] = &channels_[i][start_frame];
     return slice;
   }
@@ -116,10 +117,15 @@ class ChannelBuffer {
 
   size_t num_frames() const { return num_frames_; }
   size_t num_frames_per_band() const { return num_frames_per_band_; }
-  size_t num_channels() const { return num_channels_; }
+  size_t num_channels() const { return num_official_channels_; }
   size_t num_bands() const { return num_bands_; }
   size_t size() const {return num_frames_ * num_channels_; }
 
+  void set_num_channels(size_t num_channels) {
+    RTC_DCHECK_LE(num_channels, num_channels_);
+    num_official_channels_ = num_channels;
+  }
+
   void SetDataForTesting(const T* data, size_t size) {
     RTC_CHECK_EQ(size, this->size());
     memcpy(data_.get(), data, size * sizeof(*data));
@@ -131,7 +137,10 @@ class ChannelBuffer {
   std::unique_ptr<T* []> bands_;
   const size_t num_frames_;
   const size_t num_frames_per_band_;
+  // Number of channels the internal buffer holds.
   const size_t num_channels_;
+  // Number of channels the user sees.
+  size_t num_official_channels_;
   const size_t num_bands_;
 };
 
@@ -152,7 +161,13 @@ class IFChannelBuffer {
 
   size_t num_frames() const { return ibuf_.num_frames(); }
   size_t num_frames_per_band() const { return ibuf_.num_frames_per_band(); }
-  size_t num_channels() const { return ibuf_.num_channels(); }
+  size_t num_channels() const {
+    return std::min(ibuf_.num_channels(), fbuf_.num_channels());
+  }
+  void set_num_channels(size_t num_channels) {
+    ibuf_.set_num_channels(num_channels);
+    fbuf_.set_num_channels(num_channels);
+  }
   size_t num_bands() const { return ibuf_.num_bands(); }
 
  private: