commit 422b9e098283bb9eece8689d580d11e78c9ebd66
author Gustaf Ullberg <gustaf@webrtc.org> Wed Oct 09 13:02:14 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Wed Oct 09 12:09:06 2019 +0000
tree a4b950eca842ec809ec3f4ed7a89d9dd3bc00806
parent 1d3008bfc6fccc2ebde8a4b8b615254f9a445aed

Run fullband processing at output rate on ARM

The audio processing in the band-split domain on ARM platforms
operate at a sampling frequency of 32 kHz. This CL upsamples
the signal to fullband before the "fullband processing"
if an output rate of 48 kHz is chosen.

Change-Id: I268acd33aff1fcfa4f75ba8c0fb3e16abb9f74e8
Bug: b/130016532
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155640
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29415}

modules/audio_processing/audio_buffer.cc

diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 4b0ca20..81ded91 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -65,9 +65,8 @@
       num_channels_(buffer_num_channels),
       num_bands_(NumBandsFromFramesPerChannel(buffer_num_frames_)),
       num_split_frames_(rtc::CheckedDivExact(buffer_num_frames_, num_bands_)),
-      data_(new ChannelBuffer<float>(buffer_num_frames_, buffer_num_channels_)),
-      output_buffer_(
-          new ChannelBuffer<float>(output_num_frames_, num_channels_)) {
+      data_(
+          new ChannelBuffer<float>(buffer_num_frames_, buffer_num_channels_)) {
   RTC_DCHECK_GT(input_num_frames_, 0);
   RTC_DCHECK_GT(buffer_num_frames_, 0);
   RTC_DCHECK_GT(output_num_frames_, 0);
@@ -185,6 +184,29 @@
   }
 }
 
+void AudioBuffer::CopyTo(AudioBuffer* buffer) const {
+  RTC_DCHECK_EQ(buffer->num_frames(), output_num_frames_);
+
+  const bool resampling_needed = output_num_frames_ != buffer_num_frames_;
+  if (resampling_needed) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_,
+                                      buffer->channels()[i],
+                                      buffer->num_frames());
+    }
+  } else {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      memcpy(buffer->channels()[i], data_->channels()[i],
+             buffer_num_frames_ * sizeof(**buffer->channels()));
+    }
+  }
+
+  for (size_t i = num_channels_; i < buffer->num_channels(); ++i) {
+    memcpy(buffer->channels()[i], buffer->channels()[0],
+           output_num_frames_ * sizeof(**buffer->channels()));
+  }
+}
+
 void AudioBuffer::RestoreNumChannels() {
   num_channels_ = buffer_num_channels_;
   data_->set_num_channels(buffer_num_channels_);