Add resampling support in AudioBuffer::DeinterleaveFrom
It is necessary for adding 48kHz support to the AudioProcessing::AnalyzeReverseStream int interface (It was not necessary for 32kHz since in that case the splitting filter is more efficient).
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56389004
Cr-Commit-Position: refs/heads/master@{#9241}
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index 794eb97..ec5e227 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -10,6 +10,7 @@
#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_audio/channel_buffer.h"
@@ -96,11 +97,6 @@
assert(num_input_channels_ > 0 && num_input_channels_ <= 2);
assert(num_proc_channels_ > 0 && num_proc_channels_ <= num_input_channels_);
- if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
- input_buffer_.reset(new ChannelBuffer<float>(input_num_frames_,
- num_proc_channels_));
- }
-
if (input_num_frames_ != proc_num_frames_ ||
output_num_frames_ != proc_num_frames_) {
// Create an intermediate buffer for resampling.
@@ -142,6 +138,12 @@
assert(num_frames == input_num_frames_);
assert(ChannelsFromLayout(layout) == num_input_channels_);
InitForNewData();
+ // Initialized lazily because there's a different condition in
+ // DeinterleaveFrom.
+ if ((num_input_channels_ == 2 && num_proc_channels_ == 1) && !input_buffer_) {
+ input_buffer_.reset(
+ new IFChannelBuffer(input_num_frames_, num_proc_channels_));
+ }
if (HasKeyboardChannel(layout)) {
keyboard_data_ = data[KeyboardChannelIndex(layout)];
@@ -152,9 +154,9 @@
if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
StereoToMono(data[0],
data[1],
- input_buffer_->channels()[0],
+ input_buffer_->fbuf()->channels()[0],
input_num_frames_);
- data_ptr = input_buffer_->channels();
+ data_ptr = input_buffer_->fbuf_const()->channels();
}
// Resample.
@@ -394,30 +396,44 @@
return num_bands_;
}
-// TODO(andrew): Do deinterleaving and mixing in one step?
+// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
- assert(proc_num_frames_ == input_num_frames_);
assert(frame->num_channels_ == num_input_channels_);
- assert(frame->samples_per_channel_ == proc_num_frames_);
+ assert(frame->samples_per_channel_ == input_num_frames_);
InitForNewData();
+ // Initialized lazily because there's a different condition in CopyFrom.
+ if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
+ input_buffer_.reset(
+ new IFChannelBuffer(input_num_frames_, num_proc_channels_));
+ }
activity_ = frame->vad_activity_;
+ int16_t* const* deinterleaved;
+ if (input_num_frames_ == proc_num_frames_) {
+ deinterleaved = data_->ibuf()->channels();
+ } else {
+ deinterleaved = input_buffer_->ibuf()->channels();
+ }
if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
// Downmix directly; no explicit deinterleaving needed.
- int16_t* downmixed = data_->ibuf()->channels()[0];
for (int i = 0; i < input_num_frames_; ++i) {
- downmixed[i] = (frame->data_[i * 2] + frame->data_[i * 2 + 1]) / 2;
+ deinterleaved[0][i] = (frame->data_[i * 2] + frame->data_[i * 2 + 1]) / 2;
}
} else {
assert(num_proc_channels_ == num_input_channels_);
- int16_t* interleaved = frame->data_;
+ Deinterleave(frame->data_,
+ input_num_frames_,
+ num_proc_channels_,
+ deinterleaved);
+ }
+
+ // Resample.
+ if (input_num_frames_ != proc_num_frames_) {
for (int i = 0; i < num_proc_channels_; ++i) {
- int16_t* deinterleaved = data_->ibuf()->channels()[i];
- int interleaved_idx = i;
- for (int j = 0; j < proc_num_frames_; ++j) {
- deinterleaved[j] = interleaved[interleaved_idx];
- interleaved_idx += num_proc_channels_;
- }
+ input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
+ input_num_frames_,
+ data_->fbuf()->channels()[i],
+ proc_num_frames_);
}
}
}
@@ -433,15 +449,10 @@
return;
}
- int16_t* interleaved = frame->data_;
- for (int i = 0; i < num_channels_; i++) {
- int16_t* deinterleaved = data_->ibuf()->channels()[i];
- int interleaved_idx = i;
- for (int j = 0; j < proc_num_frames_; j++) {
- interleaved[interleaved_idx] = deinterleaved[j];
- interleaved_idx += num_channels_;
- }
- }
+ Interleave(data_->ibuf()->channels(),
+ proc_num_frames_,
+ num_channels_,
+ frame->data_);
}
void AudioBuffer::CopyLowPassToReference() {
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 5288f25..4291fb3 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -12,7 +12,6 @@
#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/splitting_filter.h"
@@ -156,7 +155,7 @@
rtc::scoped_ptr<SplittingFilter> splitting_filter_;
rtc::scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
- rtc::scoped_ptr<ChannelBuffer<float> > input_buffer_;
+ rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
ScopedVector<PushSincResampler> input_resamplers_;
ScopedVector<PushSincResampler> output_resamplers_;
