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Add resampling support in AudioBuffer::DeinterleaveFrom

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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}
This commit is contained in:
Alejandro Luebs 2015-05-20 14:39:39 -07:00
parent 76b62ff1ad
commit 05c760533f
2 changed files with 40 additions and 30 deletions
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67
webrtc/modules/audio_processing/audio_buffer.cc
View File

@ -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 @@ AudioBuffer::AudioBuffer(int input_num_frames,
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 @@ void AudioBuffer::CopyFrom(const float* const* data,
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 @@ void AudioBuffer::CopyFrom(const float* const* data,
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 @@ int AudioBuffer::num_bands() const {
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 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
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() {

3
webrtc/modules/audio_processing/audio_buffer.h
View File

@ -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 @@ class AudioBuffer {
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_;