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Move the downmixing out of AudioBuffer

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This provides more flexibility if some component in AudioProcessing wants to operate before downmixing.
Now the AudioProcessing does only track the processing rate, but not the processing number of channels. This is tracked by the AudioBuffer itself and can be changed at any time to one smaller or equal the input number of channels. For each chunk it is reset to input number of channels and the end it should be equal to the output number of channels.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7879 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org
2014-12-11 17:09:21 +00:00
parent 0ca768b131
commit 27d106bcf7
4 changed files with 43 additions and 24 deletions
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27
webrtc/modules/audio_processing/audio_buffer.cc
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@ -65,6 +65,7 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
proc_samples_per_channel_(process_samples_per_channel),
num_proc_channels_(num_process_channels),
output_samples_per_channel_(output_samples_per_channel),
num_channels_(num_process_channels),
num_bands_(1),
samples_per_split_channel_(proc_samples_per_channel_),
mixed_low_pass_valid_(false),
@ -77,7 +78,7 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
assert(proc_samples_per_channel_ > 0);
assert(output_samples_per_channel_ > 0);
assert(num_input_channels_ > 0 && num_input_channels_ <= 2);
assert(num_proc_channels_ <= num_input_channels);
assert(num_proc_channels_ <= num_input_channels_);
if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
input_buffer_.reset(new ChannelBuffer<float>(input_samples_per_channel_,
@ -172,7 +173,7 @@ void AudioBuffer::CopyTo(int samples_per_channel,
AudioProcessing::ChannelLayout layout,
float* const* data) {
assert(samples_per_channel == output_samples_per_channel_);
assert(ChannelsFromLayout(layout) == num_proc_channels_);
assert(ChannelsFromLayout(layout) == num_channels_);
// Convert to the float range.
float* const* data_ptr = data;
@ -180,14 +181,14 @@ void AudioBuffer::CopyTo(int samples_per_channel,
// Convert to an intermediate buffer for subsequent resampling.
data_ptr = process_buffer_->channels();
}
for (int i = 0; i < num_proc_channels_; ++i) {
for (int i = 0; i < num_channels_; ++i) {
FloatS16ToFloat(channels_->fbuf()->channel(i), proc_samples_per_channel_,
data_ptr[i]);
}
// Resample.
if (output_samples_per_channel_ != proc_samples_per_channel_) {
for (int i = 0; i < num_proc_channels_; ++i) {
for (int i = 0; i < num_channels_; ++i) {
output_resamplers_[i]->Resample(data_ptr[i],
proc_samples_per_channel_,
data[i],
@ -201,6 +202,7 @@ void AudioBuffer::InitForNewData() {
mixed_low_pass_valid_ = false;
reference_copied_ = false;
activity_ = AudioFrame::kVadUnknown;
num_channels_ = num_proc_channels_;
}
const int16_t* AudioBuffer::data_const(int channel) const {
@ -362,7 +364,11 @@ AudioFrame::VADActivity AudioBuffer::activity() const {
}
int AudioBuffer::num_channels() const {
return num_proc_channels_;
return num_channels_;
}
void AudioBuffer::set_num_channels(int num_channels) {
num_channels_ = num_channels;
}
int AudioBuffer::samples_per_channel() const {
@ -412,8 +418,8 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
assert(proc_samples_per_channel_ == output_samples_per_channel_);
assert(num_proc_channels_ == num_input_channels_);
assert(frame->num_channels_ == num_proc_channels_);
assert(num_channels_ == num_input_channels_);
assert(frame->num_channels_ == num_channels_);
assert(frame->samples_per_channel_ == proc_samples_per_channel_);
frame->vad_activity_ = activity_;
@ -422,19 +428,20 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
}
int16_t* interleaved = frame->data_;
for (int i = 0; i < num_proc_channels_; i++) {
for (int i = 0; i < num_channels_; i++) {
int16_t* deinterleaved = channels_->ibuf()->channel(i);
int interleaved_idx = i;
for (int j = 0; j < proc_samples_per_channel_; j++) {
interleaved[interleaved_idx] = deinterleaved[j];
interleaved_idx += num_proc_channels_;
interleaved_idx += num_channels_;
}
}
}
void AudioBuffer::CopyLowPassToReference() {
reference_copied_ = true;
if (!low_pass_reference_channels_.get()) {
if (!low_pass_reference_channels_.get() ||
low_pass_reference_channels_->num_channels() != num_channels_) {
low_pass_reference_channels_.reset(
new ChannelBuffer<int16_t>(samples_per_split_channel_,
num_proc_channels_));