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Support 48kHz in Noise Suppression

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Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7865 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org
2014-12-10 19:30:57 +00:00
parent d8ca723de7
commit c5ebbd98f5
17 changed files with 240 additions and 219 deletions
Download Patch File Download Diff File Expand all files Collapse all files

68
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_bands_(1),
samples_per_split_channel_(proc_samples_per_channel_),
mixed_low_pass_valid_(false),
reference_copied_(false),
@ -111,6 +112,7 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
if (proc_samples_per_channel_ == kSamplesPer32kHzChannel ||
proc_samples_per_channel_ == kSamplesPer48kHzChannel) {
samples_per_split_channel_ = kSamplesPer16kHzChannel;
num_bands_ = proc_samples_per_channel_ / samples_per_split_channel_;
split_channels_.push_back(new IFChannelBuffer(samples_per_split_channel_,
num_proc_channels_));
split_channels_.push_back(new IFChannelBuffer(samples_per_split_channel_,
@ -121,6 +123,8 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
num_proc_channels_));
}
}
bands_.reset(new int16_t*[num_proc_channels_ * kMaxNumBands]);
bands_f_.reset(new float*[num_proc_channels_ * kMaxNumBands]);
}
AudioBuffer::~AudioBuffer() {}
@ -216,14 +220,28 @@ int16_t* const* AudioBuffer::channels() {
return channels_->ibuf()->channels();
}
const int16_t* AudioBuffer::split_data_const(int channel, Band band) const {
const int16_t* const* chs = split_channels_const(band);
return chs ? chs[channel] : NULL;
const int16_t* const* AudioBuffer::split_bands_const(int channel) const {
// This is necessary to make sure that the int16_t data is up to date in the
// IFChannelBuffer.
// TODO(aluebs): Having to depend on this to get the updated data is bug
// prone. One solution is to have ChannelBuffer track the bands as well.
for (int i = 0; i < kMaxNumBands; ++i) {
int16_t* const* channels =
const_cast<int16_t* const*>(split_channels_const(static_cast<Band>(i)));
bands_[kMaxNumBands * channel + i] = channels ? channels[channel] : NULL;
}
return &bands_[kMaxNumBands * channel];
}
int16_t* AudioBuffer::split_data(int channel, Band band) {
int16_t* const* chs = split_channels(band);
return chs ? chs[channel] : NULL;
int16_t* const* AudioBuffer::split_bands(int channel) {
mixed_low_pass_valid_ = false;
// This is necessary to make sure that the int16_t data is up to date and the
// float data is marked as invalid in the IFChannelBuffer.
for (int i = 0; i < kMaxNumBands; ++i) {
int16_t* const* channels = split_channels(static_cast<Band>(i));
bands_[kMaxNumBands * channel + i] = channels ? channels[channel] : NULL;
}
return &bands_[kMaxNumBands * channel];
}
const int16_t* const* AudioBuffer::split_channels_const(Band band) const {
@ -260,14 +278,28 @@ float* const* AudioBuffer::channels_f() {
return channels_->fbuf()->channels();
}
const float* AudioBuffer::split_data_const_f(int channel, Band band) const {
const float* const* chs = split_channels_const_f(band);
return chs ? chs[channel] : NULL;
const float* const* AudioBuffer::split_bands_const_f(int channel) const {
// This is necessary to make sure that the float data is up to date in the
// IFChannelBuffer.
for (int i = 0; i < kMaxNumBands; ++i) {
float* const* channels =
const_cast<float* const*>(split_channels_const_f(static_cast<Band>(i)));
bands_f_[kMaxNumBands * channel + i] = channels ? channels[channel] : NULL;
}
return &bands_f_[kMaxNumBands * channel];
}
float* AudioBuffer::split_data_f(int channel, Band band) {
float* const* chs = split_channels_f(band);
return chs ? chs[channel] : NULL;
float* const* AudioBuffer::split_bands_f(int channel) {
mixed_low_pass_valid_ = false;
// This is necessary to make sure that the float data is up to date and the
// int16_t data is marked as invalid in the IFChannelBuffer.
for (int i = 0; i < kMaxNumBands; ++i) {
float* const* channels = split_channels_f(static_cast<Band>(i));
bands_f_[kMaxNumBands * channel + i] = channels ? channels[channel] : NULL;
}
return &bands_f_[kMaxNumBands * channel];
}
const float* const* AudioBuffer::split_channels_const_f(Band band) const {
@ -292,7 +324,7 @@ const int16_t* AudioBuffer::mixed_low_pass_data() {
assert(num_proc_channels_ == 1 || num_proc_channels_ == 2);
if (num_proc_channels_ == 1) {
return split_data_const(0, kBand0To8kHz);
return split_bands_const(0)[kBand0To8kHz];
}
if (!mixed_low_pass_valid_) {
@ -300,8 +332,8 @@ const int16_t* AudioBuffer::mixed_low_pass_data() {
mixed_low_pass_channels_.reset(
new ChannelBuffer<int16_t>(samples_per_split_channel_, 1));
}
StereoToMono(split_data_const(0, kBand0To8kHz),
split_data_const(1, kBand0To8kHz),
StereoToMono(split_bands_const(0)[kBand0To8kHz],
split_bands_const(1)[kBand0To8kHz],
mixed_low_pass_channels_->data(),
samples_per_split_channel_);
mixed_low_pass_valid_ = true;
@ -346,6 +378,10 @@ int AudioBuffer::samples_per_keyboard_channel() const {
return input_samples_per_channel_;
}
int AudioBuffer::num_bands() const {
return num_bands_;
}
// TODO(andrew): Do deinterleaving and mixing in one step?
void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
assert(proc_samples_per_channel_ == input_samples_per_channel_);
@ -404,7 +440,7 @@ void AudioBuffer::CopyLowPassToReference() {
num_proc_channels_));
}
for (int i = 0; i < num_proc_channels_; i++) {
low_pass_reference_channels_->CopyFrom(split_data_const(i, kBand0To8kHz),
low_pass_reference_channels_->CopyFrom(split_bands_const(i)[kBand0To8kHz],
i);
}
}