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Remove PushResampler<T>::InitializeIfNeeded

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This switches from accepting a sample rate and convert to channel
size over to accepting the channel size.

Instead of InitializeIfNeeded:

* Offer a way to explicitly initialize PushResampler via the ctor
  (needed for VoiceActivityDetectorWrapper)
* Implicitly check for the right configuration from within Resample().
  (All calls to Resample() were preceded by a call to Initialize)

As part of this, refactor VoiceActivityDetectorWrapper (VADW):
* VADW is now initialized in the constructor and more const.
* Remove VADW::Initialize() and instead reconstruct VADW if needed.

Add constants for max sample rate and num channels to audio_util.h
In many cases the numbers for these values are embedded in the code
which has led to some inconsistency.

Bug: chromium:335805780
Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42587}
This commit is contained in:
Tommi 2024-07-03 14:23:48 +02:00 committed by WebRTC LUCI CQ
parent 5e49544a76
commit d6ef33e59b
14 changed files with 94 additions and 110 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
audio/audio_transport_impl.cc
View File

@ -81,9 +81,6 @@ int Resample(const AudioFrame& frame,
RTC_CHECK_EQ(destination.data().size(),
frame.num_channels_ * target_number_of_samples_per_channel);
resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
static_cast<int>(frame.num_channels()));
// TODO(yujo): Add special case handling of muted frames.
return resampler->Resample(frame.data_view(), destination);
}

8
audio/remix_resample.cc
View File

@ -61,14 +61,6 @@ void RemixAndResample(InterleavedView<const int16_t> src_data,
src_data = downmixed;
}
if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
src_data.num_channels()) == -1) {
RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
<< sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
<< dst_frame->sample_rate_hz_
<< ", num_channels = " << src_data.num_channels();
}
// TODO(yujo): for muted input frames, don't resample. Either 1) allow
// resampler to return output length without doing the resample, so we know
// how much to zero here; or 2) make resampler accept a hint that the input is

11
common_audio/include/audio_util.h
View File

@ -25,6 +25,17 @@ namespace webrtc {
typedef std::numeric_limits<int16_t> limits_int16;
// TODO(tommi, peah): Move these constants to their own header, e.g.
// `audio_constants.h`. Also consider if they should be in api/.
// Absolute highest acceptable sample rate supported for audio processing,
// capture and codecs. Note that for some components some cases a lower limit
// applies which typically is 48000 but in some cases is lower.
constexpr int kMaxSampleRateHz = 384000;
// Number of samples per channel for 10ms of audio at the highest sample rate.
constexpr size_t kMaxSamplesPerChannel10ms = kMaxSampleRateHz / 100u;
// The conversion functions use the following naming convention:
// S16: int16_t [-32768, 32767]
// Float: float [-1.0, 1.0]

17
common_audio/resampler/include/push_resampler.h
View File

@ -23,16 +23,13 @@ class PushSincResampler;
// Wraps PushSincResampler to provide stereo support.
// Note: This implementation assumes 10ms buffer sizes throughout.
template <typename T>
class PushResampler {
class PushResampler final {
public:
PushResampler();
virtual ~PushResampler();
// Must be called whenever the parameters change. Free to be called at any
// time as it is a no-op if parameters have not changed since the last call.
int InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
PushResampler(size_t src_samples_per_channel,
size_t dst_samples_per_channel,
size_t num_channels);
~PushResampler();
// Returns the total number of samples provided in destination (e.g. 32 kHz,
// 2 channel audio gives 640 samples).
@ -42,6 +39,12 @@ class PushResampler {
int Resample(MonoView<const T> src, MonoView<T> dst);
private:
// Ensures that source and destination buffers for deinterleaving are
// correctly configured prior to resampling that requires deinterleaving.
void EnsureInitialized(size_t src_samples_per_channel,
size_t dst_samples_per_channel,
size_t num_channels);
// Buffers used for when a deinterleaving step is necessary.
std::unique_ptr<T[]> source_;
std::unique_ptr<T[]> destination_;

61
common_audio/resampler/push_resampler.cc
View File

@ -22,55 +22,66 @@
namespace webrtc {
namespace {
// Maximum concurrent number of channels for `PushResampler<>`.
// Note that this may be different from what the maximum is for audio codecs.
constexpr int kMaxNumberOfChannels = 8;
} // namespace
template <typename T>
PushResampler<T>::PushResampler() = default;
template <typename T>
PushResampler<T>::PushResampler(size_t src_samples_per_channel,
size_t dst_samples_per_channel,
size_t num_channels) {
EnsureInitialized(src_samples_per_channel, dst_samples_per_channel,
num_channels);
}
template <typename T>
PushResampler<T>::~PushResampler() = default;
template <typename T>
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
void PushResampler<T>::EnsureInitialized(size_t src_samples_per_channel,
size_t dst_samples_per_channel,
size_t num_channels) {
// These checks used to be factored out of this template function due to
// Windows debug build issues with clang. http://crbug.com/615050
RTC_CHECK_GT(src_sample_rate_hz, 0);
RTC_CHECK_GT(dst_sample_rate_hz, 0);
RTC_CHECK_GT(num_channels, 0);
RTC_DCHECK_GT(src_samples_per_channel, 0);
RTC_DCHECK_GT(dst_samples_per_channel, 0);
RTC_DCHECK_GT(num_channels, 0);
RTC_DCHECK_LE(src_samples_per_channel, kMaxSamplesPerChannel10ms);
RTC_DCHECK_LE(dst_samples_per_channel, kMaxSamplesPerChannel10ms);
RTC_DCHECK_LE(num_channels, kMaxNumberOfChannels);
const size_t src_size_10ms_mono =
static_cast<size_t>(src_sample_rate_hz / 100);
const size_t dst_size_10ms_mono =
static_cast<size_t>(dst_sample_rate_hz / 100);
if (src_size_10ms_mono == SamplesPerChannel(source_view_) &&
dst_size_10ms_mono == SamplesPerChannel(destination_view_) &&
if (src_samples_per_channel == SamplesPerChannel(source_view_) &&
dst_samples_per_channel == SamplesPerChannel(destination_view_) &&
num_channels == NumChannels(source_view_)) {
// No-op if settings haven't changed.
return 0;
return;
}
// Allocate two buffers for all source and destination channels.
// Then organize source and destination views together with an array of
// resamplers for each channel in the deinterlaved buffers.
source_.reset(new T[src_size_10ms_mono * num_channels]);
destination_.reset(new T[dst_size_10ms_mono * num_channels]);
source_view_ =
DeinterleavedView<T>(source_.get(), src_size_10ms_mono, num_channels);
destination_view_ = DeinterleavedView<T>(destination_.get(),
dst_size_10ms_mono, num_channels);
source_.reset(new T[src_samples_per_channel * num_channels]);
destination_.reset(new T[dst_samples_per_channel * num_channels]);
source_view_ = DeinterleavedView<T>(source_.get(), src_samples_per_channel,
num_channels);
destination_view_ = DeinterleavedView<T>(
destination_.get(), dst_samples_per_channel, num_channels);
resamplers_.resize(num_channels);
for (size_t i = 0; i < num_channels; ++i) {
resamplers_[i] = std::make_unique<PushSincResampler>(src_size_10ms_mono,
dst_size_10ms_mono);
resamplers_[i] = std::make_unique<PushSincResampler>(
src_samples_per_channel, dst_samples_per_channel);
}
return 0;
}
template <typename T>
int PushResampler<T>::Resample(InterleavedView<const T> src,
InterleavedView<T> dst) {
EnsureInitialized(SamplesPerChannel(src), SamplesPerChannel(dst),
NumChannels(src));
RTC_DCHECK_EQ(NumChannels(src), NumChannels(source_view_));
RTC_DCHECK_EQ(NumChannels(dst), NumChannels(destination_view_));
RTC_DCHECK_EQ(SamplesPerChannel(src), SamplesPerChannel(source_view_));

21
common_audio/resampler/push_resampler_unittest.cc
View File

@ -19,29 +19,24 @@
namespace webrtc {
TEST(PushResamplerTest, VerifiesInputParameters) {
PushResampler<int16_t> resampler;
EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 8));
PushResampler<int16_t> resampler1(160, 160, 1);
PushResampler<int16_t> resampler2(160, 160, 2);
PushResampler<int16_t> resampler3(160, 160, 8);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(PushResamplerDeathTest, VerifiesBadInputParameters1) {
PushResampler<int16_t> resampler;
RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1),
"src_sample_rate_hz");
RTC_EXPECT_DEATH(PushResampler<int16_t>(-1, 160, 1),
"src_samples_per_channel");
}
TEST(PushResamplerDeathTest, VerifiesBadInputParameters2) {
PushResampler<int16_t> resampler;
RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1),
"dst_sample_rate_hz");
RTC_EXPECT_DEATH(PushResampler<int16_t>(160, -1, 1),
"dst_samples_per_channel");
}
TEST(PushResamplerDeathTest, VerifiesBadInputParameters3) {
PushResampler<int16_t> resampler;
RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0),
"num_channels");
RTC_EXPECT_DEATH(PushResampler<int16_t>(160, 16000, 0), "num_channels");
}
#endif

8
modules/audio_coding/acm2/acm_resampler.cc
View File

@ -44,14 +44,6 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
return static_cast<int>(dst.samples_per_channel());
}
if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
num_audio_channels) != 0) {
RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
<< out_freq_hz << ", " << num_audio_channels
<< ") failed.";
return -1;
}
int out_length = resampler_.Resample(src, dst);
if (out_length == -1) {
RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << src.data().size()

24
modules/audio_processing/agc2/vad_wrapper.cc
View File

@ -73,28 +73,20 @@ VoiceActivityDetectorWrapper::VoiceActivityDetectorWrapper(
int sample_rate_hz)
: vad_reset_period_frames_(
rtc::CheckedDivExact(vad_reset_period_ms, kFrameDurationMs)),
frame_size_(rtc::CheckedDivExact(sample_rate_hz, kNumFramesPerSecond)),
time_to_vad_reset_(vad_reset_period_frames_),
vad_(std::move(vad)) {
RTC_DCHECK(vad_);
vad_(std::move(vad)),
resampled_buffer_(
rtc::CheckedDivExact(vad_->SampleRateHz(), kNumFramesPerSecond)),
resampler_(frame_size_,
resampled_buffer_.size(),
/*num_channels=*/1) {
RTC_DCHECK_GT(vad_reset_period_frames_, 1);
resampled_buffer_.resize(
rtc::CheckedDivExact(vad_->SampleRateHz(), kNumFramesPerSecond));
Initialize(sample_rate_hz);
vad_->Reset();
}
VoiceActivityDetectorWrapper::~VoiceActivityDetectorWrapper() = default;
void VoiceActivityDetectorWrapper::Initialize(int sample_rate_hz) {
RTC_DCHECK_GT(sample_rate_hz, 0);
frame_size_ = rtc::CheckedDivExact(sample_rate_hz, kNumFramesPerSecond);
int status =
resampler_.InitializeIfNeeded(sample_rate_hz, vad_->SampleRateHz(),
/*num_channels=*/1);
constexpr int kStatusOk = 0;
RTC_DCHECK_EQ(status, kStatusOk);
vad_->Reset();
}
float VoiceActivityDetectorWrapper::Analyze(AudioFrameView<const float> frame) {
// Periodically reset the VAD.
time_to_vad_reset_--;

7
modules/audio_processing/agc2/vad_wrapper.h
View File

@ -60,9 +60,6 @@ class VoiceActivityDetectorWrapper {
delete;
~VoiceActivityDetectorWrapper();
// Initializes the VAD wrapper.
void Initialize(int sample_rate_hz);
// Analyzes the first channel of `frame` and returns the speech probability.
// `frame` must be a 10 ms frame with the sample rate specified in the last
// `Initialize()` call.
@ -70,11 +67,11 @@ class VoiceActivityDetectorWrapper {
private:
const int vad_reset_period_frames_;
int frame_size_;
const int frame_size_;
int time_to_vad_reset_;
PushResampler<float> resampler_;
std::unique_ptr<MonoVad> vad_;
std::vector<float> resampled_buffer_;
PushResampler<float> resampler_;
};
} // namespace webrtc

2
modules/audio_processing/agc2/vad_wrapper_unittest.cc
View File

@ -50,7 +50,7 @@ class MockVad : public VoiceActivityDetectorWrapper::MonoVad {
TEST(GainController2VoiceActivityDetectorWrapper, CtorAndInitReadSampleRate) {
auto vad = std::make_unique<MockVad>();
EXPECT_CALL(*vad, SampleRateHz)
.Times(2)
.Times(1)
.WillRepeatedly(Return(kSampleRate8kHz));
EXPECT_CALL(*vad, Reset).Times(AnyNumber());
auto vad_wrapper = std::make_unique<VoiceActivityDetectorWrapper>(

16
modules/audio_processing/audio_buffer.cc
View File

@ -15,7 +15,6 @@
#include <cstdint>
#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "modules/audio_processing/splitting_filter.h"
#include "rtc_base/checks.h"
@ -25,7 +24,6 @@ namespace {
constexpr size_t kSamplesPer32kHzChannel = 320;
constexpr size_t kSamplesPer48kHzChannel = 480;
constexpr size_t kMaxSamplesPerChannel = AudioBuffer::kMaxSampleRate / 100;
size_t NumBandsFromFramesPerChannel(size_t num_frames) {
if (num_frames == kSamplesPer32kHzChannel) {
@ -110,9 +108,9 @@ void AudioBuffer::CopyFrom(const float* const* stacked_data,
const bool resampling_needed = input_num_frames_ != buffer_num_frames_;
if (downmix_needed) {
RTC_DCHECK_GE(kMaxSamplesPerChannel, input_num_frames_);
RTC_DCHECK_GE(kMaxSamplesPerChannel10ms, input_num_frames_);
std::array<float, kMaxSamplesPerChannel> downmix;
std::array<float, kMaxSamplesPerChannel10ms> downmix;
if (downmix_by_averaging_) {
const float kOneByNumChannels = 1.f / input_num_channels_;
for (size_t i = 0; i < input_num_frames_; ++i) {
@ -230,7 +228,7 @@ void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
if (num_channels_ == 1) {
if (input_num_channels_ == 1) {
if (resampling_required) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
S16ToFloatS16(interleaved, input_num_frames_, float_buffer.data());
input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_,
data_->channels()[0],
@ -239,7 +237,7 @@ void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
S16ToFloatS16(interleaved, input_num_frames_, data_->channels()[0]);
}
} else {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
float* downmixed_data =
resampling_required ? float_buffer.data() : data_->channels()[0];
if (downmix_by_averaging_) {
@ -274,7 +272,7 @@ void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
};
if (resampling_required) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
for (size_t i = 0; i < num_channels_; ++i) {
deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
float_buffer.data());
@ -302,7 +300,7 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
int16_t* interleaved = interleaved_data;
if (num_channels_ == 1) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
if (resampling_required) {
output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_,
@ -335,7 +333,7 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
if (resampling_required) {
for (size_t i = 0; i < num_channels_; ++i) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
output_resamplers_[i]->Resample(data_->channels()[i],
buffer_num_frames_, float_buffer.data(),
output_num_frames_);

4
modules/audio_processing/audio_buffer.h
View File

@ -19,6 +19,7 @@
#include "api/audio/audio_processing.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"
namespace webrtc {
@ -32,7 +33,8 @@ enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
class AudioBuffer {
public:
static const int kSplitBandSize = 160;
static const int kMaxSampleRate = 384000;
// TODO(tommi): Remove this (`AudioBuffer::kMaxSampleRate`) constant.
static const int kMaxSampleRate = webrtc::kMaxSampleRateHz;
AudioBuffer(size_t input_rate,
size_t input_num_channels,
size_t buffer_rate,

6
modules/audio_processing/audio_processing_impl.cc
View File

@ -2157,17 +2157,11 @@ void AudioProcessingImpl::InitializeVoiceActivityDetector() {
return;
}
if (!submodules_.voice_activity_detector) {
RTC_DCHECK(!!submodules_.gain_controller2);
// TODO(bugs.webrtc.org/13663): Cache CPU features in APM and use here.
submodules_.voice_activity_detector =
std::make_unique<VoiceActivityDetectorWrapper>(
submodules_.gain_controller2->GetCpuFeatures(),
proc_fullband_sample_rate_hz());
} else {
submodules_.voice_activity_detector->Initialize(
proc_fullband_sample_rate_hz());
}
}
void AudioProcessingImpl::InitializeNoiseSuppressor() {

4
modules/audio_processing/audio_processing_unittest.cc
View File

@ -2173,8 +2173,8 @@ TEST_P(AudioProcessingTest, Formats) {
// don't match.
std::unique_ptr<float[]> cmp_data(new float[ref_length]);
PushResampler<float> resampler;
resampler.InitializeIfNeeded(out_rate, ref_rate, out_num);
PushResampler<float> resampler(out_samples_per_channel,
ref_samples_per_channel, out_num);
// Compute the resampling delay of the output relative to the reference,
// to find the region over which we should search for the best SNR.