corysanin/tubestation
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Bug 1766646 - Vendor libwebrtc from 96315752d2

Browse Source 
Upstream commit: https://webrtc.googlesource.com/src/+/96315752d2499e12794507d6dea0c2652373754e
    Use backticks not vertical bars to denote variables in comments for /common_audio

    Bug: webrtc:12338
    Change-Id: I884db28e6d9a87d343be7c2616571a8bee28252c
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226944
    Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    Commit-Queue: Artem Titov <titovartem@webrtc.org>
    Cr-Commit-Position: refs/heads/master@{#34568}
This commit is contained in:
Michael Froman
2022-06-01 10:11:52 -05:00
committed by Connor Sheehan
parent 83fb78d619
commit a2cbdbfea0
44 changed files with 339 additions and 332 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
third_party/libwebrtc/README.moz-ff-commit vendored
View File

@@ -7242,3 +7242,6 @@ b0ea637ec2
# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
37f664f6d5
# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
96315752d2

2
third_party/libwebrtc/README.mozilla vendored
View File

@@ -4832,3 +4832,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-01T15:10:25.192841.
# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-01T15:11:05.918430.
# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-01T15:11:47.288140.

6
third_party/libwebrtc/common_audio/audio_converter.h vendored
View File

@@ -20,7 +20,7 @@
namespace webrtc {
// Format conversion (remixing and resampling) for audio. Only simple remixing
// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
// conversions are supported: downmix to mono (i.e. `dst_channels` == 1) or
// upmix from mono (i.e. |src_channels == 1|).
//
// The source and destination chunks have the same duration in time; specifying
@@ -35,8 +35,8 @@ class AudioConverter {
size_t dst_frames);
virtual ~AudioConverter() {}
// Convert |src|, containing |src_size| samples, to |dst|, having a sample
// capacity of |dst_capacity|. Both point to a series of buffers containing
// Convert `src`, containing `src_size` samples, to `dst`, having a sample
// capacity of `dst_capacity`. Both point to a series of buffers containing
// the samples for each channel. The sizes must correspond to the format
// passed to Create().
virtual void Convert(const float* const* src,

6
third_party/libwebrtc/common_audio/audio_converter_unittest.cc vendored
View File

@@ -25,7 +25,7 @@ namespace webrtc {
typedef std::unique_ptr<ChannelBuffer<float>> ScopedBuffer;
// Sets the signal value to increase by |data| with every sample.
// Sets the signal value to increase by `data` with every sample.
ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) {
const size_t num_channels = data.size();
ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
@@ -41,8 +41,8 @@ void VerifyParams(const ChannelBuffer<float>& ref,
EXPECT_EQ(ref.num_frames(), test.num_frames());
}
// Computes the best SNR based on the error between |ref_frame| and
// |test_frame|. It searches around |expected_delay| in samples between the
// Computes the best SNR based on the error between `ref_frame` and
// `test_frame`. It searches around `expected_delay` in samples between the
// signals to compensate for the resampling delay.
float ComputeSNR(const ChannelBuffer<float>& ref,
const ChannelBuffer<float>& test,

26
third_party/libwebrtc/common_audio/channel_buffer.h vendored
View File

@@ -29,15 +29,15 @@ namespace webrtc {
//
// The buffer structure is showed below for a 2 channel and 2 bands case:
//
// |data_|:
// `data_`:
// { [ --- b1ch1 --- ] [ --- b2ch1 --- ] [ --- b1ch2 --- ] [ --- b2ch2 --- ] }
//
// The pointer arrays for the same example are as follows:
//
// |channels_|:
// `channels_`:
// { [ b1ch1* ] [ b1ch2* ] [ b2ch1* ] [ b2ch2* ] }
//
// |bands_|:
// `bands_`:
// { [ b1ch1* ] [ b2ch1* ] [ b1ch2* ] [ b2ch2* ] }
template <typename T>
class ChannelBuffer {
@@ -81,15 +81,15 @@ class ChannelBuffer {
// If band is explicitly specificed, the channels for a specific band are
// returned and the usage becomes: channels(band)[channel][sample].
// Where:
// 0 <= band < |num_bands_|
// 0 <= channel < |num_allocated_channels_|
// 0 <= sample < |num_frames_per_band_|
// 0 <= band < `num_bands_`
// 0 <= channel < `num_allocated_channels_`
// 0 <= sample < `num_frames_per_band_`
// If band is not explicitly specified, the full-band channels (or lower band
// channels) are returned and the usage becomes: channels()[channel][sample].
// Where:
// 0 <= channel < |num_allocated_channels_|
// 0 <= sample < |num_frames_|
// 0 <= channel < `num_allocated_channels_`
// 0 <= sample < `num_frames_`
const T* const* channels(size_t band = 0) const {
RTC_DCHECK_LT(band, num_bands_);
return &channels_[band * num_allocated_channels_];
@@ -109,9 +109,9 @@ class ChannelBuffer {
// Usage:
// bands(channel)[band][sample].
// Where:
// 0 <= channel < |num_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_frames_per_band_|
// 0 <= channel < `num_channels_`
// 0 <= band < `num_bands_`
// 0 <= sample < `num_frames_per_band_`
const T* const* bands(size_t channel) const {
RTC_DCHECK_LT(channel, num_channels_);
RTC_DCHECK_GE(channel, 0);
@@ -129,8 +129,8 @@ class ChannelBuffer {
return bands_view_[channel];
}
// Sets the |slice| pointers to the |start_frame| position for each channel.
// Returns |slice| for convenience.
// Sets the `slice` pointers to the `start_frame` position for each channel.
// Returns `slice` for convenience.
const T* const* Slice(T** slice, size_t start_frame) const {
RTC_DCHECK_LT(start_frame, num_frames_);
for (size_t i = 0; i < num_channels_; ++i)

4
third_party/libwebrtc/common_audio/fir_filter.h vendored
View File

@@ -20,8 +20,8 @@ class FIRFilter {
public:
virtual ~FIRFilter() {}
// Filters the |in| data supplied.
// |out| must be previously allocated and it must be at least of |length|.
// Filters the `in` data supplied.
// `out` must be previously allocated and it must be at least of `length`.
virtual void Filter(const float* in, size_t length, float* out) = 0;
};

2
third_party/libwebrtc/common_audio/fir_filter_avx2.cc vendored
View File

@@ -54,7 +54,7 @@ void FIRFilterAVX2::Filter(const float* in, size_t length, float* out) {
memcpy(&state_[state_length_], in, length * sizeof(*in));
// Convolves the input signal |in| with the filter kernel |coefficients_|
// Convolves the input signal `in` with the filter kernel `coefficients_`
// taking into account the previous state.
for (size_t i = 0; i < length; ++i) {
float* in_ptr = &state_[i];

2
third_party/libwebrtc/common_audio/fir_filter_c.cc vendored
View File

@@ -34,7 +34,7 @@ FIRFilterC::FIRFilterC(const float* coefficients, size_t coefficients_length)
void FIRFilterC::Filter(const float* in, size_t length, float* out) {
RTC_DCHECK_GT(length, 0);
// Convolves the input signal |in| with the filter kernel |coefficients_|
// Convolves the input signal `in` with the filter kernel `coefficients_`
// taking into account the previous state.
for (size_t i = 0; i < length; ++i) {
out[i] = 0.f;

2
third_party/libwebrtc/common_audio/fir_filter_factory.h vendored
View File

@@ -20,7 +20,7 @@ class FIRFilter;
// Creates a filter with the given coefficients. All initial state values will
// be zeros.
// The length of the chunks fed to the filter should never be greater than
// |max_input_length|. This is needed because, when vectorizing it is
// `max_input_length`. This is needed because, when vectorizing it is
// necessary to concatenate the input after the state, and resizing this array
// dynamically is expensive.
FIRFilter* CreateFirFilter(const float* coefficients,

2
third_party/libwebrtc/common_audio/fir_filter_neon.cc vendored
View File

@@ -48,7 +48,7 @@ void FIRFilterNEON::Filter(const float* in, size_t length, float* out) {
memcpy(&state_[state_length_], in, length * sizeof(*in));
// Convolves the input signal |in| with the filter kernel |coefficients_|
// Convolves the input signal `in` with the filter kernel `coefficients_`
// taking into account the previous state.
for (size_t i = 0; i < length; ++i) {
float* in_ptr = &state_[i];

2
third_party/libwebrtc/common_audio/fir_filter_sse.cc vendored
View File

@@ -49,7 +49,7 @@ void FIRFilterSSE2::Filter(const float* in, size_t length, float* out) {
memcpy(&state_[state_length_], in, length * sizeof(*in));
// Convolves the input signal |in| with the filter kernel |coefficients_|
// Convolves the input signal `in` with the filter kernel `coefficients_`
// taking into account the previous state.
for (size_t i = 0; i < length; ++i) {
float* in_ptr = &state_[i];

24
third_party/libwebrtc/common_audio/include/audio_util.h vendored
View File

@@ -91,9 +91,9 @@ inline float FloatS16ToDbfs(float v) {
return 20.0f * std::log10(v) + kMinDbfs;
}
// Copy audio from |src| channels to |dest| channels unless |src| and |dest|
// point to the same address. |src| and |dest| must have the same number of
// channels, and there must be sufficient space allocated in |dest|.
// Copy audio from `src` channels to `dest` channels unless `src` and `dest`
// point to the same address. `src` and `dest` must have the same number of
// channels, and there must be sufficient space allocated in `dest`.
template <typename T>
void CopyAudioIfNeeded(const T* const* src,
int num_frames,
@@ -106,9 +106,9 @@ void CopyAudioIfNeeded(const T* const* src,
}
}
// Deinterleave audio from |interleaved| to the channel buffers pointed to
// by |deinterleaved|. There must be sufficient space allocated in the
// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
// Deinterleave audio from `interleaved` to the channel buffers pointed to
// by `deinterleaved`. There must be sufficient space allocated in the
// `deinterleaved` buffers (`num_channel` buffers with `samples_per_channel`
// per buffer).
template <typename T>
void Deinterleave(const T* interleaved,
@@ -125,9 +125,9 @@ void Deinterleave(const T* interleaved,
}
}
// Interleave audio from the channel buffers pointed to by |deinterleaved| to
// |interleaved|. There must be sufficient space allocated in |interleaved|
// (|samples_per_channel| * |num_channels|).
// Interleave audio from the channel buffers pointed to by `deinterleaved` to
// `interleaved`. There must be sufficient space allocated in `interleaved`
// (`samples_per_channel` * `num_channels`).
template <typename T>
void Interleave(const T* const* deinterleaved,
size_t samples_per_channel,
@@ -143,9 +143,9 @@ void Interleave(const T* const* deinterleaved,
}
}
// Copies audio from a single channel buffer pointed to by |mono| to each
// channel of |interleaved|. There must be sufficient space allocated in
// |interleaved| (|samples_per_channel| * |num_channels|).
// Copies audio from a single channel buffer pointed to by `mono` to each
// channel of `interleaved`. There must be sufficient space allocated in
// `interleaved` (`samples_per_channel` * `num_channels`).
template <typename T>
void UpmixMonoToInterleaved(const T* mono,
int num_frames,

2
third_party/libwebrtc/common_audio/real_fourier.h vendored
View File

@@ -50,7 +50,7 @@ class RealFourier {
// output (i.e. |2^order / 2 + 1|).
static size_t ComplexLength(int order);
// Buffer allocation helpers. The buffers are large enough to hold |count|
// Buffer allocation helpers. The buffers are large enough to hold `count`
// floats/complexes and suitably aligned for use by the implementation.
// The returned scopers are set up with proper deleters; the caller owns
// the allocated memory.

4
third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.cc vendored
View File

@@ -63,12 +63,12 @@ size_t PushSincResampler::Resample(const float* source,
// request through Run().
//
// If this wasn't done, SincResampler would call Run() twice on the first
// pass, and we'd have to introduce an entire |source_frames| of delay, rather
// pass, and we'd have to introduce an entire `source_frames` of delay, rather
// than the minimum half kernel.
//
// It works out that ChunkSize() is exactly the amount of output we need to
// request in order to prime the buffer with a single Run() request for
// |source_frames|.
// `source_frames`.
if (first_pass_)
resampler_->Resample(resampler_->ChunkSize(), destination);

8
third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.h vendored
View File

@@ -33,11 +33,11 @@ class PushSincResampler : public SincResamplerCallback {
PushSincResampler(size_t source_frames, size_t destination_frames);
~PushSincResampler() override;
// Perform the resampling. |source_frames| must always equal the
// |source_frames| provided at construction. |destination_capacity| must be
// at least as large as |destination_frames|. Returns the number of samples
// Perform the resampling. `source_frames` must always equal the
// `source_frames` provided at construction. `destination_capacity` must be
// at least as large as `destination_frames`. Returns the number of samples
// provided in destination (for convenience, since this will always be equal
// to |destination_frames|).
// to `destination_frames`).
size_t Resample(const int16_t* source,
size_t source_frames,
int16_t* destination,

2
third_party/libwebrtc/common_audio/resampler/push_sinc_resampler_unittest.cc vendored
View File

@@ -305,7 +305,7 @@ INSTANTIATE_TEST_SUITE_P(
// Next run through some additional cases interesting for WebRTC.
// We skip some extreme downsampled cases (192 -> {8, 16}, 96 -> 8)
// because they violate |kHighFrequencyMaxError|, which is not
// because they violate `kHighFrequencyMaxError`, which is not
// unexpected. It's very unlikely that we'll see these conversions in
// practice anyway.

18
third_party/libwebrtc/common_audio/resampler/sinc_resampler.cc vendored
View File

@@ -80,7 +80,7 @@
// 8) Else, if we're not on the second load, goto (4).
//
// Note: we're glossing over how the sub-sample handling works with
// |virtual_source_idx_|, etc.
// `virtual_source_idx_`, etc.
// MSVC++ requires this to be set before any other includes to get M_PI.
#define _USE_MATH_DEFINES
@@ -102,7 +102,7 @@ namespace webrtc {
namespace {
double SincScaleFactor(double io_ratio) {
// |sinc_scale_factor| is basically the normalized cutoff frequency of the
// `sinc_scale_factor` is basically the normalized cutoff frequency of the
// low-pass filter.
double sinc_scale_factor = io_ratio > 1.0 ? 1.0 / io_ratio : 1.0;
@@ -238,7 +238,7 @@ void SincResampler::SetRatio(double io_sample_rate_ratio) {
io_sample_rate_ratio_ = io_sample_rate_ratio;
// Optimize reinitialization by reusing values which are independent of
// |sinc_scale_factor|. Provides a 3x speedup.
// `sinc_scale_factor`. Provides a 3x speedup.
const double sinc_scale_factor = SincScaleFactor(io_sample_rate_ratio_);
for (size_t offset_idx = 0; offset_idx <= kKernelOffsetCount; ++offset_idx) {
for (size_t i = 0; i < kKernelSize; ++i) {
@@ -268,8 +268,8 @@ void SincResampler::Resample(size_t frames, float* destination) {
const double current_io_ratio = io_sample_rate_ratio_;
const float* const kernel_ptr = kernel_storage_.get();
while (remaining_frames) {
// |i| may be negative if the last Resample() call ended on an iteration
// that put |virtual_source_idx_| over the limit.
// `i` may be negative if the last Resample() call ended on an iteration
// that put `virtual_source_idx_` over the limit.
//
// Note: The loop construct here can severely impact performance on ARM
// or when built with clang. See https://codereview.chromium.org/18566009/
@@ -278,7 +278,7 @@ void SincResampler::Resample(size_t frames, float* destination) {
i > 0; --i) {
RTC_DCHECK_LT(virtual_source_idx_, block_size_);
// |virtual_source_idx_| lies in between two kernel offsets so figure out
// `virtual_source_idx_` lies in between two kernel offsets so figure out
// what they are.
const int source_idx = static_cast<int>(virtual_source_idx_);
const double subsample_remainder = virtual_source_idx_ - source_idx;
@@ -288,16 +288,16 @@ void SincResampler::Resample(size_t frames, float* destination) {
const int offset_idx = static_cast<int>(virtual_offset_idx);
// We'll compute "convolutions" for the two kernels which straddle
// |virtual_source_idx_|.
// `virtual_source_idx_`.
const float* const k1 = kernel_ptr + offset_idx * kKernelSize;
const float* const k2 = k1 + kKernelSize;
// Ensure |k1|, |k2| are 32-byte aligned for SIMD usage. Should always be
// Ensure `k1`, `k2` are 32-byte aligned for SIMD usage. Should always be
// true so long as kKernelSize is a multiple of 32.
RTC_DCHECK_EQ(0, reinterpret_cast<uintptr_t>(k1) % 32);
RTC_DCHECK_EQ(0, reinterpret_cast<uintptr_t>(k2) % 32);
// Initialize input pointer based on quantized |virtual_source_idx_|.
// Initialize input pointer based on quantized `virtual_source_idx_`.
const float* const input_ptr = r1_ + source_idx;
// Figure out how much to weight each kernel's "convolution".

30
third_party/libwebrtc/common_audio/resampler/sinc_resampler.h vendored
View File

@@ -25,8 +25,8 @@
namespace webrtc {
// Callback class for providing more data into the resampler. Expects |frames|
// of data to be rendered into |destination|; zero padded if not enough frames
// Callback class for providing more data into the resampler. Expects `frames`
// of data to be rendered into `destination`; zero padded if not enough frames
// are available to satisfy the request.
class SincResamplerCallback {
public:
@@ -53,10 +53,10 @@ class SincResampler {
static const size_t kKernelStorageSize =
kKernelSize * (kKernelOffsetCount + 1);
// Constructs a SincResampler with the specified |read_cb|, which is used to
// acquire audio data for resampling. |io_sample_rate_ratio| is the ratio
// of input / output sample rates. |request_frames| controls the size in
// frames of the buffer requested by each |read_cb| call. The value must be
// Constructs a SincResampler with the specified `read_cb`, which is used to
// acquire audio data for resampling. `io_sample_rate_ratio` is the ratio
// of input / output sample rates. `request_frames` controls the size in
// frames of the buffer requested by each `read_cb` call. The value must be
// greater than kKernelSize. Specify kDefaultRequestSize if there are no
// request size constraints.
SincResampler(double io_sample_rate_ratio,
@@ -64,11 +64,11 @@ class SincResampler {
SincResamplerCallback* read_cb);
virtual ~SincResampler();
// Resample |frames| of data from |read_cb_| into |destination|.
// Resample `frames` of data from `read_cb_` into `destination`.
void Resample(size_t frames, float* destination);
// The maximum size in frames that guarantees Resample() will only make a
// single call to |read_cb_| for more data.
// single call to `read_cb_` for more data.
size_t ChunkSize() const;
size_t request_frames() const { return request_frames_; }
@@ -77,12 +77,12 @@ class SincResampler {
// not call while Resample() is in progress.
void Flush();
// Update |io_sample_rate_ratio_|. SetRatio() will cause a reconstruction of
// Update `io_sample_rate_ratio_`. SetRatio() will cause a reconstruction of
// the kernels used for resampling. Not thread safe, do not call while
// Resample() is in progress.
//
// TODO(ajm): Use this in PushSincResampler rather than reconstructing
// SincResampler. We would also need a way to update |request_frames_|.
// SincResampler. We would also need a way to update `request_frames_`.
void SetRatio(double io_sample_rate_ratio);
float* get_kernel_for_testing() { return kernel_storage_.get(); }
@@ -97,11 +97,11 @@ class SincResampler {
// Selects runtime specific CPU features like SSE. Must be called before
// using SincResampler.
// TODO(ajm): Currently managed by the class internally. See the note with
// |convolve_proc_| below.
// `convolve_proc_` below.
void InitializeCPUSpecificFeatures();
// Compute convolution of |k1| and |k2| over |input_ptr|, resultant sums are
// linearly interpolated using |kernel_interpolation_factor|. On x86 and ARM
// Compute convolution of `k1` and `k2` over `input_ptr`, resultant sums are
// linearly interpolated using `kernel_interpolation_factor`. On x86 and ARM
// the underlying implementation is chosen at run time.
static float Convolve_C(const float* input_ptr,
const float* k1,
@@ -136,7 +136,7 @@ class SincResampler {
// Source of data for resampling.
SincResamplerCallback* read_cb_;
// The size (in samples) to request from each |read_cb_| execution.
// The size (in samples) to request from each `read_cb_` execution.
const size_t request_frames_;
// The number of source frames processed per pass.
@@ -165,7 +165,7 @@ class SincResampler {
double);
ConvolveProc convolve_proc_;
// Pointers to the various regions inside |input_buffer_|. See the diagram at
// Pointers to the various regions inside `input_buffer_`. See the diagram at
// the top of the .cc file for more information.
float* r0_;
float* const r1_;

2
third_party/libwebrtc/common_audio/resampler/sinc_resampler_avx2.cc vendored
View File

@@ -25,7 +25,7 @@ float SincResampler::Convolve_AVX2(const float* input_ptr,
__m256 m_sums1 = _mm256_setzero_ps();
__m256 m_sums2 = _mm256_setzero_ps();
// Based on |input_ptr| alignment, we need to use loadu or load. Unrolling
// Based on `input_ptr` alignment, we need to use loadu or load. Unrolling
// these loops has not been tested or benchmarked.
bool aligned_input = (reinterpret_cast<uintptr_t>(input_ptr) & 0x1F) == 0;
if (!aligned_input) {

2
third_party/libwebrtc/common_audio/resampler/sinc_resampler_sse.cc vendored
View File

@@ -27,7 +27,7 @@ float SincResampler::Convolve_SSE(const float* input_ptr,
__m128 m_sums1 = _mm_setzero_ps();
__m128 m_sums2 = _mm_setzero_ps();
// Based on |input_ptr| alignment, we need to use loadu or load. Unrolling
// Based on `input_ptr` alignment, we need to use loadu or load. Unrolling
// these loops hurt performance in local testing.
if (reinterpret_cast<uintptr_t>(input_ptr) & 0x0F) {
for (size_t i = 0; i < kKernelSize; i += 4) {

2
third_party/libwebrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h vendored
View File

@@ -24,7 +24,7 @@ namespace webrtc {
// resampler for the specific sample rate conversion being used.
class SinusoidalLinearChirpSource : public SincResamplerCallback {
public:
// |delay_samples| can be used to insert a fractional sample delay into the
// `delay_samples` can be used to insert a fractional sample delay into the
// source. It will produce zeros until non-negative time is reached.
SinusoidalLinearChirpSource(int sample_rate,
size_t samples,

10
third_party/libwebrtc/common_audio/ring_buffer.c vendored
View File

@@ -18,9 +18,9 @@
#include <string.h>
// Get address of region(s) from which we can read data.
// If the region is contiguous, |data_ptr_bytes_2| will be zero.
// If non-contiguous, |data_ptr_bytes_2| will be the size in bytes of the second
// region. Returns room available to be read or |element_count|, whichever is
// If the region is contiguous, `data_ptr_bytes_2` will be zero.
// If non-contiguous, `data_ptr_bytes_2` will be the size in bytes of the second
// region. Returns room available to be read or `element_count`, whichever is
// smaller.
static size_t GetBufferReadRegions(RingBuffer* buf,
size_t element_count,
@@ -120,7 +120,7 @@ size_t WebRtc_ReadBuffer(RingBuffer* self,
&buf_ptr_bytes_2);
if (buf_ptr_bytes_2 > 0) {
// We have a wrap around when reading the buffer. Copy the buffer data to
// |data| and point to it.
// `data` and point to it.
memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
memcpy(((char*) data) + buf_ptr_bytes_1, buf_ptr_2, buf_ptr_bytes_2);
buf_ptr_1 = data;
@@ -129,7 +129,7 @@ size_t WebRtc_ReadBuffer(RingBuffer* self,
memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
}
if (data_ptr) {
// |buf_ptr_1| == |data| in the case of a wrap.
// `buf_ptr_1` == `data` in the case of a wrap.
*data_ptr = read_count == 0 ? NULL : buf_ptr_1;
}

18
third_party/libwebrtc/common_audio/ring_buffer.h vendored
View File

@@ -39,14 +39,14 @@ void WebRtc_InitBuffer(RingBuffer* handle);
void WebRtc_FreeBuffer(void* handle);
// Reads data from the buffer. Returns the number of elements that were read.
// The |data_ptr| will point to the address where the read data is located.
// If no data can be read, |data_ptr| is set to |NULL|. If all data can be read
// without buffer wrap around then |data_ptr| will point to the location in the
// buffer. Otherwise, the data will be copied to |data| (memory allocation done
// by the user) and |data_ptr| points to the address of |data|. |data_ptr| is
// The `data_ptr` will point to the address where the read data is located.
// If no data can be read, `data_ptr` is set to `NULL`. If all data can be read
// without buffer wrap around then `data_ptr` will point to the location in the
// buffer. Otherwise, the data will be copied to `data` (memory allocation done
// by the user) and `data_ptr` points to the address of `data`. `data_ptr` is
// only guaranteed to be valid until the next call to WebRtc_WriteBuffer().
//
// To force a copying to |data|, pass a null |data_ptr|.
// To force a copying to `data`, pass a null `data_ptr`.
//
// Returns number of elements read.
size_t WebRtc_ReadBuffer(RingBuffer* handle,
@@ -54,14 +54,14 @@ size_t WebRtc_ReadBuffer(RingBuffer* handle,
void* data,
size_t element_count);
// Writes |data| to buffer and returns the number of elements written.
// Writes `data` to buffer and returns the number of elements written.
size_t WebRtc_WriteBuffer(RingBuffer* handle,
const void* data,
size_t element_count);
// Moves the buffer read position and returns the number of elements moved.
// Positive |element_count| moves the read position towards the write position,
// that is, flushing the buffer. Negative |element_count| moves the read
// Positive `element_count` moves the read position towards the write position,
// that is, flushing the buffer. Negative `element_count` moves the read
// position away from the the write position, that is, stuffing the buffer.
// Returns number of elements moved.
int WebRtc_MoveReadPtr(RingBuffer* handle, int element_count);

4
third_party/libwebrtc/common_audio/ring_buffer_unittest.cc vendored
View File

@@ -128,14 +128,14 @@ TEST(RingBufferTest, PassingNulltoReadBufferForcesMemcpy) {
EXPECT_EQ(kDataSize,
WebRtc_ReadBuffer(buffer.get(), reinterpret_cast<void**>(&data_ptr),
read_data, kDataSize));
// Copying was not necessary, so |read_data| has not been updated.
// Copying was not necessary, so `read_data` has not been updated.
CheckIncrementingData(data_ptr, kDataSize, 0);
CheckIncrementingData(read_data, kDataSize, kDataSize);
EXPECT_EQ(kDataSize, WebRtc_WriteBuffer(buffer.get(), write_data, kDataSize));
EXPECT_EQ(kDataSize,
WebRtc_ReadBuffer(buffer.get(), nullptr, read_data, kDataSize));
// Passing null forces a memcpy, so |read_data| is now updated.
// Passing null forces a memcpy, so `read_data` is now updated.
CheckIncrementingData(read_data, kDataSize, 0);
}

2
third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.h vendored
View File

@@ -26,7 +26,7 @@ extern "C" {
// - vector_length : Number of samples used in the dot product
// - scaling : The number of right bit shifts to apply on each term
// during calculation to avoid overflow, i.e., the
// output will be in Q(-|scaling|)
// output will be in Q(-`scaling`)
//
// Return value : The dot product in Q(-scaling)
int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,

2
third_party/libwebrtc/common_audio/signal_processing/include/real_fft.h vendored
View File

@@ -81,7 +81,7 @@ int WebRtcSpl_RealForwardFFT(struct RealFFT* self,
// boundary.
//
// Return Value:
// 0 or a positive number - a value that the elements in the |real_data_out|
// 0 or a positive number - a value that the elements in the `real_data_out`
// should be shifted left with in order to get
// correct physical values.
// -1 - Error with bad arguments (null pointers).

200
third_party/libwebrtc/common_audio/signal_processing/include/signal_processing_library.h vendored
View File

@@ -166,7 +166,7 @@ int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, size_t length);
// - vector : 16-bit input vector.
// - length : Number of samples in vector.
//
// Return value : Maximum sample value in |vector|.
// Return value : Maximum sample value in `vector`.
typedef int16_t (*MaxValueW16)(const int16_t* vector, size_t length);
extern const MaxValueW16 WebRtcSpl_MaxValueW16;
int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length);
@@ -183,7 +183,7 @@ int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, size_t length);
// - vector : 32-bit input vector.
// - length : Number of samples in vector.
//
// Return value : Maximum sample value in |vector|.
// Return value : Maximum sample value in `vector`.
typedef int32_t (*MaxValueW32)(const int32_t* vector, size_t length);
extern const MaxValueW32 WebRtcSpl_MaxValueW32;
int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length);
@@ -200,7 +200,7 @@ int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, size_t length);
// - vector : 16-bit input vector.
// - length : Number of samples in vector.
//
// Return value : Minimum sample value in |vector|.
// Return value : Minimum sample value in `vector`.
typedef int16_t (*MinValueW16)(const int16_t* vector, size_t length);
extern const MinValueW16 WebRtcSpl_MinValueW16;
int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length);
@@ -217,7 +217,7 @@ int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, size_t length);
// - vector : 32-bit input vector.
// - length : Number of samples in vector.
//
// Return value : Minimum sample value in |vector|.
// Return value : Minimum sample value in `vector`.
typedef int32_t (*MinValueW32)(const int32_t* vector, size_t length);
extern const MinValueW32 WebRtcSpl_MinValueW32;
int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length);
@@ -234,8 +234,8 @@ int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, size_t length);
// - vector : 16-bit input vector.
// - length : Number of samples in vector.
// Ouput:
// - max_val : Maximum sample value in |vector|.
// - min_val : Minimum sample value in |vector|.
// - max_val : Maximum sample value in `vector`.
// - min_val : Minimum sample value in `vector`.
void WebRtcSpl_MinMaxW16(const int16_t* vector,
size_t length,
int16_t* min_val,
@@ -426,7 +426,7 @@ void WebRtcSpl_AffineTransformVector(int16_t* out_vector,
//
// Input:
// - in_vector : Vector to calculate autocorrelation upon
// - in_vector_length : Length (in samples) of |vector|
// - in_vector_length : Length (in samples) of `vector`
// - order : The order up to which the autocorrelation should be
// calculated
//
@@ -438,7 +438,7 @@ void WebRtcSpl_AffineTransformVector(int16_t* out_vector,
// - scale : The number of left shifts required to obtain the
// auto-correlation in Q0
//
// Return value : Number of samples in |result|, i.e. (order+1)
// Return value : Number of samples in `result`, i.e. (order+1)
size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
size_t in_vector_length,
size_t order,
@@ -449,7 +449,7 @@ size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
// does NOT use the 64 bit class
//
// Input:
// - auto_corr : Vector with autocorrelation values of length >= |order|+1
// - auto_corr : Vector with autocorrelation values of length >= `order`+1
// - order : The LPC filter order (support up to order 20)
//
// Output:
@@ -462,7 +462,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* auto_corr,
int16_t* refl_coef,
size_t order);
// Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
// Converts reflection coefficients `refl_coef` to LPC coefficients `lpc_coef`.
// This version is a 16 bit operation.
//
// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
@@ -472,7 +472,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* auto_corr,
// Input:
// - refl_coef : Reflection coefficients in Q15 that should be converted
// to LPC coefficients
// - use_order : Number of coefficients in |refl_coef|
// - use_order : Number of coefficients in `refl_coef`
//
// Output:
// - lpc_coef : LPC coefficients in Q12
@@ -480,14 +480,14 @@ void WebRtcSpl_ReflCoefToLpc(const int16_t* refl_coef,
int use_order,
int16_t* lpc_coef);
// Converts LPC coefficients |lpc_coef| to reflection coefficients |refl_coef|.
// Converts LPC coefficients `lpc_coef` to reflection coefficients `refl_coef`.
// This version is a 16 bit operation.
// The conversion is implemented by the step-down algorithm.
//
// Input:
// - lpc_coef : LPC coefficients in Q12, that should be converted to
// reflection coefficients
// - use_order : Number of coefficients in |lpc_coef|
// - use_order : Number of coefficients in `lpc_coef`
//
// Output:
// - refl_coef : Reflection coefficients in Q15.
@@ -508,24 +508,24 @@ void WebRtcSpl_AutoCorrToReflCoef(const int32_t* auto_corr,
int16_t* refl_coef);
// The functions (with related pointer) calculate the cross-correlation between
// two sequences |seq1| and |seq2|.
// |seq1| is fixed and |seq2| slides as the pointer is increased with the
// amount |step_seq2|. Note the arguments should obey the relationship:
// |dim_seq| - 1 + |step_seq2| * (|dim_cross_correlation| - 1) <
// buffer size of |seq2|
// two sequences `seq1` and `seq2`.
// `seq1` is fixed and `seq2` slides as the pointer is increased with the
// amount `step_seq2`. Note the arguments should obey the relationship:
// `dim_seq` - 1 + `step_seq2` * (`dim_cross_correlation` - 1) <
// buffer size of `seq2`
//
// Input:
// - seq1 : First sequence (fixed throughout the correlation)
// - seq2 : Second sequence (slides |step_vector2| for each
// - seq2 : Second sequence (slides `step_vector2` for each
// new correlation)
// - dim_seq : Number of samples to use in the cross-correlation
// - dim_cross_correlation : Number of cross-correlations to calculate (the
// start position for |vector2| is updated for each
// start position for `vector2` is updated for each
// new one)
// - right_shifts : Number of right bit shifts to use. This will
// become the output Q-domain.
// - step_seq2 : How many (positive or negative) steps the
// |vector2| pointer should be updated for each new
// `vector2` pointer should be updated for each new
// cross-correlation value.
//
// Output:
@@ -575,11 +575,11 @@ void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
void WebRtcSpl_GetHanningWindow(int16_t* window, size_t size);
// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
// |in_vector|. Input and output values are in Q15.
// `in_vector`. Input and output values are in Q15.
//
// Inputs:
// - in_vector : Values to calculate sqrt(1 - x^2) of
// - vector_length : Length of vector |in_vector|
// - vector_length : Length of vector `in_vector`
//
// Output:
// - out_vector : Output values in Q15
@@ -667,9 +667,9 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
// Input:
// - data_in : Input samples (state in positions
// data_in[-order] .. data_in[-1])
// - data_in_length : Number of samples in |data_in| to be filtered.
// - data_in_length : Number of samples in `data_in` to be filtered.
// This must be at least
// |delay| + |factor|*(|out_vector_length|-1) + 1)
// `delay` + `factor`*(`out_vector_length`-1) + 1)
// - data_out_length : Number of down sampled samples desired
// - coefficients : Filter coefficients (in Q12)
// - coefficients_length: Number of coefficients (order+1)
@@ -677,7 +677,7 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
// - delay : Delay of filter (compensated for in out_vector)
// Output:
// - data_out : Filtered samples
// Return value : 0 if OK, -1 if |in_vector| is too short
// Return value : 0 if OK, -1 if `in_vector` is too short
typedef int (*DownsampleFast)(const int16_t* data_in,
size_t data_in_length,
int16_t* data_out,
@@ -723,12 +723,12 @@ int WebRtcSpl_DownsampleFast_mips(const int16_t* data_in,
int WebRtcSpl_ComplexFFT(int16_t vector[], int stages, int mode);
int WebRtcSpl_ComplexIFFT(int16_t vector[], int stages, int mode);
// Treat a 16-bit complex data buffer |complex_data| as an array of 32-bit
// Treat a 16-bit complex data buffer `complex_data` as an array of 32-bit
// values, and swap elements whose indexes are bit-reverses of each other.
//
// Input:
// - complex_data : Complex data buffer containing 2^|stages| real
// elements interleaved with 2^|stages| imaginary
// - complex_data : Complex data buffer containing 2^`stages` real
// elements interleaved with 2^`stages` imaginary
// elements: [Re Im Re Im Re Im....]
// - stages : Number of FFT stages. Must be at least 3 and at most
// 10, since the table WebRtcSpl_kSinTable1024[] is 1024
@@ -938,7 +938,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// WebRtcSpl_AddSatW32(...)
//
// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
// the numbers specified by the |var1| and |var2| parameters.
// the numbers specified by the `var1` and `var2` parameters.
//
// Input:
// - var1 : Input variable 1
@@ -952,7 +952,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// WebRtcSpl_SubSatW32(...)
//
// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
// of the numbers specified by the |var1| and |var2| parameters.
// of the numbers specified by the `var1` and `var2` parameters.
//
// Input:
// - var1 : Input variable 1
@@ -965,61 +965,61 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// WebRtcSpl_GetSizeInBits(...)
//
// Returns the # of bits that are needed at the most to represent the number
// specified by the |value| parameter.
// specified by the `value` parameter.
//
// Input:
// - value : Input value
//
// Return value : Number of bits needed to represent |value|
// Return value : Number of bits needed to represent `value`
//
//
// WebRtcSpl_NormW32(...)
//
// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
// signed number specified by the |value| parameter.
// signed number specified by the `value` parameter.
//
// Input:
// - value : Input value
//
// Return value : Number of bit shifts needed to 32-bit normalize |value|
// Return value : Number of bit shifts needed to 32-bit normalize `value`
//
//
// WebRtcSpl_NormW16(...)
//
// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
// signed number specified by the |value| parameter.
// signed number specified by the `value` parameter.
//
// Input:
// - value : Input value
//
// Return value : Number of bit shifts needed to 32-bit normalize |value|
// Return value : Number of bit shifts needed to 32-bit normalize `value`
//
//
// WebRtcSpl_NormU32(...)
//
// Norm returns the # of left shifts required to 32-bit normalize the unsigned
// 32-bit number specified by the |value| parameter.
// 32-bit number specified by the `value` parameter.
//
// Input:
// - value : Input value
//
// Return value : Number of bit shifts needed to 32-bit normalize |value|
// Return value : Number of bit shifts needed to 32-bit normalize `value`
//
//
// WebRtcSpl_GetScalingSquare(...)
//
// Returns the # of bits required to scale the samples specified in the
// |in_vector| parameter so that, if the squares of the samples are added the
// # of times specified by the |times| parameter, the 32-bit addition will not
// `in_vector` parameter so that, if the squares of the samples are added the
// # of times specified by the `times` parameter, the 32-bit addition will not
// overflow (result in int32_t).
//
// Input:
// - in_vector : Input vector to check scaling on
// - in_vector_length : Samples in |in_vector|
// - in_vector_length : Samples in `in_vector`
// - times : Number of additions to be performed
//
// Return value : Number of right bit shifts needed to avoid
@@ -1029,8 +1029,8 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_MemSetW16(...)
//
// Sets all the values in the int16_t vector |vector| of length
// |vector_length| to the specified value |set_value|
// Sets all the values in the int16_t vector `vector` of length
// `vector_length` to the specified value `set_value`
//
// Input:
// - vector : Pointer to the int16_t vector
@@ -1041,8 +1041,8 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_MemSetW32(...)
//
// Sets all the values in the int32_t vector |vector| of length
// |vector_length| to the specified value |set_value|
// Sets all the values in the int32_t vector `vector` of length
// `vector_length` to the specified value `set_value`
//
// Input:
// - vector : Pointer to the int16_t vector
@@ -1053,34 +1053,34 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_MemCpyReversedOrder(...)
//
// Copies all the values from the source int16_t vector |in_vector| to a
// destination int16_t vector |out_vector|. It is done in reversed order,
// meaning that the first sample of |in_vector| is copied to the last sample of
// the |out_vector|. The procedure continues until the last sample of
// |in_vector| has been copied to the first sample of |out_vector|. This
// Copies all the values from the source int16_t vector `in_vector` to a
// destination int16_t vector `out_vector`. It is done in reversed order,
// meaning that the first sample of `in_vector` is copied to the last sample of
// the `out_vector`. The procedure continues until the last sample of
// `in_vector` has been copied to the first sample of `out_vector`. This
// creates a reversed vector. Used in e.g. prediction in iLBC.
//
// Input:
// - in_vector : Pointer to the first sample in a int16_t vector
// of length |length|
// of length `length`
// - vector_length : Number of elements to copy
//
// Output:
// - out_vector : Pointer to the last sample in a int16_t vector
// of length |length|
// of length `length`
//
//
// WebRtcSpl_CopyFromEndW16(...)
//
// Copies the rightmost |samples| of |in_vector| (of length |in_vector_length|)
// to the vector |out_vector|.
// Copies the rightmost `samples` of `in_vector` (of length `in_vector_length`)
// to the vector `out_vector`.
//
// Input:
// - in_vector : Input vector
// - in_vector_length : Number of samples in |in_vector|
// - in_vector_length : Number of samples in `in_vector`
// - samples : Number of samples to extract (from right side)
// from |in_vector|
// from `in_vector`
//
// Output:
// - out_vector : Vector with the requested samples
@@ -1115,7 +1115,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// Output:
// - out_vector : Pointer to the result vector (can be the same as
// |in_vector|)
// `in_vector`)
//
//
@@ -1133,7 +1133,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// Output:
// - out_vector : Pointer to the result vector (can be the same as
// |in_vector|)
// `in_vector`)
//
//
@@ -1145,11 +1145,11 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// Input:
// - in_vector : Input vector
// - gain : Scaling gain
// - vector_length : Elements in the |in_vector|
// - vector_length : Elements in the `in_vector`
// - right_shifts : Number of right bit shifts applied
//
// Output:
// - out_vector : Output vector (can be the same as |in_vector|)
// - out_vector : Output vector (can be the same as `in_vector`)
//
//
@@ -1161,11 +1161,11 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// Input:
// - in_vector : Input vector
// - gain : Scaling gain
// - vector_length : Elements in the |in_vector|
// - vector_length : Elements in the `in_vector`
// - right_shifts : Number of right bit shifts applied
//
// Output:
// - out_vector : Output vector (can be the same as |in_vector|)
// - out_vector : Output vector (can be the same as `in_vector`)
//
//
@@ -1200,10 +1200,10 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// should be set to the last value in the vector
// - right_shifts : Number of right bit shift to be applied after the
// multiplication
// - vector_length : Number of elements in |in_vector|
// - vector_length : Number of elements in `in_vector`
//
// Output:
// - out_vector : Output vector (can be same as |in_vector|)
// - out_vector : Output vector (can be same as `in_vector`)
//
//
@@ -1217,10 +1217,10 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - window : Window vector.
// - right_shifts : Number of right bit shift to be applied after the
// multiplication
// - vector_length : Number of elements in |in_vector|
// - vector_length : Number of elements in `in_vector`
//
// Output:
// - out_vector : Output vector (can be same as |in_vector|)
// - out_vector : Output vector (can be same as `in_vector`)
//
//
@@ -1234,16 +1234,16 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - in_vector2 : Input vector 2
// - right_shifts : Number of right bit shift to be applied after the
// multiplication
// - vector_length : Number of elements in |in_vector1| and |in_vector2|
// - vector_length : Number of elements in `in_vector1` and `in_vector2`
//
// Output:
// - out_vector : Output vector (can be same as |in_vector1|)
// - out_vector : Output vector (can be same as `in_vector1`)
//
//
// WebRtcSpl_AddAffineVectorToVector(...)
//
// Adds an affine transformed vector to another vector |out_vector|, i.e,
// Adds an affine transformed vector to another vector `out_vector`, i.e,
// performs
// out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
//
@@ -1253,7 +1253,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
// but others can be used as well
// - right_shifts : Number of right bit shifts (0-16)
// - vector_length : Number of samples in |in_vector| and |out_vector|
// - vector_length : Number of samples in `in_vector` and `out_vector`
//
// Output:
// - out_vector : Vector with the output
@@ -1271,7 +1271,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
// but others can be used as well
// - right_shifts : Number of right bit shifts (0-16)
// - vector_length : Number of samples in |in_vector| and |out_vector|
// - vector_length : Number of samples in `in_vector` and `out_vector`
//
// Output:
// - out_vector : Vector with the output
@@ -1334,15 +1334,15 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - vector : Vector with the uniform values
// - seed : Updated seed value
//
// Return value : Number of samples in vector, i.e., |vector_length|
// Return value : Number of samples in vector, i.e., `vector_length`
//
//
// WebRtcSpl_Sqrt(...)
//
// Returns the square root of the input value |value|. The precision of this
// Returns the square root of the input value `value`. The precision of this
// function is integer precision, i.e., sqrt(8) gives 2 as answer.
// If |value| is a negative number then 0 is returned.
// If `value` is a negative number then 0 is returned.
//
// Algorithm:
//
@@ -1362,9 +1362,9 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_DivU32U16(...)
//
// Divides a uint32_t |num| by a uint16_t |den|.
// Divides a uint32_t `num` by a uint16_t `den`.
//
// If |den|==0, (uint32_t)0xFFFFFFFF is returned.
// If `den`==0, (uint32_t)0xFFFFFFFF is returned.
//
// Input:
// - num : Numerator
@@ -1377,9 +1377,9 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_DivW32W16(...)
//
// Divides a int32_t |num| by a int16_t |den|.
// Divides a int32_t `num` by a int16_t `den`.
//
// If |den|==0, (int32_t)0x7FFFFFFF is returned.
// If `den`==0, (int32_t)0x7FFFFFFF is returned.
//
// Input:
// - num : Numerator
@@ -1392,10 +1392,10 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_DivW32W16ResW16(...)
//
// Divides a int32_t |num| by a int16_t |den|, assuming that the
// Divides a int32_t `num` by a int16_t `den`, assuming that the
// result is less than 32768, otherwise an unpredictable result will occur.
//
// If |den|==0, (int16_t)0x7FFF is returned.
// If `den`==0, (int16_t)0x7FFF is returned.
//
// Input:
// - num : Numerator
@@ -1408,7 +1408,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_DivResultInQ31(...)
//
// Divides a int32_t |num| by a int16_t |den|, assuming that the
// Divides a int32_t `num` by a int16_t `den`, assuming that the
// absolute value of the denominator is larger than the numerator, otherwise
// an unpredictable result will occur.
//
@@ -1422,7 +1422,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// WebRtcSpl_DivW32HiLow(...)
//
// Divides a int32_t |num| by a denominator in hi, low format. The
// Divides a int32_t `num` by a denominator in hi, low format. The
// absolute value of the denominator has to be larger (or equal to) the
// numerator.
//
@@ -1447,7 +1447,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - scale_factor : Number of left bit shifts needed to get the physical
// energy value, i.e, to get the Q0 value
//
// Return value : Energy value in Q(-|scale_factor|)
// Return value : Energy value in Q(-`scale_factor`)
//
//
@@ -1458,15 +1458,15 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// Input:
// - ar_coef : AR-coefficient vector (values in Q12),
// ar_coef[0] must be 4096.
// - ar_coef_length : Number of coefficients in |ar_coef|.
// - ar_coef_length : Number of coefficients in `ar_coef`.
// - in_vector : Vector to be filtered.
// - in_vector_length : Number of samples in |in_vector|.
// - in_vector_length : Number of samples in `in_vector`.
// - filter_state : Current state (higher part) of the filter.
// - filter_state_length : Length (in samples) of |filter_state|.
// - filter_state_length : Length (in samples) of `filter_state`.
// - filter_state_low : Current state (lower part) of the filter.
// - filter_state_low_length : Length (in samples) of |filter_state_low|.
// - filter_state_low_length : Length (in samples) of `filter_state_low`.
// - out_vector_low_length : Maximum length (in samples) of
// |out_vector_low|.
// `out_vector_low`.
//
// Output:
// - filter_state : Updated state (upper part) vector.
@@ -1476,7 +1476,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - out_vector_low : Vector containing the lower part of the
// filtered values.
//
// Return value : Number of samples in the |out_vector|.
// Return value : Number of samples in the `out_vector`.
//
//
@@ -1484,11 +1484,11 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// Complex Inverse FFT
//
// Computes an inverse complex 2^|stages|-point FFT on the input vector, which
// Computes an inverse complex 2^`stages`-point FFT on the input vector, which
// is in bit-reversed order. The original content of the vector is destroyed in
// the process, since the input is overwritten by the output, normal-ordered,
// FFT vector. With X as the input complex vector, y as the output complex
// vector and with M = 2^|stages|, the following is computed:
// vector and with M = 2^`stages`, the following is computed:
//
// M-1
// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
@@ -1498,8 +1498,8 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// decimation-in-time algorithm with radix-2 butterfly technique.
//
// Input:
// - vector : In pointer to complex vector containing 2^|stages|
// real elements interleaved with 2^|stages| imaginary
// - vector : In pointer to complex vector containing 2^`stages`
// real elements interleaved with 2^`stages` imaginary
// elements.
// [ReImReImReIm....]
// The elements are in Q(-scale) domain, see more on Return
@@ -1518,10 +1518,10 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// - vector : Out pointer to the FFT vector (the same as input).
//
// Return Value : The scale value that tells the number of left bit shifts
// that the elements in the |vector| should be shifted with
// that the elements in the `vector` should be shifted with
// in order to get Q0 values, i.e. the physically correct
// values. The scale parameter is always 0 or positive,
// except if N>1024 (|stages|>10), which returns a scale
// except if N>1024 (`stages`>10), which returns a scale
// value of -1, indicating error.
//
@@ -1530,11 +1530,11 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
//
// Complex FFT
//
// Computes a complex 2^|stages|-point FFT on the input vector, which is in
// Computes a complex 2^`stages`-point FFT on the input vector, which is in
// bit-reversed order. The original content of the vector is destroyed in
// the process, since the input is overwritten by the output, normal-ordered,
// FFT vector. With x as the input complex vector, Y as the output complex
// vector and with M = 2^|stages|, the following is computed:
// vector and with M = 2^`stages`, the following is computed:
//
// M-1
// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
@@ -1549,8 +1549,8 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
// accuracy.
//
// Input:
// - vector : In pointer to complex vector containing 2^|stages| real
// elements interleaved with 2^|stages| imaginary elements.
// - vector : In pointer to complex vector containing 2^`stages` real
// elements interleaved with 2^`stages` imaginary elements.
// [ReImReImReIm....]
// The output is in the Q0 domain.
//

8
third_party/libwebrtc/common_audio/signal_processing/signal_processing_unittest.cc vendored
View File

@@ -482,13 +482,13 @@ TEST(SplTest, FilterTest) {
}
// MA filters.
// Note that the input data has |kFilterOrder| states before the actual
// Note that the input data has `kFilterOrder` states before the actual
// data (one sample).
WebRtcSpl_FilterMAFastQ12(&data_in[kFilterOrder], data_out, B,
kFilterOrder + 1, 1);
EXPECT_EQ(0, data_out[0]);
// AR filters.
// Note that the output data has |kFilterOrder| states before the actual
// Note that the output data has `kFilterOrder` states before the actual
// data (one sample).
WebRtcSpl_FilterARFastQ12(data_in, &data_out[kFilterOrder], A,
kFilterOrder + 1, 1);
@@ -639,11 +639,11 @@ TEST(SplTest, Resample48WithSaturationTest) {
32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
32767, 32767, 32767, 32767, 32767, 32767, 32767};
// All values in |out_vector| should be |kRefValue32kHz|.
// All values in `out_vector` should be `kRefValue32kHz`.
const int32_t kRefValue32kHz1 = -1077493760;
const int32_t kRefValue32kHz2 = 1077493645;
// After bit shift with saturation, |out_vector_w16| is saturated.
// After bit shift with saturation, `out_vector_w16` is saturated.
const int16_t kRefValue16kHz1 = -32768;
const int16_t kRefValue16kHz2 = 32767;

30
third_party/libwebrtc/common_audio/signal_processing/splitting_filter.c vendored
View File

@@ -41,7 +41,7 @@ static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
//
// Output:
// - out_data : Output data sequence (Q10), length equal to
// |data_length|
// `data_length`
//
static void WebRtcSpl_AllPassQMF(int32_t* in_data,
@@ -50,28 +50,30 @@ static void WebRtcSpl_AllPassQMF(int32_t* in_data,
const uint16_t* filter_coefficients,
int32_t* filter_state)
{
// The procedure is to filter the input with three first order all pass filters
// (cascade operations).
// The procedure is to filter the input with three first order all pass
// filters (cascade operations).
//
// a_3 + q^-1 a_2 + q^-1 a_1 + q^-1
// y[n] = ----------- ----------- ----------- x[n]
// 1 + a_3q^-1 1 + a_2q^-1 1 + a_1q^-1
//
// The input vector |filter_coefficients| includes these three filter coefficients.
// The filter state contains the in_data state, in_data[-1], followed by
// the out_data state, out_data[-1]. This is repeated for each cascade.
// The first cascade filter will filter the |in_data| and store the output in
// |out_data|. The second will the take the |out_data| as input and make an
// intermediate storage in |in_data|, to save memory. The third, and final, cascade
// filter operation takes the |in_data| (which is the output from the previous cascade
// filter) and store the output in |out_data|.
// Note that the input vector values are changed during the process.
// The input vector `filter_coefficients` includes these three filter
// coefficients. The filter state contains the in_data state, in_data[-1],
// followed by the out_data state, out_data[-1]. This is repeated for each
// cascade. The first cascade filter will filter the `in_data` and store
// the output in `out_data`. The second will the take the `out_data` as
// input and make an intermediate storage in `in_data`, to save memory. The
// third, and final, cascade filter operation takes the `in_data` (which is
// the output from the previous cascade filter) and store the output in
// `out_data`. Note that the input vector values are changed during the
// process.
size_t k;
int32_t diff;
// First all-pass cascade; filter from in_data to out_data.
// Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
// vector position n. Then the final output will be y[n] = y_3[n]
// Let y_i[n] indicate the output of cascade filter i (with filter
// coefficient a_i) at vector position n. Then the final output will be
// y[n] = y_3[n]
// First loop, use the states stored in memory.
// "diff" should be safe from wrap around since max values are 2^25

12
third_party/libwebrtc/common_audio/smoothing_filter.cc vendored
View File

@@ -23,12 +23,12 @@ SmoothingFilterImpl::SmoothingFilterImpl(int init_time_ms)
: init_time_ms_(init_time_ms),
// Duing the initalization time, we use an increasing alpha. Specifically,
// alpha(n) = exp(-powf(init_factor_, n)),
// where |init_factor_| is chosen such that
// where `init_factor_` is chosen such that
// alpha(init_time_ms_) = exp(-1.0f / init_time_ms_),
init_factor_(init_time_ms_ == 0
? 0.0f
: powf(init_time_ms_, -1.0f / init_time_ms_)),
// |init_const_| is to a factor to help the calculation during
// `init_const_` is to a factor to help the calculation during
// initialization phase.
init_const_(init_time_ms_ == 0
? 0.0f
@@ -57,7 +57,7 @@ void SmoothingFilterImpl::AddSample(float sample) {
absl::optional<float> SmoothingFilterImpl::GetAverage() {
if (!init_end_time_ms_) {
// |init_end_time_ms_| undefined since we have not received any sample.
// `init_end_time_ms_` undefined since we have not received any sample.
return absl::nullopt;
}
ExtrapolateLastSample(rtc::TimeMillis());
@@ -84,17 +84,17 @@ void SmoothingFilterImpl::ExtrapolateLastSample(int64_t time_ms) {
if (time_ms <= *init_end_time_ms_) {
// Current update is to be made during initialization phase.
// We update the state as if the |alpha| has been increased according
// We update the state as if the `alpha` has been increased according
// alpha(n) = exp(-powf(init_factor_, n)),
// where n is the time (in millisecond) since the first sample received.
// With algebraic derivation as shown in the Appendix, we can find that the
// state can be updated in a similar manner as if alpha is a constant,
// except for a different multiplier.
if (init_time_ms_ == 0) {
// This means |init_factor_| = 0.
// This means `init_factor_` = 0.
multiplier = 0.0f;
} else if (init_time_ms_ == 1) {
// This means |init_factor_| = 1.
// This means `init_factor_` = 1.
multiplier = std::exp(last_state_time_ms_ - time_ms);
} else {
multiplier = std::exp(

6
third_party/libwebrtc/common_audio/smoothing_filter.h vendored
View File

@@ -33,13 +33,13 @@ class SmoothingFilter {
// assumed to equal the last received sample.
class SmoothingFilterImpl final : public SmoothingFilter {
public:
// |init_time_ms| is initialization time. It defines a period starting from
// `init_time_ms` is initialization time. It defines a period starting from
// the arriving time of the first sample. During this period, the exponential
// filter uses a varying time constant so that a smaller time constant will be
// applied to the earlier samples. This is to allow the the filter to adapt to
// earlier samples quickly. After the initialization period, the time constant
// will be set to |init_time_ms| first and can be changed through
// |SetTimeConstantMs|.
// will be set to `init_time_ms` first and can be changed through
// `SetTimeConstantMs`.
explicit SmoothingFilterImpl(int init_time_ms);
SmoothingFilterImpl() = delete;

8
third_party/libwebrtc/common_audio/smoothing_filter_unittest.cc vendored
View File

@@ -34,7 +34,7 @@ struct SmoothingFilterStates {
// This function does the following:
// 1. Add a sample to filter at current clock,
// 2. Advance the clock by |advance_time_ms|,
// 2. Advance the clock by `advance_time_ms`,
// 3. Get the output of both SmoothingFilter and verify that it equals to an
// expected value.
void CheckOutput(SmoothingFilterStates* states,
@@ -141,7 +141,7 @@ TEST(SmoothingFilterTest, CannotChangeTimeConstantDuringInitialization) {
SmoothingFilterStates states(kInitTimeMs);
states.smoothing_filter.AddSample(0.0);
// During initialization, |SetTimeConstantMs| does not take effect.
// During initialization, `SetTimeConstantMs` does not take effect.
states.fake_clock.AdvanceTime(TimeDelta::Millis(kInitTimeMs - 1));
states.smoothing_filter.AddSample(0.0);
@@ -152,11 +152,11 @@ TEST(SmoothingFilterTest, CannotChangeTimeConstantDuringInitialization) {
states.fake_clock.AdvanceTime(TimeDelta::Millis(1));
states.smoothing_filter.AddSample(0.0);
// When initialization finishes, the time constant should be come
// |kInitTimeConstantMs|.
// `kInitTimeConstantMs`.
EXPECT_FLOAT_EQ(std::exp(-1.0f / kInitTimeMs),
states.smoothing_filter.alpha());
// After initialization, |SetTimeConstantMs| takes effect.
// After initialization, `SetTimeConstantMs` takes effect.
EXPECT_TRUE(states.smoothing_filter.SetTimeConstantMs(kInitTimeMs * 2));
EXPECT_FLOAT_EQ(std::exp(-1.0f / (kInitTimeMs * 2)),
states.smoothing_filter.alpha());

4
third_party/libwebrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h vendored
View File

@@ -13,9 +13,9 @@
//
// WebRtcSpl_SqrtFloor(...)
//
// Returns the square root of the input value |value|. The precision of this
// Returns the square root of the input value `value`. The precision of this
// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
// If |value| is a negative number then 0 is returned.
// If `value` is a negative number then 0 is returned.
//
// Algorithm:
//

4
third_party/libwebrtc/common_audio/vad/include/webrtc_vad.h vendored
View File

@@ -54,7 +54,7 @@ int WebRtcVad_Init(VadInst* handle);
// has not been initialized).
int WebRtcVad_set_mode(VadInst* handle, int mode);
// Calculates a VAD decision for the |audio_frame|. For valid sampling rates
// Calculates a VAD decision for the `audio_frame`. For valid sampling rates
// frame lengths, see the description of WebRtcVad_ValidRatesAndFrameLengths().
//
// - handle [i/o] : VAD Instance. Needs to be initialized by
@@ -71,7 +71,7 @@ int WebRtcVad_Process(VadInst* handle,
const int16_t* audio_frame,
size_t frame_length);
// Checks for valid combinations of |rate| and |frame_length|. We support 10,
// Checks for valid combinations of `rate` and `frame_length`. We support 10,
// 20 and 30 ms frames and the rates 8000, 16000 and 32000 Hz.
//
// - rate [i] : Sampling frequency (Hz).

38
third_party/libwebrtc/common_audio/vad/vad_core.c vendored
View File

@@ -90,11 +90,11 @@ static const int16_t kOverHangMax2VAG[3] = { 9, 5, 3 };
static const int16_t kLocalThresholdVAG[3] = { 94, 94, 94 };
static const int16_t kGlobalThresholdVAG[3] = { 1100, 1050, 1100 };
// Calculates the weighted average w.r.t. number of Gaussians. The |data| are
// updated with an |offset| before averaging.
// Calculates the weighted average w.r.t. number of Gaussians. The `data` are
// updated with an `offset` before averaging.
//
// - data [i/o] : Data to average.
// - offset [i] : An offset added to |data|.
// - offset [i] : An offset added to `data`.
// - weights [i] : Weights used for averaging.
//
// returns : The weighted average.
@@ -124,7 +124,7 @@ static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
// type of signal is most probable.
//
// - self [i/o] : Pointer to VAD instance
// - features [i] : Feature vector of length |kNumChannels|
// - features [i] : Feature vector of length `kNumChannels`
// = log10(energy in frequency band)
// - total_power [i] : Total power in audio frame.
// - frame_length [i] : Number of input samples
@@ -183,10 +183,10 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// H1: Speech
//
// We combine a global LRT with local tests, for each frequency sub-band,
// here defined as |channel|.
// here defined as `channel`.
for (channel = 0; channel < kNumChannels; channel++) {
// For each channel we model the probability with a GMM consisting of
// |kNumGaussians|, with different means and standard deviations depending
// `kNumGaussians`, with different means and standard deviations depending
// on H0 or H1.
h0_test = 0;
h1_test = 0;
@@ -234,7 +234,7 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
}
log_likelihood_ratio = shifts_h0 - shifts_h1;
// Update |sum_log_likelihood_ratios| with spectrum weighting. This is
// Update `sum_log_likelihood_ratios` with spectrum weighting. This is
// used for the global VAD decision.
sum_log_likelihood_ratios +=
(int32_t) (log_likelihood_ratio * kSpectrumWeight[channel]);
@@ -326,7 +326,7 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
if (vadflag) {
// Update speech mean vector:
// |deltaS| = (x-mu)/sigma^2
// `deltaS` = (x-mu)/sigma^2
// sgprvec[k] = |speech_probability[k]| /
// (|speech_probability[0]| + |speech_probability[1]|)
@@ -409,35 +409,35 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
}
// Separate models if they are too close.
// |noise_global_mean| in Q14 (= Q7 * Q7).
// `noise_global_mean` in Q14 (= Q7 * Q7).
noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
&kNoiseDataWeights[channel]);
// |speech_global_mean| in Q14 (= Q7 * Q7).
// `speech_global_mean` in Q14 (= Q7 * Q7).
speech_global_mean = WeightedAverage(&self->speech_means[channel], 0,
&kSpeechDataWeights[channel]);
// |diff| = "global" speech mean - "global" noise mean.
// `diff` = "global" speech mean - "global" noise mean.
// (Q14 >> 9) - (Q14 >> 9) = Q5.
diff = (int16_t) (speech_global_mean >> 9) -
(int16_t) (noise_global_mean >> 9);
if (diff < kMinimumDifference[channel]) {
tmp_s16 = kMinimumDifference[channel] - diff;
// |tmp1_s16| = ~0.8 * (kMinimumDifference - diff) in Q7.
// |tmp2_s16| = ~0.2 * (kMinimumDifference - diff) in Q7.
// `tmp1_s16` = ~0.8 * (kMinimumDifference - diff) in Q7.
// `tmp2_s16` = ~0.2 * (kMinimumDifference - diff) in Q7.
tmp1_s16 = (int16_t)((13 * tmp_s16) >> 2);
tmp2_s16 = (int16_t)((3 * tmp_s16) >> 2);
// Move Gaussian means for speech model by |tmp1_s16| and update
// |speech_global_mean|. Note that |self->speech_means[channel]| is
// Move Gaussian means for speech model by `tmp1_s16` and update
// `speech_global_mean`. Note that |self->speech_means[channel]| is
// changed after the call.
speech_global_mean = WeightedAverage(&self->speech_means[channel],
tmp1_s16,
&kSpeechDataWeights[channel]);
// Move Gaussian means for noise model by -|tmp2_s16| and update
// |noise_global_mean|. Note that |self->noise_means[channel]| is
// Move Gaussian means for noise model by -`tmp2_s16` and update
// `noise_global_mean`. Note that |self->noise_means[channel]| is
// changed after the call.
noise_global_mean = WeightedAverage(&self->noise_means[channel],
-tmp2_s16,
@@ -534,7 +534,7 @@ int WebRtcVad_InitCore(VadInstT* self) {
self->mean_value[i] = 1600;
}
// Set aggressiveness mode to default (=|kDefaultMode|).
// Set aggressiveness mode to default (=`kDefaultMode`).
if (WebRtcVad_set_mode_core(self, kDefaultMode) != 0) {
return -1;
}
@@ -609,7 +609,7 @@ int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
int vad;
size_t i;
int16_t speech_nb[240]; // 30 ms in 8 kHz.
// |tmp_mem| is a temporary memory used by resample function, length is
// `tmp_mem` is a temporary memory used by resample function, length is
// frame length in 10 ms (480 samples) + 256 extra.
int32_t tmp_mem[480 + 256] = { 0 };
const size_t kFrameLen10ms48khz = 480;

8
third_party/libwebrtc/common_audio/vad/vad_core.h vendored
View File

@@ -30,14 +30,14 @@ typedef struct VadInstT_ {
int16_t speech_means[kTableSize];
int16_t noise_stds[kTableSize];
int16_t speech_stds[kTableSize];
// TODO(bjornv): Change to |frame_count|.
// TODO(bjornv): Change to `frame_count`.
int32_t frame_counter;
int16_t over_hang; // Over Hang
int16_t num_of_speech;
// TODO(bjornv): Change to |age_vector|.
// TODO(bjornv): Change to `age_vector`.
int16_t index_vector[16 * kNumChannels];
int16_t low_value_vector[16 * kNumChannels];
// TODO(bjornv): Change to |median|.
// TODO(bjornv): Change to `median`.
int16_t mean_value[kNumChannels];
int16_t upper_state[5];
int16_t lower_state[5];
@@ -51,7 +51,7 @@ typedef struct VadInstT_ {
} VadInstT;
// Initializes the core VAD component. The default aggressiveness mode is
// controlled by |kDefaultMode| in vad_core.c.
// controlled by `kDefaultMode` in vad_core.c.
//
// - self [i/o] : Instance that should be initialized
//

98
third_party/libwebrtc/common_audio/vad/vad_filterbank.c vendored
View File

@@ -28,7 +28,7 @@ static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
// Adjustment for division with two in SplitFilter.
static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
// High pass filtering, with a cut-off frequency at 80 Hz, if the |data_in| is
// High pass filtering, with a cut-off frequency at 80 Hz, if the `data_in` is
// sampled at 500 Hz.
//
// - data_in [i] : Input audio data sampled at 500 Hz.
@@ -69,9 +69,9 @@ static void HighPassFilter(const int16_t* data_in, size_t data_length,
}
}
// All pass filtering of |data_in|, used before splitting the signal into two
// All pass filtering of `data_in`, used before splitting the signal into two
// frequency bands (low pass vs high pass).
// Note that |data_in| and |data_out| can NOT correspond to the same address.
// Note that `data_in` and `data_out` can NOT correspond to the same address.
//
// - data_in [i] : Input audio signal given in Q0.
// - data_length [i] : Length of input and output data.
@@ -104,17 +104,17 @@ static void AllPassFilter(const int16_t* data_in, size_t data_length,
*filter_state = (int16_t) (state32 >> 16); // Q(-1)
}
// Splits |data_in| into |hp_data_out| and |lp_data_out| corresponding to
// Splits `data_in` into `hp_data_out` and `lp_data_out` corresponding to
// an upper (high pass) part and a lower (low pass) part respectively.
//
// - data_in [i] : Input audio data to be split into two frequency bands.
// - data_length [i] : Length of |data_in|.
// - data_length [i] : Length of `data_in`.
// - upper_state [i/o] : State of the upper filter, given in Q(-1).
// - lower_state [i/o] : State of the lower filter, given in Q(-1).
// - hp_data_out [o] : Output audio data of the upper half of the spectrum.
// The length is |data_length| / 2.
// The length is `data_length` / 2.
// - lp_data_out [o] : Output audio data of the lower half of the spectrum.
// The length is |data_length| / 2.
// The length is `data_length` / 2.
static void SplitFilter(const int16_t* data_in, size_t data_length,
int16_t* upper_state, int16_t* lower_state,
int16_t* hp_data_out, int16_t* lp_data_out) {
@@ -138,23 +138,23 @@ static void SplitFilter(const int16_t* data_in, size_t data_length,
}
}
// Calculates the energy of |data_in| in dB, and also updates an overall
// |total_energy| if necessary.
// Calculates the energy of `data_in` in dB, and also updates an overall
// `total_energy` if necessary.
//
// - data_in [i] : Input audio data for energy calculation.
// - data_length [i] : Length of input data.
// - offset [i] : Offset value added to |log_energy|.
// - offset [i] : Offset value added to `log_energy`.
// - total_energy [i/o] : An external energy updated with the energy of
// |data_in|.
// NOTE: |total_energy| is only updated if
// |total_energy| <= |kMinEnergy|.
// - log_energy [o] : 10 * log10("energy of |data_in|") given in Q4.
// `data_in`.
// NOTE: `total_energy` is only updated if
// `total_energy` <= `kMinEnergy`.
// - log_energy [o] : 10 * log10("energy of `data_in`") given in Q4.
static void LogOfEnergy(const int16_t* data_in, size_t data_length,
int16_t offset, int16_t* total_energy,
int16_t* log_energy) {
// |tot_rshifts| accumulates the number of right shifts performed on |energy|.
// `tot_rshifts` accumulates the number of right shifts performed on `energy`.
int tot_rshifts = 0;
// The |energy| will be normalized to 15 bits. We use unsigned integer because
// The `energy` will be normalized to 15 bits. We use unsigned integer because
// we eventually will mask out the fractional part.
uint32_t energy = 0;
@@ -169,14 +169,14 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
// zeros of an unsigned 32 bit value.
int normalizing_rshifts = 17 - WebRtcSpl_NormU32(energy);
// In a 15 bit representation the leading bit is 2^14. log2(2^14) in Q10 is
// (14 << 10), which is what we initialize |log2_energy| with. For a more
// (14 << 10), which is what we initialize `log2_energy` with. For a more
// detailed derivations, see below.
int16_t log2_energy = kLogEnergyIntPart;
tot_rshifts += normalizing_rshifts;
// Normalize |energy| to 15 bits.
// |tot_rshifts| is now the total number of right shifts performed on
// |energy| after normalization. This means that |energy| is in
// Normalize `energy` to 15 bits.
// `tot_rshifts` is now the total number of right shifts performed on
// `energy` after normalization. This means that `energy` is in
// Q(-tot_rshifts).
if (normalizing_rshifts < 0) {
energy <<= -normalizing_rshifts;
@@ -184,30 +184,30 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
energy >>= normalizing_rshifts;
}
// Calculate the energy of |data_in| in dB, in Q4.
// Calculate the energy of `data_in` in dB, in Q4.
//
// 10 * log10("true energy") in Q4 = 2^4 * 10 * log10("true energy") =
// 160 * log10(|energy| * 2^|tot_rshifts|) =
// 160 * log10(2) * log2(|energy| * 2^|tot_rshifts|) =
// 160 * log10(2) * (log2(|energy|) + log2(2^|tot_rshifts|)) =
// (160 * log10(2)) * (log2(|energy|) + |tot_rshifts|) =
// |kLogConst| * (|log2_energy| + |tot_rshifts|)
// 160 * log10(`energy` * 2^`tot_rshifts`) =
// 160 * log10(2) * log2(`energy` * 2^`tot_rshifts`) =
// 160 * log10(2) * (log2(`energy`) + log2(2^`tot_rshifts`)) =
// (160 * log10(2)) * (log2(`energy`) + `tot_rshifts`) =
// `kLogConst` * (`log2_energy` + `tot_rshifts`)
//
// We know by construction that |energy| is normalized to 15 bits. Hence,
// |energy| = 2^14 + frac_Q15, where frac_Q15 is a fractional part in Q15.
// Further, we'd like |log2_energy| in Q10
// log2(|energy|) in Q10 = 2^10 * log2(2^14 + frac_Q15) =
// We know by construction that `energy` is normalized to 15 bits. Hence,
// `energy` = 2^14 + frac_Q15, where frac_Q15 is a fractional part in Q15.
// Further, we'd like `log2_energy` in Q10
// log2(`energy`) in Q10 = 2^10 * log2(2^14 + frac_Q15) =
// 2^10 * log2(2^14 * (1 + frac_Q15 * 2^-14)) =
// 2^10 * (14 + log2(1 + frac_Q15 * 2^-14)) ~=
// (14 << 10) + 2^10 * (frac_Q15 * 2^-14) =
// (14 << 10) + (frac_Q15 * 2^-4) = (14 << 10) + (frac_Q15 >> 4)
//
// Note that frac_Q15 = (|energy| & 0x00003FFF)
// Note that frac_Q15 = (`energy` & 0x00003FFF)
// Calculate and add the fractional part to |log2_energy|.
// Calculate and add the fractional part to `log2_energy`.
log2_energy += (int16_t) ((energy & 0x00003FFF) >> 4);
// |kLogConst| is in Q9, |log2_energy| in Q10 and |tot_rshifts| in Q0.
// `kLogConst` is in Q9, `log2_energy` in Q10 and `tot_rshifts` in Q0.
// Note that we in our derivation above have accounted for an output in Q4.
*log_energy = (int16_t)(((kLogConst * log2_energy) >> 19) +
((tot_rshifts * kLogConst) >> 9));
@@ -222,19 +222,19 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
*log_energy += offset;
// Update the approximate |total_energy| with the energy of |data_in|, if
// |total_energy| has not exceeded |kMinEnergy|. |total_energy| is used as an
// Update the approximate `total_energy` with the energy of `data_in`, if
// `total_energy` has not exceeded `kMinEnergy`. `total_energy` is used as an
// energy indicator in WebRtcVad_GmmProbability() in vad_core.c.
if (*total_energy <= kMinEnergy) {
if (tot_rshifts >= 0) {
// We know by construction that the |energy| > |kMinEnergy| in Q0, so add
// an arbitrary value such that |total_energy| exceeds |kMinEnergy|.
// We know by construction that the `energy` > `kMinEnergy` in Q0, so add
// an arbitrary value such that `total_energy` exceeds `kMinEnergy`.
*total_energy += kMinEnergy + 1;
} else {
// By construction |energy| is represented by 15 bits, hence any number of
// right shifted |energy| will fit in an int16_t. In addition, adding the
// value to |total_energy| is wrap around safe as long as
// |kMinEnergy| < 8192.
// By construction `energy` is represented by 15 bits, hence any number of
// right shifted `energy` will fit in an int16_t. In addition, adding the
// value to `total_energy` is wrap around safe as long as
// `kMinEnergy` < 8192.
*total_energy += (int16_t) (energy >> -tot_rshifts); // Q0.
}
}
@@ -243,14 +243,14 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
size_t data_length, int16_t* features) {
int16_t total_energy = 0;
// We expect |data_length| to be 80, 160 or 240 samples, which corresponds to
// We expect `data_length` to be 80, 160 or 240 samples, which corresponds to
// 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
// have at most 120 samples after the first split and at most 60 samples after
// the second split.
int16_t hp_120[120], lp_120[120];
int16_t hp_60[60], lp_60[60];
const size_t half_data_length = data_length >> 1;
size_t length = half_data_length; // |data_length| / 2, corresponds to
size_t length = half_data_length; // `data_length` / 2, corresponds to
// bandwidth = 2000 Hz after downsampling.
// Initialize variables for the first SplitFilter().
@@ -260,7 +260,7 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
int16_t* lp_out_ptr = lp_120; // [0 - 2000] Hz.
RTC_DCHECK_LE(data_length, 240);
RTC_DCHECK_LT(4, kNumChannels - 1); // Checking maximum |frequency_band|.
RTC_DCHECK_LT(4, kNumChannels - 1); // Checking maximum `frequency_band`.
// Split at 2000 Hz and downsample.
SplitFilter(in_ptr, data_length, &self->upper_state[frequency_band],
@@ -275,7 +275,7 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 3000 Hz - 4000 Hz.
length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
length >>= 1; // `data_length` / 4 <=> bandwidth = 1000 Hz.
LogOfEnergy(hp_60, length, kOffsetVector[5], &total_energy, &features[5]);
@@ -287,12 +287,12 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
in_ptr = lp_120; // [0 - 2000] Hz.
hp_out_ptr = hp_60; // [1000 - 2000] Hz.
lp_out_ptr = lp_60; // [0 - 1000] Hz.
length = half_data_length; // |data_length| / 2 <=> bandwidth = 2000 Hz.
length = half_data_length; // `data_length` / 2 <=> bandwidth = 2000 Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 1000 Hz - 2000 Hz.
length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
length >>= 1; // `data_length` / 4 <=> bandwidth = 1000 Hz.
LogOfEnergy(hp_60, length, kOffsetVector[3], &total_energy, &features[3]);
// For the lower band (0 Hz - 1000 Hz) split at 500 Hz and downsample.
@@ -304,7 +304,7 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 500 Hz - 1000 Hz.
length >>= 1; // |data_length| / 8 <=> bandwidth = 500 Hz.
length >>= 1; // `data_length` / 8 <=> bandwidth = 500 Hz.
LogOfEnergy(hp_120, length, kOffsetVector[2], &total_energy, &features[2]);
// For the lower band (0 Hz - 500 Hz) split at 250 Hz and downsample.
@@ -316,7 +316,7 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 250 Hz - 500 Hz.
length >>= 1; // |data_length| / 16 <=> bandwidth = 250 Hz.
length >>= 1; // `data_length` / 16 <=> bandwidth = 250 Hz.
LogOfEnergy(hp_60, length, kOffsetVector[1], &total_energy, &features[1]);
// Remove 0 Hz - 80 Hz, by high pass filtering the lower band.

8
third_party/libwebrtc/common_audio/vad/vad_filterbank.h vendored
View File

@@ -17,8 +17,8 @@
#include "common_audio/vad/vad_core.h"
// Takes |data_length| samples of |data_in| and calculates the logarithm of the
// energy of each of the |kNumChannels| = 6 frequency bands used by the VAD:
// Takes `data_length` samples of `data_in` and calculates the logarithm of the
// energy of each of the `kNumChannels` = 6 frequency bands used by the VAD:
// 80 Hz - 250 Hz
// 250 Hz - 500 Hz
// 500 Hz - 1000 Hz
@@ -26,10 +26,10 @@
// 2000 Hz - 3000 Hz
// 3000 Hz - 4000 Hz
//
// The values are given in Q4 and written to |features|. Further, an approximate
// The values are given in Q4 and written to `features`. Further, an approximate
// overall energy is returned. The return value is used in
// WebRtcVad_GmmProbability() as a signal indicator, hence it is arbitrary above
// the threshold |kMinEnergy|.
// the threshold `kMinEnergy`.
//
// - self [i/o] : State information of the VAD.
// - data_in [i] : Input audio data, for feature extraction.

28
third_party/libwebrtc/common_audio/vad/vad_gmm.c vendored
View File

@@ -15,16 +15,16 @@
static const int32_t kCompVar = 22005;
static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12.
// For a normal distribution, the probability of |input| is calculated and
// For a normal distribution, the probability of `input` is calculated and
// returned (in Q20). The formula for normal distributed probability is
//
// 1 / s * exp(-(x - m)^2 / (2 * s^2))
//
// where the parameters are given in the following Q domains:
// m = |mean| (Q7)
// s = |std| (Q7)
// x = |input| (Q4)
// in addition to the probability we output |delta| (in Q11) used when updating
// m = `mean` (Q7)
// s = `std` (Q7)
// x = `input` (Q4)
// in addition to the probability we output `delta` (in Q11) used when updating
// the noise/speech model.
int32_t WebRtcVad_GaussianProbability(int16_t input,
int16_t mean,
@@ -33,13 +33,13 @@ int32_t WebRtcVad_GaussianProbability(int16_t input,
int16_t tmp16, inv_std, inv_std2, exp_value = 0;
int32_t tmp32;
// Calculate |inv_std| = 1 / s, in Q10.
// 131072 = 1 in Q17, and (|std| >> 1) is for rounding instead of truncation.
// Calculate `inv_std` = 1 / s, in Q10.
// 131072 = 1 in Q17, and (`std` >> 1) is for rounding instead of truncation.
// Q-domain: Q17 / Q7 = Q10.
tmp32 = (int32_t) 131072 + (int32_t) (std >> 1);
inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std);
// Calculate |inv_std2| = 1 / s^2, in Q14.
// Calculate `inv_std2` = 1 / s^2, in Q14.
tmp16 = (inv_std >> 2); // Q10 -> Q8.
// Q-domain: (Q8 * Q8) >> 2 = Q14.
inv_std2 = (int16_t)((tmp16 * tmp16) >> 2);
@@ -51,20 +51,20 @@ int32_t WebRtcVad_GaussianProbability(int16_t input,
tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
// To be used later, when updating noise/speech model.
// |delta| = (x - m) / s^2, in Q11.
// `delta` = (x - m) / s^2, in Q11.
// Q-domain: (Q14 * Q7) >> 10 = Q11.
*delta = (int16_t)((inv_std2 * tmp16) >> 10);
// Calculate the exponent |tmp32| = (x - m)^2 / (2 * s^2), in Q10. Replacing
// Calculate the exponent `tmp32` = (x - m)^2 / (2 * s^2), in Q10. Replacing
// division by two with one shift.
// Q-domain: (Q11 * Q7) >> 8 = Q10.
tmp32 = (*delta * tmp16) >> 9;
// If the exponent is small enough to give a non-zero probability we calculate
// |exp_value| ~= exp(-(x - m)^2 / (2 * s^2))
// ~= exp2(-log2(exp(1)) * |tmp32|).
// `exp_value` ~= exp(-(x - m)^2 / (2 * s^2))
// ~= exp2(-log2(exp(1)) * `tmp32`).
if (tmp32 < kCompVar) {
// Calculate |tmp16| = log2(exp(1)) * |tmp32|, in Q10.
// Calculate `tmp16` = log2(exp(1)) * `tmp32`, in Q10.
// Q-domain: (Q12 * Q10) >> 12 = Q10.
tmp16 = (int16_t)((kLog2Exp * tmp32) >> 12);
tmp16 = -tmp16;
@@ -72,7 +72,7 @@ int32_t WebRtcVad_GaussianProbability(int16_t input,
tmp16 ^= 0xFFFF;
tmp16 >>= 10;
tmp16 += 1;
// Get |exp_value| = exp(-|tmp32|) in Q10.
// Get `exp_value` = exp(-`tmp32`) in Q10.
exp_value >>= tmp16;
}

10
third_party/libwebrtc/common_audio/vad/vad_gmm.h vendored
View File

@@ -15,8 +15,8 @@
#include <stdint.h>
// Calculates the probability for |input|, given that |input| comes from a
// normal distribution with mean and standard deviation (|mean|, |std|).
// Calculates the probability for `input`, given that `input` comes from a
// normal distribution with mean and standard deviation (`mean`, `std`).
//
// Inputs:
// - input : input sample in Q4.
@@ -26,11 +26,11 @@
// Output:
//
// - delta : input used when updating the model, Q11.
// |delta| = (|input| - |mean|) / |std|^2.
// `delta` = (`input` - `mean`) / `std`^2.
//
// Return:
// (probability for |input|) =
// 1 / |std| * exp(-(|input| - |mean|)^2 / (2 * |std|^2));
// (probability for `input`) =
// 1 / `std` * exp(-(`input` - `mean`)^2 / (2 * `std`^2));
int32_t WebRtcVad_GaussianProbability(int16_t input,
int16_t mean,
int16_t std,

14
third_party/libwebrtc/common_audio/vad/vad_sp.c vendored
View File

@@ -52,7 +52,7 @@ void WebRtcVad_Downsampling(const int16_t* signal_in,
filter_state[1] = tmp32_2;
}
// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
// Inserts `feature_value` into `low_value_vector`, if it is one of the 16
// smallest values the last 100 frames. Then calculates and returns the median
// of the five smallest values.
int16_t WebRtcVad_FindMinimum(VadInstT* self,
@@ -66,13 +66,13 @@ int16_t WebRtcVad_FindMinimum(VadInstT* self,
int16_t alpha = 0;
int32_t tmp32 = 0;
// Pointer to memory for the 16 minimum values and the age of each value of
// the |channel|.
// the `channel`.
int16_t* age = &self->index_vector[offset];
int16_t* smallest_values = &self->low_value_vector[offset];
RTC_DCHECK_LT(channel, kNumChannels);
// Each value in |smallest_values| is getting 1 loop older. Update |age|, and
// Each value in `smallest_values` is getting 1 loop older. Update `age`, and
// remove old values.
for (i = 0; i < 16; i++) {
if (age[i] != 100) {
@@ -88,9 +88,9 @@ int16_t WebRtcVad_FindMinimum(VadInstT* self,
}
}
// Check if |feature_value| is smaller than any of the values in
// |smallest_values|. If so, find the |position| where to insert the new value
// (|feature_value|).
// Check if `feature_value` is smaller than any of the values in
// `smallest_values`. If so, find the `position` where to insert the new value
// (`feature_value`).
if (feature_value < smallest_values[7]) {
if (feature_value < smallest_values[3]) {
if (feature_value < smallest_values[1]) {
@@ -152,7 +152,7 @@ int16_t WebRtcVad_FindMinimum(VadInstT* self,
age[position] = 1;
}
// Get |current_median|.
// Get `current_median`.
if (self->frame_counter > 2) {
current_median = smallest_values[2];
} else if (self->frame_counter > 0) {

4
third_party/libwebrtc/common_audio/vad/vad_sp.h vendored
View File

@@ -23,11 +23,11 @@
//
// Input & Output:
// - filter_state : Current filter states of the two all-pass filters. The
// |filter_state| is updated after all samples have been
// `filter_state` is updated after all samples have been
// processed.
//
// Output:
// - signal_out : Downsampled signal (of length |in_length| / 2).
// - signal_out : Downsampled signal (of length `in_length` / 2).
void WebRtcVad_Downsampling(const int16_t* signal_in,
int16_t* signal_out,
int32_t* filter_state,

2
third_party/libwebrtc/common_audio/vad/vad_sp_unittest.cc vendored
View File

@@ -29,7 +29,7 @@ TEST_F(VadTest, vad_sp) {
int16_t data_in[kMaxFrameLenSp];
int16_t data_out[kMaxFrameLenSp];
// We expect the first value to be 1600 as long as |frame_counter| is zero,
// We expect the first value to be 1600 as long as `frame_counter` is zero,
// which is true for the first iteration.
static const int16_t kReferenceMin[32] = {
1600, 720, 509, 512, 532, 552, 570, 588, 606, 624, 642,

2
third_party/libwebrtc/common_audio/wav_header.cc vendored
View File

@@ -161,7 +161,7 @@ uint16_t BlockAlign(size_t num_channels, size_t bytes_per_sample) {
return static_cast<uint16_t>(num_channels * bytes_per_sample);
}
// Finds a chunk having the sought ID. If found, then |readable| points to the
// Finds a chunk having the sought ID. If found, then `readable` points to the
// first byte of the sought chunk data. If not found, the end of the file is
// reached.
bool FindWaveChunk(ChunkHeader* chunk_header,