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Unified Diff: webrtc/modules/audio_processing/vad/vad_audio_proc.cc

Issue 1181933002: Pull the Voice Activity Detector out from the AGC (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Remove unused files from isolate Created 5 years, 6 months ago
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Index: webrtc/modules/audio_processing/vad/vad_audio_proc.cc
diff --git a/webrtc/modules/audio_processing/agc/agc_audio_proc.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
similarity index 78%
rename from webrtc/modules/audio_processing/agc/agc_audio_proc.cc
rename to webrtc/modules/audio_processing/vad/vad_audio_proc.cc
index dc4a5a711c7ef168e7e534190e03c6384dbd0cd8..e8f27f802de5468ca97e36e3e06ea82ef6d7cf44 100644
--- a/webrtc/modules/audio_processing/agc/agc_audio_proc.cc
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h"
#include <math.h>
#include <stdio.h>
#include "webrtc/common_audio/fft4g.h"
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc_internal.h"
-#include "webrtc/modules/audio_processing/agc/pitch_internal.h"
-#include "webrtc/modules/audio_processing/agc/pole_zero_filter.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h"
+#include "webrtc/modules/audio_processing/vad/pitch_internal.h"
+#include "webrtc/modules/audio_processing/vad/pole_zero_filter.h"
extern "C" {
#include "webrtc/modules/audio_coding/codecs/isac/main/source/codec.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h"
@@ -29,23 +29,25 @@ namespace webrtc {
// The following structures are declared anonymous in iSAC's structs.h. To
// forward declare them, we use this derived class trick.
-struct AgcAudioProc::PitchAnalysisStruct : public ::PitchAnalysisStruct {};
-struct AgcAudioProc::PreFiltBankstr : public ::PreFiltBankstr {};
+struct VadAudioProc::PitchAnalysisStruct : public ::PitchAnalysisStruct {};
+struct VadAudioProc::PreFiltBankstr : public ::PreFiltBankstr {};
-static const float kFrequencyResolution = kSampleRateHz /
- static_cast<float>(AgcAudioProc::kDftSize);
+static const float kFrequencyResolution =
+ kSampleRateHz / static_cast<float>(VadAudioProc::kDftSize);
static const int kSilenceRms = 5;
-// TODO(turajs): Make a Create or Init for AgcAudioProc.
-AgcAudioProc::AgcAudioProc()
+// TODO(turajs): Make a Create or Init for VadAudioProc.
+VadAudioProc::VadAudioProc()
: audio_buffer_(),
num_buffer_samples_(kNumPastSignalSamples),
log_old_gain_(-2),
old_lag_(50), // Arbitrary but valid as pitch-lag (in samples).
pitch_analysis_handle_(new PitchAnalysisStruct),
pre_filter_handle_(new PreFiltBankstr),
- high_pass_filter_(PoleZeroFilter::Create(
- kCoeffNumerator, kFilterOrder, kCoeffDenominator, kFilterOrder)) {
+ high_pass_filter_(PoleZeroFilter::Create(kCoeffNumerator,
+ kFilterOrder,
+ kCoeffDenominator,
+ kFilterOrder)) {
static_assert(kNumPastSignalSamples + kNumSubframeSamples ==
sizeof(kLpcAnalWin) / sizeof(kLpcAnalWin[0]),
"lpc analysis window incorrect size");
@@ -64,15 +66,16 @@ AgcAudioProc::AgcAudioProc()
WebRtcIsac_InitPitchAnalysis(pitch_analysis_handle_.get());
}
-AgcAudioProc::~AgcAudioProc() {}
+VadAudioProc::~VadAudioProc() {
+}
-void AgcAudioProc::ResetBuffer() {
+void VadAudioProc::ResetBuffer() {
memcpy(audio_buffer_, &audio_buffer_[kNumSamplesToProcess],
sizeof(audio_buffer_[0]) * kNumPastSignalSamples);
num_buffer_samples_ = kNumPastSignalSamples;
}
-int AgcAudioProc::ExtractFeatures(const int16_t* frame,
+int VadAudioProc::ExtractFeatures(const int16_t* frame,
int length,
AudioFeatures* features) {
features->num_frames = 0;
@@ -85,7 +88,7 @@ int AgcAudioProc::ExtractFeatures(const int16_t* frame,
// classification.
if (high_pass_filter_->Filter(frame, kNumSubframeSamples,
&audio_buffer_[num_buffer_samples_]) != 0) {
- return -1;
+ return -1;
}
num_buffer_samples_ += kNumSubframeSamples;
@@ -115,7 +118,8 @@ int AgcAudioProc::ExtractFeatures(const int16_t* frame,
}
// Computes |kLpcOrder + 1| correlation coefficients.
-void AgcAudioProc::SubframeCorrelation(double* corr, int length_corr,
+void VadAudioProc::SubframeCorrelation(double* corr,
+ int length_corr,
int subframe_index) {
assert(length_corr >= kLpcOrder + 1);
double windowed_audio[kNumSubframeSamples + kNumPastSignalSamples];
@@ -124,20 +128,20 @@ void AgcAudioProc::SubframeCorrelation(double* corr, int length_corr,
for (int n = 0; n < kNumSubframeSamples + kNumPastSignalSamples; n++)
windowed_audio[n] = audio_buffer_[buffer_index++] * kLpcAnalWin[n];
- WebRtcIsac_AutoCorr(corr, windowed_audio, kNumSubframeSamples +
- kNumPastSignalSamples, kLpcOrder);
+ WebRtcIsac_AutoCorr(corr, windowed_audio,
+ kNumSubframeSamples + kNumPastSignalSamples, kLpcOrder);
}
// Compute |kNum10msSubframes| sets of LPC coefficients, one per 10 ms input.
// The analysis window is 15 ms long and it is centered on the first half of
// each 10ms sub-frame. This is equivalent to computing LPC coefficients for the
// first half of each 10 ms subframe.
-void AgcAudioProc::GetLpcPolynomials(double* lpc, int length_lpc) {
+void VadAudioProc::GetLpcPolynomials(double* lpc, int length_lpc) {
assert(length_lpc >= kNum10msSubframes * (kLpcOrder + 1));
double corr[kLpcOrder + 1];
double reflec_coeff[kLpcOrder];
for (int i = 0, offset_lpc = 0; i < kNum10msSubframes;
- i++, offset_lpc += kLpcOrder + 1) {
+ i++, offset_lpc += kLpcOrder + 1) {
SubframeCorrelation(corr, kLpcOrder + 1, i);
corr[0] *= 1.0001;
// This makes Lev-Durb a bit more stable.
@@ -150,7 +154,8 @@ void AgcAudioProc::GetLpcPolynomials(double* lpc, int length_lpc) {
// Fit a second order curve to these 3 points and find the location of the
// extremum. The points are inverted before curve fitting.
-static float QuadraticInterpolation(float prev_val, float curr_val,
+static float QuadraticInterpolation(float prev_val,
+ float curr_val,
float next_val) {
// Doing the interpolation in |1 / A(z)|^2.
float fractional_index = 0;
@@ -158,8 +163,8 @@ static float QuadraticInterpolation(float prev_val, float curr_val,
prev_val = 1.0f / prev_val;
curr_val = 1.0f / curr_val;
- fractional_index = -(next_val - prev_val) * 0.5f / (next_val + prev_val -
- 2.f * curr_val);
+ fractional_index =
+ -(next_val - prev_val) * 0.5f / (next_val + prev_val - 2.f * curr_val);
assert(fabs(fractional_index) < 1);
return fractional_index;
}
@@ -169,7 +174,7 @@ static float QuadraticInterpolation(float prev_val, float curr_val,
// with the local minimum of A(z). It saves complexity, as we save one
// inversion. Furthermore, we find the first local maximum of magnitude squared,
// to save on one square root.
-void AgcAudioProc::FindFirstSpectralPeaks(double* f_peak, int length_f_peak) {
+void VadAudioProc::FindFirstSpectralPeaks(double* f_peak, int length_f_peak) {
assert(length_f_peak >= kNum10msSubframes);
double lpc[kNum10msSubframes * (kLpcOrder + 1)];
// For all sub-frames.
@@ -193,8 +198,8 @@ void AgcAudioProc::FindFirstSpectralPeaks(double* f_peak, int length_f_peak) {
float next_magn_sqr;
bool found_peak = false;
for (int n = 2; n < kNumDftCoefficients - 1; n++) {
- next_magn_sqr = data[2 * n] * data[2 * n] +
- data[2 * n + 1] * data[2 * n + 1];
+ next_magn_sqr =
+ data[2 * n] * data[2 * n] + data[2 * n + 1] * data[2 * n + 1];
if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
found_peak = true;
index_peak = n - 1;
@@ -213,15 +218,16 @@ void AgcAudioProc::FindFirstSpectralPeaks(double* f_peak, int length_f_peak) {
} else {
// A peak is found, do a simple quadratic interpolation to get a more
// accurate estimate of the peak location.
- fractional_index = QuadraticInterpolation(prev_magn_sqr, curr_magn_sqr,
- next_magn_sqr);
+ fractional_index =
+ QuadraticInterpolation(prev_magn_sqr, curr_magn_sqr, next_magn_sqr);
}
f_peak[i] = (index_peak + fractional_index) * kFrequencyResolution;
}
}
// Using iSAC functions to estimate pitch gains & lags.
-void AgcAudioProc::PitchAnalysis(double* log_pitch_gains, double* pitch_lags_hz,
+void VadAudioProc::PitchAnalysis(double* log_pitch_gains,
+ double* pitch_lags_hz,
int length) {
// TODO(turajs): This can be "imported" from iSAC & and the next two
// constants.
@@ -241,28 +247,27 @@ void AgcAudioProc::PitchAnalysis(double* log_pitch_gains, double* pitch_lags_hz,
kNumLookaheadSamples];
// Split signal to lower and upper bands
- WebRtcIsac_SplitAndFilterFloat(&audio_buffer_[kNumPastSignalSamples],
- lower, upper, lower_lookahead, upper_lookahead,
+ WebRtcIsac_SplitAndFilterFloat(&audio_buffer_[kNumPastSignalSamples], lower,
+ upper, lower_lookahead, upper_lookahead,
pre_filter_handle_.get());
WebRtcIsac_PitchAnalysis(lower_lookahead, lower_lookahead_pre_filter,
pitch_analysis_handle_.get(), lags, gains);
// Lags are computed on lower-band signal with sampling rate half of the
// input signal.
- GetSubframesPitchParameters(kSampleRateHz / 2, gains, lags,
- kNumPitchSubframes, kNum10msSubframes,
- &log_old_gain_, &old_lag_,
- log_pitch_gains, pitch_lags_hz);
+ GetSubframesPitchParameters(
+ kSampleRateHz / 2, gains, lags, kNumPitchSubframes, kNum10msSubframes,
+ &log_old_gain_, &old_lag_, log_pitch_gains, pitch_lags_hz);
}
-void AgcAudioProc::Rms(double* rms, int length_rms) {
+void VadAudioProc::Rms(double* rms, int length_rms) {
assert(length_rms >= kNum10msSubframes);
int offset = kNumPastSignalSamples;
for (int i = 0; i < kNum10msSubframes; i++) {
rms[i] = 0;
for (int n = 0; n < kNumSubframeSamples; n++, offset++)
rms[i] += audio_buffer_[offset] * audio_buffer_[offset];
- rms[i] = sqrt(rms[i] / kNumSubframeSamples);
+ rms[i] = sqrt(rms[i] / kNumSubframeSamples);
}
}
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