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Format the rest of C files in the repo

Browse Source 
Formatting done via:
git ls-files | grep -E '.*\.c$' | grep -Ev '^common_audio/signal_processing.*\.c$' | xargs clang-format -i

No-Iwyu: Includes didn't change and it isn't related to formatting
Bug: webrtc:42225392
Change-Id: Id78af8e3eceada9995e53b6a0fdc1a8cb5ffd1f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373907
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43699}
This commit is contained in:
Boris Tsirkin 2025-01-09 02:19:34 -08:00 committed by WebRTC LUCI CQ
parent 7eb83a3a18
commit c940dba16a
11 changed files with 588 additions and 545 deletions
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43
common_audio/ring_buffer.c
View File

@ -28,10 +28,9 @@ static size_t GetBufferReadRegions(RingBuffer* buf,
size_t* data_ptr_bytes_1,
void** data_ptr_2,
size_t* data_ptr_bytes_2) {
const size_t readable_elements = WebRtc_available_read(buf);
const size_t read_elements = (readable_elements < element_count ?
readable_elements : element_count);
const size_t read_elements =
(readable_elements < element_count ? readable_elements : element_count);
const size_t margin = buf->element_count - buf->read_pos;
// Check to see if read is not contiguous.
@ -99,7 +98,6 @@ size_t WebRtc_ReadBuffer(RingBuffer* self,
void** data_ptr,
void* data,
size_t element_count) {
if (self == NULL) {
return 0;
}
@ -112,17 +110,14 @@ size_t WebRtc_ReadBuffer(RingBuffer* self,
void* buf_ptr_2 = NULL;
size_t buf_ptr_bytes_1 = 0;
size_t buf_ptr_bytes_2 = 0;
const size_t read_count = GetBufferReadRegions(self,
element_count,
&buf_ptr_1,
&buf_ptr_bytes_1,
&buf_ptr_2,
&buf_ptr_bytes_2);
const size_t read_count =
GetBufferReadRegions(self, element_count, &buf_ptr_1, &buf_ptr_bytes_1,
&buf_ptr_2, &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.
memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
memcpy(((char*) data) + buf_ptr_bytes_1, buf_ptr_2, buf_ptr_bytes_2);
memcpy(((char*)data) + buf_ptr_bytes_1, buf_ptr_2, buf_ptr_bytes_2);
buf_ptr_1 = data;
} else if (!data_ptr) {
// No wrap, but a memcpy was requested.
@ -134,7 +129,7 @@ size_t WebRtc_ReadBuffer(RingBuffer* self,
}
// Update read position
WebRtc_MoveReadPtr(self, (int) read_count);
WebRtc_MoveReadPtr(self, (int)read_count);
return read_count;
}
@ -152,21 +147,21 @@ size_t WebRtc_WriteBuffer(RingBuffer* self,
{
const size_t free_elements = WebRtc_available_write(self);
const size_t write_elements = (free_elements < element_count ? free_elements
: element_count);
const size_t write_elements =
(free_elements < element_count ? free_elements : element_count);
size_t n = write_elements;
const size_t margin = self->element_count - self->write_pos;
if (write_elements > margin) {
// Buffer wrap around when writing.
memcpy(self->data + self->write_pos * self->element_size,
data, margin * self->element_size);
memcpy(self->data + self->write_pos * self->element_size, data,
margin * self->element_size);
self->write_pos = 0;
n -= margin;
self->rw_wrap = DIFF_WRAP;
}
memcpy(self->data + self->write_pos * self->element_size,
((const char*) data) + ((write_elements - n) * self->element_size),
((const char*)data) + ((write_elements - n) * self->element_size),
n * self->element_size);
self->write_pos += n;
@ -182,9 +177,9 @@ int WebRtc_MoveReadPtr(RingBuffer* self, int element_count) {
{
// We need to be able to take care of negative changes, hence use "int"
// instead of "size_t".
const int free_elements = (int) WebRtc_available_write(self);
const int readable_elements = (int) WebRtc_available_read(self);
int read_pos = (int) self->read_pos;
const int free_elements = (int)WebRtc_available_write(self);
const int readable_elements = (int)WebRtc_available_read(self);
int read_pos = (int)self->read_pos;
if (element_count > readable_elements) {
element_count = readable_elements;
@ -194,18 +189,18 @@ int WebRtc_MoveReadPtr(RingBuffer* self, int element_count) {
}
read_pos += element_count;
if (read_pos > (int) self->element_count) {
if (read_pos > (int)self->element_count) {
// Buffer wrap around. Restart read position and wrap indicator.
read_pos -= (int) self->element_count;
read_pos -= (int)self->element_count;
self->rw_wrap = SAME_WRAP;
}
if (read_pos < 0) {
// Buffer wrap around. Restart read position and wrap indicator.
read_pos += (int) self->element_count;
read_pos += (int)self->element_count;
self->rw_wrap = DIFF_WRAP;
}
self->read_pos = (size_t) read_pos;
self->read_pos = (size_t)read_pos;
return element_count;
}

200
common_audio/vad/vad_core.c
View File

@ -10,48 +10,48 @@
#include "common_audio/vad/vad_core.h"
#include "rtc_base/sanitizer.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "common_audio/vad/vad_filterbank.h"
#include "common_audio/vad/vad_gmm.h"
#include "common_audio/vad/vad_sp.h"
#include "rtc_base/sanitizer.h"
// Spectrum Weighting
static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };
static const int16_t kSpectrumWeight[kNumChannels] = {6, 8, 10, 12, 14, 16};
static const int16_t kNoiseUpdateConst = 655; // Q15
static const int16_t kSpeechUpdateConst = 6554; // Q15
static const int16_t kBackEta = 154; // Q8
// Minimum difference between the two models, Q5
static const int16_t kMinimumDifference[kNumChannels] = {
544, 544, 576, 576, 576, 576 };
static const int16_t kMinimumDifference[kNumChannels] = {544, 544, 576,
576, 576, 576};
// Upper limit of mean value for speech model, Q7
static const int16_t kMaximumSpeech[kNumChannels] = {
11392, 11392, 11520, 11520, 11520, 11520 };
static const int16_t kMaximumSpeech[kNumChannels] = {11392, 11392, 11520,
11520, 11520, 11520};
// Minimum value for mean value
static const int16_t kMinimumMean[kNumGaussians] = { 640, 768 };
static const int16_t kMinimumMean[kNumGaussians] = {640, 768};
// Upper limit of mean value for noise model, Q7
static const int16_t kMaximumNoise[kNumChannels] = {
9216, 9088, 8960, 8832, 8704, 8576 };
static const int16_t kMaximumNoise[kNumChannels] = {9216, 9088, 8960,
8832, 8704, 8576};
// Start values for the Gaussian models, Q7
// Weights for the two Gaussians for the six channels (noise)
static const int16_t kNoiseDataWeights[kTableSize] = {
34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
static const int16_t kNoiseDataWeights[kTableSize] = {34, 62, 72, 66, 53, 25,
94, 66, 56, 62, 75, 103};
// Weights for the two Gaussians for the six channels (speech)
static const int16_t kSpeechDataWeights[kTableSize] = {
48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
static const int16_t kSpeechDataWeights[kTableSize] = {48, 82, 45, 87, 50, 47,
80, 46, 83, 41, 78, 81};
// Means for the two Gaussians for the six channels (noise)
static const int16_t kNoiseDataMeans[kTableSize] = {
6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362};
// Means for the two Gaussians for the six channels (speech)
static const int16_t kSpeechDataMeans[kTableSize] = {
8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
};
static const int16_t kSpeechDataMeans[kTableSize] = {8306, 10085, 10078, 11823,
11843, 6309, 9473, 9571,
10879, 7581, 8180, 7483};
// Stds for the two Gaussians for the six channels (noise)
static const int16_t kNoiseDataStds[kTableSize] = {
378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455};
// Stds for the two Gaussians for the six channels (speech)
static const int16_t kSpeechDataStds[kTableSize] = {
555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850};
// Constants used in GmmProbability().
//
@ -70,25 +70,25 @@ static const int kInitCheck = 42;
// Thresholds for different frame lengths (10 ms, 20 ms and 30 ms).
//
// Mode 0, Quality.
static const int16_t kOverHangMax1Q[3] = { 8, 4, 3 };
static const int16_t kOverHangMax2Q[3] = { 14, 7, 5 };
static const int16_t kLocalThresholdQ[3] = { 24, 21, 24 };
static const int16_t kGlobalThresholdQ[3] = { 57, 48, 57 };
static const int16_t kOverHangMax1Q[3] = {8, 4, 3};
static const int16_t kOverHangMax2Q[3] = {14, 7, 5};
static const int16_t kLocalThresholdQ[3] = {24, 21, 24};
static const int16_t kGlobalThresholdQ[3] = {57, 48, 57};
// Mode 1, Low bitrate.
static const int16_t kOverHangMax1LBR[3] = { 8, 4, 3 };
static const int16_t kOverHangMax2LBR[3] = { 14, 7, 5 };
static const int16_t kLocalThresholdLBR[3] = { 37, 32, 37 };
static const int16_t kGlobalThresholdLBR[3] = { 100, 80, 100 };
static const int16_t kOverHangMax1LBR[3] = {8, 4, 3};
static const int16_t kOverHangMax2LBR[3] = {14, 7, 5};
static const int16_t kLocalThresholdLBR[3] = {37, 32, 37};
static const int16_t kGlobalThresholdLBR[3] = {100, 80, 100};
// Mode 2, Aggressive.
static const int16_t kOverHangMax1AGG[3] = { 6, 3, 2 };
static const int16_t kOverHangMax2AGG[3] = { 9, 5, 3 };
static const int16_t kLocalThresholdAGG[3] = { 82, 78, 82 };
static const int16_t kGlobalThresholdAGG[3] = { 285, 260, 285 };
static const int16_t kOverHangMax1AGG[3] = {6, 3, 2};
static const int16_t kOverHangMax2AGG[3] = {9, 5, 3};
static const int16_t kLocalThresholdAGG[3] = {82, 78, 82};
static const int16_t kGlobalThresholdAGG[3] = {285, 260, 285};
// Mode 3, Very aggressive.
static const int16_t kOverHangMax1VAG[3] = { 6, 3, 2 };
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 };
static const int16_t kOverHangMax1VAG[3] = {6, 3, 2};
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.
@ -98,7 +98,8 @@ static const int16_t kGlobalThresholdVAG[3] = { 1100, 1050, 1100 };
// - weights [i] : Weights used for averaging.
//
// returns : The weighted average.
static int32_t WeightedAverage(int16_t* data, int16_t offset,
static int32_t WeightedAverage(int16_t* data,
int16_t offset,
const int16_t* weights) {
int k;
int32_t weighted_average = 0;
@ -130,8 +131,10 @@ static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
// - frame_length [i] : Number of input samples
//
// - returns : the VAD decision (0 - noise, 1 - speech).
static int16_t GmmProbability(VadInstT* self, int16_t* features,
int16_t total_power, size_t frame_length) {
static int16_t GmmProbability(VadInstT* self,
int16_t* features,
int16_t total_power,
size_t frame_length) {
int channel, k;
int16_t feature_minimum;
int16_t h0, h1;
@ -145,8 +148,8 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
int16_t delt, ndelt;
int16_t maxspe, maxmu;
int16_t deltaN[kTableSize], deltaS[kTableSize];
int16_t ngprvec[kTableSize] = { 0 }; // Conditional probability = 0.
int16_t sgprvec[kTableSize] = { 0 }; // Conditional probability = 0.
int16_t ngprvec[kTableSize] = {0}; // Conditional probability = 0.
int16_t sgprvec[kTableSize] = {0}; // Conditional probability = 0.
int32_t h0_test, h1_test;
int32_t tmp1_s32, tmp2_s32;
int32_t sum_log_likelihood_ratios = 0;
@ -194,19 +197,17 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
gaussian = channel + k * kNumChannels;
// Probability under H0, that is, probability of frame being noise.
// Value given in Q27 = Q7 * Q20.
tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
self->noise_means[gaussian],
self->noise_stds[gaussian],
&deltaN[gaussian]);
tmp1_s32 = WebRtcVad_GaussianProbability(
features[channel], self->noise_means[gaussian],
self->noise_stds[gaussian], &deltaN[gaussian]);
noise_probability[k] = kNoiseDataWeights[gaussian] * tmp1_s32;
h0_test += noise_probability[k]; // Q27
// Probability under H1, that is, probability of frame being speech.
// Value given in Q27 = Q7 * Q20.
tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
self->speech_means[gaussian],
self->speech_stds[gaussian],
&deltaS[gaussian]);
tmp1_s32 = WebRtcVad_GaussianProbability(
features[channel], self->speech_means[gaussian],
self->speech_stds[gaussian], &deltaS[gaussian]);
speech_probability[k] = kSpeechDataWeights[gaussian] * tmp1_s32;
h1_test += speech_probability[k]; // Q27
}
@ -237,7 +238,7 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// 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]);
(int32_t)(log_likelihood_ratio * kSpectrumWeight[channel]);
// Local VAD decision.
if ((log_likelihood_ratio * 4) > individualTest) {
@ -247,12 +248,12 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// TODO(bjornv): The conditional probabilities below are applied on the
// hard coded number of Gaussians set to two. Find a way to generalize.
// Calculate local noise probabilities used later when updating the GMM.
h0 = (int16_t) (h0_test >> 12); // Q15
h0 = (int16_t)(h0_test >> 12); // Q15
if (h0 > 0) {
// High probability of noise. Assign conditional probabilities for each
// Gaussian in the GMM.
tmp1_s32 = (noise_probability[0] & 0xFFFFF000) << 2; // Q29
ngprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h0); // Q14
ngprvec[channel] = (int16_t)WebRtcSpl_DivW32W16(tmp1_s32, h0); // Q14
ngprvec[channel + kNumChannels] = 16384 - ngprvec[channel];
} else {
// Low noise probability. Assign conditional probability 1 to the first
@ -261,12 +262,12 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
}
// Calculate local speech probabilities used later when updating the GMM.
h1 = (int16_t) (h1_test >> 12); // Q15
h1 = (int16_t)(h1_test >> 12); // Q15
if (h1 > 0) {
// High probability of speech. Assign conditional probabilities for each
// Gaussian in the GMM. Otherwise use the initialized values, i.e., 0.
tmp1_s32 = (speech_probability[0] & 0xFFFFF000) << 2; // Q29
sgprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h1); // Q14
sgprvec[channel] = (int16_t)WebRtcSpl_DivW32W16(tmp1_s32, h1); // Q14
sgprvec[channel + kNumChannels] = 16384 - sgprvec[channel];
}
}
@ -277,14 +278,13 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// Update the model parameters.
maxspe = 12800;
for (channel = 0; channel < kNumChannels; channel++) {
// Get minimum value in past which is used for long term correction in Q4.
feature_minimum = WebRtcVad_FindMinimum(self, features[channel], channel);
// Compute the "global" mean, that is the sum of the two means weighted.
noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
&kNoiseDataWeights[channel]);
tmp1_s16 = (int16_t) (noise_global_mean >> 6); // Q8
tmp1_s16 = (int16_t)(noise_global_mean >> 6); // Q8
for (k = 0; k < kNumGaussians; k++) {
gaussian = channel + k * kNumChannels;
@ -314,11 +314,11 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
nmk3 = nmk2 + (int16_t)((ndelt * kBackEta) >> 9);
// Control that the noise mean does not drift to much.
tmp_s16 = (int16_t) ((k + 5) << 7);
tmp_s16 = (int16_t)((k + 5) << 7);
if (nmk3 < tmp_s16) {
nmk3 = tmp_s16;
}
tmp_s16 = (int16_t) ((72 + k - channel) << 7);
tmp_s16 = (int16_t)((72 + k - channel) << 7);
if (nmk3 > tmp_s16) {
nmk3 = tmp_s16;
}
@ -362,9 +362,9 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// 0.1 * Q20 / Q7 = Q13.
if (tmp2_s32 > 0) {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp2_s32, ssk * 10);
tmp_s16 = (int16_t)WebRtcSpl_DivW32W16(tmp2_s32, ssk * 10);
} else {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp2_s32, ssk * 10);
tmp_s16 = (int16_t)WebRtcSpl_DivW32W16(-tmp2_s32, ssk * 10);
tmp_s16 = -tmp_s16;
}
// Divide by 4 giving an update factor of 0.025 (= 0.1 / 4).
@ -394,9 +394,9 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// Q20 / Q7 = Q13.
if (tmp1_s32 > 0) {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, nsk);
tmp_s16 = (int16_t)WebRtcSpl_DivW32W16(tmp1_s32, nsk);
} else {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp1_s32, nsk);
tmp_s16 = (int16_t)WebRtcSpl_DivW32W16(-tmp1_s32, nsk);
tmp_s16 = -tmp_s16;
}
tmp_s16 += 32; // Rounding
@ -419,8 +419,8 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// `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);
diff = (int16_t)(speech_global_mean >> 9) -
(int16_t)(noise_global_mean >> 9);
if (diff < kMinimumDifference[channel]) {
tmp_s16 = kMinimumDifference[channel] - diff;
@ -432,21 +432,21 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
// 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,
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
// changed after the call.
noise_global_mean = WeightedAverage(&self->noise_means[channel],
-tmp2_s16,
noise_global_mean =
WeightedAverage(&self->noise_means[channel], -tmp2_s16,
&kNoiseDataWeights[channel]);
}
// Control that the speech & noise means do not drift to much.
maxspe = kMaximumSpeech[channel];
tmp2_s16 = (int16_t) (speech_global_mean >> 7);
tmp2_s16 = (int16_t)(speech_global_mean >> 7);
if (tmp2_s16 > maxspe) {
// Upper limit of speech model.
tmp2_s16 -= maxspe;
@ -456,7 +456,7 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
}
}
tmp2_s16 = (int16_t) (noise_global_mean >> 7);
tmp2_s16 = (int16_t)(noise_global_mean >> 7);
if (tmp2_s16 > kMaximumNoise[channel]) {
tmp2_s16 -= kMaximumNoise[channel];
@ -555,10 +555,8 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2Q,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdQ,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdQ,
sizeof(self->total));
memcpy(self->individual, kLocalThresholdQ, sizeof(self->individual));
memcpy(self->total, kGlobalThresholdQ, sizeof(self->total));
break;
case 1:
// Low bitrate mode.
@ -566,10 +564,8 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2LBR,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdLBR,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdLBR,
sizeof(self->total));
memcpy(self->individual, kLocalThresholdLBR, sizeof(self->individual));
memcpy(self->total, kGlobalThresholdLBR, sizeof(self->total));
break;
case 2:
// Aggressive mode.
@ -577,10 +573,8 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2AGG,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdAGG,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdAGG,
sizeof(self->total));
memcpy(self->individual, kLocalThresholdAGG, sizeof(self->individual));
memcpy(self->total, kGlobalThresholdAGG, sizeof(self->total));
break;
case 3:
// Very aggressive mode.
@ -588,10 +582,8 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2VAG,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdVAG,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdVAG,
sizeof(self->total));
memcpy(self->individual, kLocalThresholdVAG, sizeof(self->individual));
memcpy(self->total, kGlobalThresholdVAG, sizeof(self->total));
break;
default:
return_value = -1;
@ -604,14 +596,15 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
// Calculate VAD decision by first extracting feature values and then calculate
// probability for both speech and background noise.
int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
int WebRtcVad_CalcVad48khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length) {
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
// frame length in 10 ms (480 samples) + 256 extra.
int32_t tmp_mem[480 + 256] = { 0 };
int32_t tmp_mem[480 + 256] = {0};
const size_t kFrameLen10ms48khz = 480;
const size_t kFrameLen10ms8khz = 80;
size_t num_10ms_frames = frame_length / kFrameLen10ms48khz;
@ -619,8 +612,7 @@ int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
for (i = 0; i < num_10ms_frames; i++) {
WebRtcSpl_Resample48khzTo8khz(speech_frame,
&speech_nb[i * kFrameLen10ms8khz],
&inst->state_48_to_8,
tmp_mem);
&inst->state_48_to_8, tmp_mem);
}
// Do VAD on an 8 kHz signal
@ -629,21 +621,21 @@ int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
return vad;
}
int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
size_t frame_length)
{
int WebRtcVad_CalcVad32khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length) {
size_t len;
int vad;
int16_t speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
// Downsample signal 32->16->8 before doing VAD
WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
frame_length);
WebRtcVad_Downsampling(speech_frame, speechWB,
&(inst->downsampling_filter_states[2]), frame_length);
len = frame_length / 2;
WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states,
len);
len /= 2;
// Do VAD on an 8 kHz signal
@ -652,16 +644,16 @@ int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
return vad;
}
int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
size_t frame_length)
{
int WebRtcVad_CalcVad16khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length) {
size_t len;
int vad;
int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
// Wideband: Downsample signal before doing VAD
WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
frame_length);
WebRtcVad_Downsampling(speech_frame, speechNB,
inst->downsampling_filter_states, frame_length);
len = frame_length / 2;
vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
@ -669,9 +661,9 @@ int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
return vad;
}
int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
size_t frame_length)
{
int WebRtcVad_CalcVad8khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length) {
int16_t feature_vector[kNumChannels], total_power;
// Get power in the bands

60
common_audio/vad/vad_filterbank.c
View File

@ -10,23 +10,23 @@
#include "common_audio/vad/vad_filterbank.h"
#include "rtc_base/checks.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/checks.h"
// Constants used in LogOfEnergy().
static const int16_t kLogConst = 24660; // 160*log10(2) in Q9.
static const int16_t kLogEnergyIntPart = 14336; // 14 in Q10
// Coefficients used by HighPassFilter, Q14.
static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
static const int16_t kHpZeroCoefs[3] = {6631, -13262, 6631};
static const int16_t kHpPoleCoefs[3] = {16384, -7756, 5620};
// Allpass filter coefficients, upper and lower, in Q15.
// Upper: 0.64, Lower: 0.17
static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
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 };
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
// sampled at 500 Hz.
@ -36,14 +36,15 @@ static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
// - filter_state [i/o] : State of the filter.
// - data_out [o] : Output audio data in the frequency interval
// 80 - 250 Hz.
static void HighPassFilter(const int16_t* data_in, size_t data_length,
int16_t* filter_state, int16_t* data_out) {
static void HighPassFilter(const int16_t* data_in,
size_t data_length,
int16_t* filter_state,
int16_t* data_out) {
size_t i;
const int16_t* in_ptr = data_in;
int16_t* out_ptr = data_out;
int32_t tmp32 = 0;
// The sum of the absolute values of the impulse response:
// The zero/pole-filter has a max amplification of a single sample of: 1.4546
// Impulse response: 0.4047 -0.6179 -0.0266 0.1993 0.1035 -0.0194
@ -64,7 +65,7 @@ static void HighPassFilter(const int16_t* data_in, size_t data_length,
tmp32 -= kHpPoleCoefs[1] * filter_state[2];
tmp32 -= kHpPoleCoefs[2] * filter_state[3];
filter_state[3] = filter_state[2];
filter_state[2] = (int16_t) (tmp32 >> 14);
filter_state[2] = (int16_t)(tmp32 >> 14);
*out_ptr++ = filter_state[2];
}
}
@ -78,8 +79,10 @@ static void HighPassFilter(const int16_t* data_in, size_t data_length,
// - filter_coefficient [i] : Given in Q15.
// - filter_state [i/o] : State of the filter given in Q(-1).
// - data_out [o] : Output audio signal given in Q(-1).
static void AllPassFilter(const int16_t* data_in, size_t data_length,
int16_t filter_coefficient, int16_t* filter_state,
static void AllPassFilter(const int16_t* data_in,
size_t data_length,
int16_t filter_coefficient,
int16_t* filter_state,
int16_t* data_out) {
// The filter can only cause overflow (in the w16 output variable)
// if more than 4 consecutive input numbers are of maximum value and
@ -90,18 +93,18 @@ static void AllPassFilter(const int16_t* data_in, size_t data_length,
size_t i;
int16_t tmp16 = 0;
int32_t tmp32 = 0;
int32_t state32 = ((int32_t) (*filter_state) * (1 << 16)); // Q15
int32_t state32 = ((int32_t)(*filter_state) * (1 << 16)); // Q15
for (i = 0; i < data_length; i++) {
tmp32 = state32 + filter_coefficient * *data_in;
tmp16 = (int16_t) (tmp32 >> 16); // Q(-1)
tmp16 = (int16_t)(tmp32 >> 16); // Q(-1)
*data_out++ = tmp16;
state32 = (*data_in * (1 << 14)) - filter_coefficient * tmp16; // Q14
state32 *= 2; // Q15.
data_in += 2;
}
*filter_state = (int16_t) (state32 >> 16); // Q(-1)
*filter_state = (int16_t)(state32 >> 16); // Q(-1)
}
// Splits `data_in` into `hp_data_out` and `lp_data_out` corresponding to
@ -115,9 +118,12 @@ static void AllPassFilter(const int16_t* data_in, size_t data_length,
// 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.
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) {
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) {
size_t i;
size_t half_length = data_length >> 1; // Downsampling by 2.
int16_t tmp_out;
@ -149,8 +155,10 @@ static void SplitFilter(const int16_t* data_in, size_t data_length,
// 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,
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`.
int tot_rshifts = 0;
@ -161,8 +169,8 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
RTC_DCHECK(data_in);
RTC_DCHECK_GT(data_length, 0);
energy = (uint32_t) WebRtcSpl_Energy((int16_t*) data_in, data_length,
&tot_rshifts);
energy =
(uint32_t)WebRtcSpl_Energy((int16_t*)data_in, data_length, &tot_rshifts);
if (energy != 0) {
// By construction, normalizing to 15 bits is equivalent with 17 leading
@ -205,7 +213,7 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
// Note that frac_Q15 = (`energy` & 0x00003FFF)
// Calculate and add the fractional part to `log2_energy`.
log2_energy += (int16_t) ((energy & 0x00003FFF) >> 4);
log2_energy += (int16_t)((energy & 0x00003FFF) >> 4);
// `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.
@ -235,13 +243,15 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
// 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.
*total_energy += (int16_t)(energy >> -tot_rshifts); // Q0.
}
}
}
int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
size_t data_length, int16_t* features) {
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
// 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will

4
common_audio/vad/vad_gmm.c
View File

@ -36,8 +36,8 @@ int32_t WebRtcVad_GaussianProbability(int16_t input,
// 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);
tmp32 = (int32_t)131072 + (int32_t)(std >> 1);
inv_std = (int16_t)WebRtcSpl_DivW32W16(tmp32, std);
// Calculate `inv_std2` = 1 / s^2, in Q14.
tmp16 = (inv_std >> 2); // Q10 -> Q8.

14
common_audio/vad/vad_sp.c
View File

@ -10,13 +10,13 @@
#include "common_audio/vad/vad_sp.h"
#include "rtc_base/checks.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "common_audio/vad/vad_core.h"
#include "rtc_base/checks.h"
// Allpass filter coefficients, upper and lower, in Q13.
// Upper: 0.64, Lower: 0.17.
static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 }; // Q13.
static const int16_t kAllPassCoefsQ13[2] = {5243, 1392}; // Q13.
static const int16_t kSmoothingDown = 6553; // 0.2 in Q15.
static const int16_t kSmoothingUp = 32439; // 0.99 in Q15.
@ -36,14 +36,14 @@ void WebRtcVad_Downsampling(const int16_t* signal_in,
// Filter coefficients in Q13, filter state in Q0.
for (n = 0; n < half_length; n++) {
// All-pass filtering upper branch.
tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
((kAllPassCoefsQ13[0] * *signal_in) >> 14));
tmp16_1 =
(int16_t)((tmp32_1 >> 1) + ((kAllPassCoefsQ13[0] * *signal_in) >> 14));
*signal_out = tmp16_1;
tmp32_1 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[0] * tmp16_1) >> 12);
// All-pass filtering lower branch.
tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
((kAllPassCoefsQ13[1] * *signal_in) >> 14));
tmp16_2 =
(int16_t)((tmp32_2 >> 1) + ((kAllPassCoefsQ13[1] * *signal_in) >> 14));
*signal_out++ += tmp16_2;
tmp32_2 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[1] * tmp16_2) >> 12);
}
@ -170,7 +170,7 @@ int16_t WebRtcVad_FindMinimum(VadInstT* self,
tmp32 = (alpha + 1) * self->mean_value[channel];
tmp32 += (WEBRTC_SPL_WORD16_MAX - alpha) * current_median;
tmp32 += 16384;
self->mean_value[channel] = (int16_t) (tmp32 >> 15);
self->mean_value[channel] = (int16_t)(tmp32 >> 15);
return self->mean_value[channel];
}

12
common_audio/vad/webrtc_vad.c
View File

@ -17,7 +17,7 @@
#include "common_audio/vad/vad_core.h"
static const int kInitCheck = 42;
static const int kValidRates[] = { 8000, 16000, 32000, 48000 };
static const int kValidRates[] = {8000, 16000, 32000, 48000};
static const size_t kRatesSize = sizeof(kValidRates) / sizeof(*kValidRates);
static const int kMaxFrameLengthMs = 30;
@ -36,12 +36,12 @@ void WebRtcVad_Free(VadInst* handle) {
// TODO(bjornv): Move WebRtcVad_InitCore() code here.
int WebRtcVad_Init(VadInst* handle) {
// Initialize the core VAD component.
return WebRtcVad_InitCore((VadInstT*) handle);
return WebRtcVad_InitCore((VadInstT*)handle);
}
// TODO(bjornv): Move WebRtcVad_set_mode_core() code here.
int WebRtcVad_set_mode(VadInst* handle, int mode) {
VadInstT* self = (VadInstT*) handle;
VadInstT* self = (VadInstT*)handle;
if (handle == NULL) {
return -1;
@ -53,10 +53,12 @@ int WebRtcVad_set_mode(VadInst* handle, int mode) {
return WebRtcVad_set_mode_core(self, mode);
}
int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
int WebRtcVad_Process(VadInst* handle,
int fs,
const int16_t* audio_frame,
size_t frame_length) {
int vad = -1;
VadInstT* self = (VadInstT*) handle;
VadInstT* self = (VadInstT*)handle;
if (handle == NULL) {
return -1;

3
modules/audio_coding/codecs/g711/g711_interface.c
View File

@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/g711/g711_interface.h"
#include <string.h>
#include "modules/third_party/g711/g711.h"
#include "modules/audio_coding/codecs/g711/g711_interface.h"
size_t WebRtcG711_EncodeA(const int16_t* speechIn,
size_t len,

78
modules/audio_coding/codecs/g722/g722_interface.c
View File

@ -8,28 +8,27 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/g722/g722_interface.h"
#include <stdlib.h>
#include <string.h>
#include "modules/audio_coding/codecs/g722/g722_interface.h"
#include "modules/third_party/g722/g722_enc_dec.h"
int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
{
*G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
if (*G722enc_inst!=NULL) {
return(0);
int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst) {
*G722enc_inst = (G722EncInst*)malloc(sizeof(G722EncoderState));
if (*G722enc_inst != NULL) {
return (0);
} else {
return(-1);
return (-1);
}
}
int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
{
int16_t WebRtcG722_EncoderInit(G722EncInst* G722enc_inst) {
// Create and/or reset the G.722 encoder
// Bitrate 64 kbps and wideband mode (2)
G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
(G722EncoderState*) G722enc_inst, 64000, 2);
G722enc_inst = (G722EncInst*)WebRtc_g722_encode_init(
(G722EncoderState*)G722enc_inst, 64000, 2);
if (G722enc_inst == NULL) {
return -1;
} else {
@ -37,30 +36,27 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
}
}
int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
{
int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst) {
// Free encoder memory
return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
return WebRtc_g722_encode_release((G722EncoderState*)G722enc_inst);
}
size_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
const int16_t* speechIn,
size_t len,
uint8_t* encoded)
{
unsigned char *codechar = (unsigned char*) encoded;
uint8_t* encoded) {
unsigned char* codechar = (unsigned char*)encoded;
// Encode the input speech vector
return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
speechIn, len);
return WebRtc_g722_encode((G722EncoderState*)G722enc_inst, codechar, speechIn,
len);
}
int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
{
*G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
if (*G722dec_inst!=NULL) {
return(0);
int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst) {
*G722dec_inst = (G722DecInst*)malloc(sizeof(G722DecoderState));
if (*G722dec_inst != NULL) {
return (0);
} else {
return(-1);
return (-1);
}
}
@ -70,35 +66,29 @@ void WebRtcG722_DecoderInit(G722DecInst* inst) {
WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
}
int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
{
int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst) {
// Free encoder memory
return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
return WebRtc_g722_decode_release((G722DecoderState*)G722dec_inst);
}
size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
const uint8_t *encoded,
size_t WebRtcG722_Decode(G722DecInst* G722dec_inst,
const uint8_t* encoded,
size_t len,
int16_t *decoded,
int16_t *speechType)
{
int16_t* decoded,
int16_t* speechType) {
// Decode the G.722 encoder stream
*speechType=G722_WEBRTC_SPEECH;
return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
encoded, len);
*speechType = G722_WEBRTC_SPEECH;
return WebRtc_g722_decode((G722DecoderState*)G722dec_inst, decoded, encoded,
len);
}
int16_t WebRtcG722_Version(char *versionStr, short len)
{
int16_t WebRtcG722_Version(char* versionStr, short len) {
// Get version string
char version[30] = "2.0.0\n";
if (strlen(version) < (unsigned int)len)
{
if (strlen(version) < (unsigned int)len) {
strcpy(versionStr, version);
return 0;
}
else
{
} else {
return -1;
}
}

114
modules/audio_coding/codecs/isac/main/source/filter_functions.c
View File

@ -14,8 +14,8 @@
#include <stdlib.h>
#endif
#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
static void WebRtcIsac_AllPoleFilter(double* InOut,
double* Coef,
@ -27,26 +27,21 @@ static void WebRtcIsac_AllPoleFilter(double* InOut,
size_t n;
int k;
//if (fabs(Coef[0]-1.0)<0.001) {
if ( (Coef[0] > 0.9999) && (Coef[0] < 1.0001) )
{
for(n = 0; n < lengthInOut; n++)
{
// if (fabs(Coef[0]-1.0)<0.001) {
if ((Coef[0] > 0.9999) && (Coef[0] < 1.0001)) {
for (n = 0; n < lengthInOut; n++) {
sum = Coef[1] * InOut[-1];
for(k = 2; k <= orderCoef; k++){
for (k = 2; k <= orderCoef; k++) {
sum += Coef[k] * InOut[-k];
}
*InOut++ -= sum;
}
}
else
{
} else {
scal = 1.0 / Coef[0];
for(n=0;n<lengthInOut;n++)
{
for (n = 0; n < lengthInOut; n++) {
*InOut *= scal;
for(k=1;k<=orderCoef;k++){
*InOut -= scal*Coef[k]*InOut[-k];
for (k = 1; k <= orderCoef; k++) {
*InOut -= scal * Coef[k] * InOut[-k];
}
InOut++;
}
@ -64,11 +59,10 @@ static void WebRtcIsac_AllZeroFilter(double* In,
int k;
double tmp;
for(n = 0; n < lengthInOut; n++)
{
for (n = 0; n < lengthInOut; n++) {
tmp = In[0] * Coef[0];
for(k = 1; k <= orderCoef; k++){
for (k = 1; k <= orderCoef; k++) {
tmp += Coef[k] * In[-k];
}
@ -83,21 +77,21 @@ static void WebRtcIsac_ZeroPoleFilter(double* In,
size_t lengthInOut,
int orderCoef,
double* Out) {
/* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
/* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
/* the state of the zero section is assumed to be in In[-1] to In[-orderCoef]
*/
/* the state of the pole section is assumed to be in Out[-1] to
* Out[-orderCoef] */
WebRtcIsac_AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
WebRtcIsac_AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
WebRtcIsac_AllZeroFilter(In, ZeroCoef, lengthInOut, orderCoef, Out);
WebRtcIsac_AllPoleFilter(Out, PoleCoef, lengthInOut, orderCoef);
}
void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order) {
size_t lag, n;
double sum, prod;
const double *x_lag;
const double* x_lag;
for (lag = 0; lag <= order; lag++)
{
for (lag = 0; lag <= order; lag++) {
sum = 0.0f;
x_lag = &x[lag];
prod = x[0] * x_lag[0];
@ -108,7 +102,6 @@ void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order) {
sum += prod;
r[lag] = sum;
}
}
static void WebRtcIsac_BwExpand(double* out,
@ -132,64 +125,67 @@ void WebRtcIsac_WeightingFilter(const double* in,
double* whiout,
WeightFiltstr* wfdata) {
double tmpbuffer[PITCH_FRAME_LEN + PITCH_WLPCBUFLEN];
double corr[PITCH_WLPCORDER+1], rc[PITCH_WLPCORDER+1];
double apol[PITCH_WLPCORDER+1], apolr[PITCH_WLPCORDER+1];
double rho=0.9, *inp, *dp, *dp2;
double corr[PITCH_WLPCORDER + 1], rc[PITCH_WLPCORDER + 1];
double apol[PITCH_WLPCORDER + 1], apolr[PITCH_WLPCORDER + 1];
double rho = 0.9, *inp, *dp, *dp2;
double whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
double weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
double *weo, *who, opol[PITCH_WLPCORDER+1], ext[PITCH_WLPCWINLEN];
double *weo, *who, opol[PITCH_WLPCORDER + 1], ext[PITCH_WLPCWINLEN];
int k, n, endpos, start;
/* Set up buffer and states */
memcpy(tmpbuffer, wfdata->buffer, sizeof(double) * PITCH_WLPCBUFLEN);
memcpy(tmpbuffer+PITCH_WLPCBUFLEN, in, sizeof(double) * PITCH_FRAME_LEN);
memcpy(wfdata->buffer, tmpbuffer+PITCH_FRAME_LEN, sizeof(double) * PITCH_WLPCBUFLEN);
memcpy(tmpbuffer + PITCH_WLPCBUFLEN, in, sizeof(double) * PITCH_FRAME_LEN);
memcpy(wfdata->buffer, tmpbuffer + PITCH_FRAME_LEN,
sizeof(double) * PITCH_WLPCBUFLEN);
dp=weoutbuf;
dp2=whoutbuf;
for (k=0;k<PITCH_WLPCORDER;k++) {
dp = weoutbuf;
dp2 = whoutbuf;
for (k = 0; k < PITCH_WLPCORDER; k++) {
*dp++ = wfdata->weostate[k];
*dp2++ = wfdata->whostate[k];
opol[k]=0.0;
opol[k] = 0.0;
}
opol[0]=1.0;
opol[PITCH_WLPCORDER]=0.0;
weo=dp;
who=dp2;
opol[0] = 1.0;
opol[PITCH_WLPCORDER] = 0.0;
weo = dp;
who = dp2;
endpos=PITCH_WLPCBUFLEN + PITCH_SUBFRAME_LEN;
inp=tmpbuffer + PITCH_WLPCBUFLEN;
endpos = PITCH_WLPCBUFLEN + PITCH_SUBFRAME_LEN;
inp = tmpbuffer + PITCH_WLPCBUFLEN;
for (n=0; n<PITCH_SUBFRAMES; n++) {
for (n = 0; n < PITCH_SUBFRAMES; n++) {
/* Windowing */
start=endpos-PITCH_WLPCWINLEN;
for (k=0; k<PITCH_WLPCWINLEN; k++) {
ext[k]=wfdata->window[k]*tmpbuffer[start+k];
start = endpos - PITCH_WLPCWINLEN;
for (k = 0; k < PITCH_WLPCWINLEN; k++) {
ext[k] = wfdata->window[k] * tmpbuffer[start + k];
}
/* Get LPC polynomial */
WebRtcIsac_AutoCorr(corr, ext, PITCH_WLPCWINLEN, PITCH_WLPCORDER);
corr[0]=1.01*corr[0]+1.0; /* White noise correction */
corr[0] = 1.01 * corr[0] + 1.0; /* White noise correction */
WebRtcIsac_LevDurb(apol, rc, corr, PITCH_WLPCORDER);
WebRtcIsac_BwExpand(apolr, apol, rho, PITCH_WLPCORDER+1);
WebRtcIsac_BwExpand(apolr, apol, rho, PITCH_WLPCORDER + 1);
/* Filtering */
WebRtcIsac_ZeroPoleFilter(inp, apol, apolr, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, weo);
WebRtcIsac_ZeroPoleFilter(inp, apolr, opol, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, who);
WebRtcIsac_ZeroPoleFilter(inp, apol, apolr, PITCH_SUBFRAME_LEN,
PITCH_WLPCORDER, weo);
WebRtcIsac_ZeroPoleFilter(inp, apolr, opol, PITCH_SUBFRAME_LEN,
PITCH_WLPCORDER, who);
inp+=PITCH_SUBFRAME_LEN;
endpos+=PITCH_SUBFRAME_LEN;
weo+=PITCH_SUBFRAME_LEN;
who+=PITCH_SUBFRAME_LEN;
inp += PITCH_SUBFRAME_LEN;
endpos += PITCH_SUBFRAME_LEN;
weo += PITCH_SUBFRAME_LEN;
who += PITCH_SUBFRAME_LEN;
}
/* Export filter states */
for (k=0;k<PITCH_WLPCORDER;k++) {
wfdata->weostate[k]=weoutbuf[PITCH_FRAME_LEN+k];
wfdata->whostate[k]=whoutbuf[PITCH_FRAME_LEN+k];
for (k = 0; k < PITCH_WLPCORDER; k++) {
wfdata->weostate[k] = weoutbuf[PITCH_FRAME_LEN + k];
wfdata->whostate[k] = whoutbuf[PITCH_FRAME_LEN + k];
}
/* Export output data */
memcpy(weiout, weoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
memcpy(whiout, whoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
memcpy(weiout, weoutbuf + PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
memcpy(whiout, whoutbuf + PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
}

328
modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
View File

@ -21,12 +21,12 @@
#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h"
#include "rtc_base/system/ignore_warnings.h"
static const double kInterpolWin[8] = {-0.00067556028640, 0.02184247643159, -0.12203175715679, 0.60086484101160,
static const double kInterpolWin[8] = {
-0.00067556028640, 0.02184247643159, -0.12203175715679, 0.60086484101160,
0.60086484101160, -0.12203175715679, 0.02184247643159, -0.00067556028640};
/* interpolation filter */
__inline static void IntrepolFilter(double *data_ptr, double *intrp)
{
__inline static void IntrepolFilter(double* data_ptr, double* intrp) {
*intrp = kInterpolWin[0] * data_ptr[-3];
*intrp += kInterpolWin[1] * data_ptr[-2];
*intrp += kInterpolWin[2] * data_ptr[-1];
@ -37,16 +37,17 @@ __inline static void IntrepolFilter(double *data_ptr, double *intrp)
*intrp += kInterpolWin[7] * data_ptr[4];
}
/* 2D parabolic interpolation */
/* probably some 0.5 factors can be eliminated, and the square-roots can be removed from the Cholesky fact. */
__inline static void Intrpol2D(double T[3][3], double *x, double *y, double *peak_val)
{
/* probably some 0.5 factors can be eliminated, and the square-roots can be
* removed from the Cholesky fact. */
__inline static void Intrpol2D(double T[3][3],
double* x,
double* y,
double* peak_val) {
double c, b[2], A[2][2];
double t1, t2, d;
double delta1, delta2;
// double T[3][3] = {{-1.25, -.25,-.25}, {-.25, .75, .75}, {-.25, .75, .75}};
// should result in: delta1 = 0.5; delta2 = 0.0; peak_val = 1.0
@ -61,7 +62,7 @@ __inline static void Intrpol2D(double T[3][3], double *x, double *y, double *pea
A[1][1] = -t2 - 0.5 * d;
/* deal with singularities or ill-conditioned cases */
if ( (A[0][0] < 1e-7) || ((A[0][0] * A[1][1] - A[0][1] * A[0][1]) < 1e-7) ) {
if ((A[0][0] < 1e-7) || ((A[0][0] * A[1][1] - A[0][1] * A[0][1]) < 1e-7)) {
*peak_val = T[1][1];
return;
}
@ -91,17 +92,15 @@ __inline static void Intrpol2D(double T[3][3], double *x, double *y, double *pea
*y += delta2;
}
static void PCorr(const double *in, double *outcorr)
{
static void PCorr(const double* in, double* outcorr) {
double sum, ysum, prod;
const double *x, *inptr;
int k, n;
//ysum = 1e-6; /* use this with float (i.s.o. double)! */
// ysum = 1e-6; /* use this with float (i.s.o. double)! */
ysum = 1e-13;
sum = 0.0;
x = in + PITCH_MAX_LAG/2 + 2;
x = in + PITCH_MAX_LAG / 2 + 2;
for (n = 0; n < PITCH_CORR_LEN2; n++) {
ysum += in[n] * in[n];
sum += x[n] * in[n];
@ -111,7 +110,7 @@ static void PCorr(const double *in, double *outcorr)
*outcorr = sum / sqrt(ysum);
for (k = 1; k < PITCH_LAG_SPAN2; k++) {
ysum -= in[k-1] * in[k-1];
ysum -= in[k - 1] * in[k - 1];
ysum += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1];
sum = 0.0;
inptr = &in[k];
@ -176,15 +175,15 @@ static void WebRtcIsac_InitializePitch(const double* in,
const double old_gain,
PitchAnalysisStruct* State,
double* lags) {
double buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
double buf_dec[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 + 2];
double ratio, log_lag, gain_bias;
double bias;
double corrvec1[PITCH_LAG_SPAN2];
double corrvec2[PITCH_LAG_SPAN2];
int m, k;
// Allocating 10 extra entries at the begining of the CorrSurf
double corrSurfBuff[10 + (2*PITCH_BW+3)*(PITCH_LAG_SPAN2+4)];
double* CorrSurf[2*PITCH_BW+3];
double corrSurfBuff[10 + (2 * PITCH_BW + 3) * (PITCH_LAG_SPAN2 + 4)];
double* CorrSurf[2 * PITCH_BW + 3];
double *CorrSurfPtr1, *CorrSurfPtr2;
double LagWin[3] = {0.2, 0.5, 0.98};
int ind1, ind2, peaks_ind, peak, max_ind;
@ -198,30 +197,38 @@ static void WebRtcIsac_InitializePitch(const double* in,
double T[3][3];
int row;
for(k = 0; k < 2*PITCH_BW+3; k++)
{
CorrSurf[k] = &corrSurfBuff[10 + k * (PITCH_LAG_SPAN2+4)];
for (k = 0; k < 2 * PITCH_BW + 3; k++) {
CorrSurf[k] = &corrSurfBuff[10 + k * (PITCH_LAG_SPAN2 + 4)];
}
/* reset CorrSurf matrix */
memset(corrSurfBuff, 0, sizeof(double) * (10 + (2*PITCH_BW+3) * (PITCH_LAG_SPAN2+4)));
memset(corrSurfBuff, 0,
sizeof(double) * (10 + (2 * PITCH_BW + 3) * (PITCH_LAG_SPAN2 + 4)));
//warnings -DH
// warnings -DH
max_ind = 0;
peak = 0;
/* copy old values from state buffer */
memcpy(buf_dec, State->dec_buffer, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
memcpy(buf_dec, State->dec_buffer,
sizeof(double) * (PITCH_CORR_LEN2 + PITCH_CORR_STEP2 +
PITCH_MAX_LAG / 2 - PITCH_FRAME_LEN / 2 + 2));
/* decimation; put result after the old values */
WebRtcIsac_DecimateAllpass(in, State->decimator_state, PITCH_FRAME_LEN,
&buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
WebRtcIsac_DecimateAllpass(
in, State->decimator_state, PITCH_FRAME_LEN,
&buf_dec[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
PITCH_FRAME_LEN / 2 + 2]);
/* low-pass filtering */
for (k = PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2; k++)
buf_dec[k] += 0.75 * buf_dec[k-1] - 0.25 * buf_dec[k-2];
for (k = PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
PITCH_FRAME_LEN / 2 + 2;
k < PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 + 2; k++)
buf_dec[k] += 0.75 * buf_dec[k - 1] - 0.25 * buf_dec[k - 2];
/* copy end part back into state buffer */
memcpy(State->dec_buffer, buf_dec+PITCH_FRAME_LEN/2, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
memcpy(State->dec_buffer, buf_dec + PITCH_FRAME_LEN / 2,
sizeof(double) * (PITCH_CORR_LEN2 + PITCH_CORR_STEP2 +
PITCH_MAX_LAG / 2 - PITCH_FRAME_LEN / 2 + 2));
/* compute correlation for first and second half of the frame */
PCorr(buf_dec, corrvec1);
@ -230,10 +237,10 @@ static void WebRtcIsac_InitializePitch(const double* in,
/* bias towards pitch lag of previous frame */
log_lag = log(0.5 * old_lag);
gain_bias = 4.0 * old_gain * old_gain;
if (gain_bias > 0.8) gain_bias = 0.8;
for (k = 0; k < PITCH_LAG_SPAN2; k++)
{
ratio = log((double) (k + (PITCH_MIN_LAG/2-2))) - log_lag;
if (gain_bias > 0.8)
gain_bias = 0.8;
for (k = 0; k < PITCH_LAG_SPAN2; k++) {
ratio = log((double)(k + (PITCH_MIN_LAG / 2 - 2))) - log_lag;
bias = 1.0 + gain_bias * exp(-5.0 * ratio * ratio);
corrvec1[k] *= bias;
}
@ -243,8 +250,8 @@ static void WebRtcIsac_InitializePitch(const double* in,
gain_tmp = LagWin[k];
corrvec1[k] *= gain_tmp;
corrvec2[k] *= gain_tmp;
corrvec1[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
corrvec2[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
corrvec1[PITCH_LAG_SPAN2 - 1 - k] *= gain_tmp;
corrvec2[PITCH_LAG_SPAN2 - 1 - k] *= gain_tmp;
}
corr_max = 0.0;
@ -264,10 +271,11 @@ static void WebRtcIsac_InitializePitch(const double* in,
ind1 = 0;
ind2 = PITCH_BW;
CorrSurfPtr1 = &CorrSurf[0][2];
CorrSurfPtr2 = &CorrSurf[2*PITCH_BW][PITCH_BW+2];
for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW; k++) {
ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
adj = 0.2 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
CorrSurfPtr2 = &CorrSurf[2 * PITCH_BW][PITCH_BW + 2];
for (k = 0; k < PITCH_LAG_SPAN2 - PITCH_BW; k++) {
ratio = ((double)(ind1 + 12)) / ((double)(ind2 + 12));
adj = 0.2 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as
a function of ratio */
corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
CorrSurfPtr1[k] = corr;
if (corr > corr_max) {
@ -283,12 +291,13 @@ static void WebRtcIsac_InitializePitch(const double* in,
}
/* fill second and next to last rows of correlation surface */
ind1 = 0;
ind2 = PITCH_BW-1;
ind2 = PITCH_BW - 1;
CorrSurfPtr1 = &CorrSurf[1][2];
CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-1][PITCH_BW+1];
for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+1; k++) {
ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
adj = 0.9 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
CorrSurfPtr2 = &CorrSurf[2 * PITCH_BW - 1][PITCH_BW + 1];
for (k = 0; k < PITCH_LAG_SPAN2 - PITCH_BW + 1; k++) {
ratio = ((double)(ind1 + 12)) / ((double)(ind2 + 12));
adj = 0.9 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as
a function of ratio */
corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
CorrSurfPtr1[k] = corr;
if (corr > corr_max) {
@ -307,10 +316,11 @@ static void WebRtcIsac_InitializePitch(const double* in,
ind1 = 0;
ind2 = PITCH_BW - m; /* always larger than ind1 */
CorrSurfPtr1 = &CorrSurf[m][2];
CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-m][PITCH_BW+2-m];
for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+m; k++) {
ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
adj = ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
CorrSurfPtr2 = &CorrSurf[2 * PITCH_BW - m][PITCH_BW + 2 - m];
for (k = 0; k < PITCH_LAG_SPAN2 - PITCH_BW + m; k++) {
ratio = ((double)(ind1 + 12)) / ((double)(ind2 + 12));
adj = ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a
function of ratio */
corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
CorrSurfPtr1[k] = corr;
if (corr > corr_max) {
@ -331,33 +341,41 @@ static void WebRtcIsac_InitializePitch(const double* in,
peaks_ind = 0;
/* find peaks */
for (m = 1; m < PITCH_BW+1; m++) {
if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
for (m = 1; m < PITCH_BW + 1; m++) {
if (peaks_ind == PITCH_MAX_NUM_PEAKS)
break;
CorrSurfPtr1 = &CorrSurf[m][2];
for (k = 2; k < PITCH_LAG_SPAN2-PITCH_BW-2+m; k++) {
for (k = 2; k < PITCH_LAG_SPAN2 - PITCH_BW - 2 + m; k++) {
corr = CorrSurfPtr1[k];
if (corr > corr_max) {
if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
if ((corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2 + 5)]) &&
(corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2 + 4)])) {
if ((corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2 + 4)]) &&
(corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2 + 5)])) {
/* found a peak; store index into matrix */
peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
if (peaks_ind == PITCH_MAX_NUM_PEAKS)
break;
}
}
}
}
}
for (m = PITCH_BW+1; m < 2*PITCH_BW; m++) {
if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
for (m = PITCH_BW + 1; m < 2 * PITCH_BW; m++) {
if (peaks_ind == PITCH_MAX_NUM_PEAKS)
break;
CorrSurfPtr1 = &CorrSurf[m][2];
for (k = 2+m-PITCH_BW; k < PITCH_LAG_SPAN2-2; k++) {
for (k = 2 + m - PITCH_BW; k < PITCH_LAG_SPAN2 - 2; k++) {
corr = CorrSurfPtr1[k];
if (corr > corr_max) {
if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
if ((corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2 + 5)]) &&
(corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2 + 4)])) {
if ((corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2 + 4)]) &&
(corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2 + 5)])) {
/* found a peak; store index into matrix */
peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
if (peaks_ind == PITCH_MAX_NUM_PEAKS)
break;
}
}
}
@ -371,28 +389,32 @@ static void WebRtcIsac_InitializePitch(const double* in,
peak = peaks[k];
/* compute four interpolated values around current peak */
IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)], &intrp_a);
IntrepolFilter(&CorrSurfPtr1[peak - 1 ], &intrp_b);
IntrepolFilter(&CorrSurfPtr1[peak ], &intrp_c);
IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)], &intrp_d);
IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 5)], &intrp_a);
IntrepolFilter(&CorrSurfPtr1[peak - 1], &intrp_b);
IntrepolFilter(&CorrSurfPtr1[peak], &intrp_c);
IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 4)], &intrp_d);
/* determine maximum of the interpolated values */
corr = CorrSurfPtr1[peak];
corr_max = intrp_a;
if (intrp_b > corr_max) corr_max = intrp_b;
if (intrp_c > corr_max) corr_max = intrp_c;
if (intrp_d > corr_max) corr_max = intrp_d;
if (intrp_b > corr_max)
corr_max = intrp_b;
if (intrp_c > corr_max)
corr_max = intrp_c;
if (intrp_d > corr_max)
corr_max = intrp_d;
/* determine where the peak sits and fill a 3x3 matrix around it */
row = peak / (PITCH_LAG_SPAN2+4);
lags1[k] = (double) ((peak - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
lags2[k] = (double) (lags1[k] + PITCH_BW - row);
if ( corr > corr_max ) {
T[0][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
T[2][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
row = peak / (PITCH_LAG_SPAN2 + 4);
lags1[k] = (double)((peak - row * (PITCH_LAG_SPAN2 + 4)) +
PITCH_MIN_LAG / 2 - 4);
lags2[k] = (double)(lags1[k] + PITCH_BW - row);
if (corr > corr_max) {
T[0][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 5)];
T[2][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 4)];
T[1][1] = corr;
T[0][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
T[2][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
T[0][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 4)];
T[2][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 5)];
T[1][0] = intrp_a;
T[0][1] = intrp_b;
T[2][1] = intrp_c;
@ -401,51 +423,55 @@ static void WebRtcIsac_InitializePitch(const double* in,
if (intrp_a == corr_max) {
lags1[k] -= 0.5;
lags2[k] += 0.5;
IntrepolFilter(&CorrSurfPtr1[peak - 2*(PITCH_LAG_SPAN2+5)], &T[0][0]);
IntrepolFilter(&CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)], &T[2][0]);
IntrepolFilter(&CorrSurfPtr1[peak - 2 * (PITCH_LAG_SPAN2 + 5)],
&T[0][0]);
IntrepolFilter(&CorrSurfPtr1[peak - (2 * PITCH_LAG_SPAN2 + 9)],
&T[2][0]);
T[1][1] = intrp_a;
T[0][2] = intrp_b;
T[2][2] = intrp_c;
T[1][0] = CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)];
T[0][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
T[2][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
T[1][0] = CorrSurfPtr1[peak - (2 * PITCH_LAG_SPAN2 + 9)];
T[0][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 5)];
T[2][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 4)];
T[1][2] = corr;
} else if (intrp_b == corr_max) {
lags1[k] -= 0.5;
lags2[k] -= 0.5;
IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+6)], &T[0][0]);
IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 6)], &T[0][0]);
T[2][0] = intrp_a;
T[1][1] = intrp_b;
IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+3)], &T[0][2]);
IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 3)], &T[0][2]);
T[2][2] = intrp_d;
T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 5)];
T[0][1] = CorrSurfPtr1[peak - 1];
T[2][1] = corr;
T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 4)];
} else if (intrp_c == corr_max) {
lags1[k] += 0.5;
lags2[k] += 0.5;
T[0][0] = intrp_a;
IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)], &T[2][0]);
IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 4)], &T[2][0]);
T[1][1] = intrp_c;
T[0][2] = intrp_d;
IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)], &T[2][2]);
T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 5)], &T[2][2]);
T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2 + 4)];
T[0][1] = corr;
T[2][1] = CorrSurfPtr1[peak + 1];
T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 5)];
} else {
lags1[k] += 0.5;
lags2[k] -= 0.5;
T[0][0] = intrp_b;
T[2][0] = intrp_c;
T[1][1] = intrp_d;
IntrepolFilter(&CorrSurfPtr1[peak + 2*(PITCH_LAG_SPAN2+4)], &T[0][2]);
IntrepolFilter(&CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)], &T[2][2]);
IntrepolFilter(&CorrSurfPtr1[peak + 2 * (PITCH_LAG_SPAN2 + 4)],
&T[0][2]);
IntrepolFilter(&CorrSurfPtr1[peak + (2 * PITCH_LAG_SPAN2 + 9)],
&T[2][2]);
T[1][0] = corr;
T[0][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
T[2][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
T[1][2] = CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)];
T[0][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 4)];
T[2][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2 + 5)];
T[1][2] = CorrSurfPtr1[peak + (2 * PITCH_LAG_SPAN2 + 9)];
}
}
@ -466,27 +492,34 @@ static void WebRtcIsac_InitializePitch(const double* in,
lags1[peak] *= 2.0;
lags2[peak] *= 2.0;
if (lags1[peak] < (double) PITCH_MIN_LAG) lags1[peak] = (double) PITCH_MIN_LAG;
if (lags2[peak] < (double) PITCH_MIN_LAG) lags2[peak] = (double) PITCH_MIN_LAG;
if (lags1[peak] > (double) PITCH_MAX_LAG) lags1[peak] = (double) PITCH_MAX_LAG;
if (lags2[peak] > (double) PITCH_MAX_LAG) lags2[peak] = (double) PITCH_MAX_LAG;
if (lags1[peak] < (double)PITCH_MIN_LAG)
lags1[peak] = (double)PITCH_MIN_LAG;
if (lags2[peak] < (double)PITCH_MIN_LAG)
lags2[peak] = (double)PITCH_MIN_LAG;
if (lags1[peak] > (double)PITCH_MAX_LAG)
lags1[peak] = (double)PITCH_MAX_LAG;
if (lags2[peak] > (double)PITCH_MAX_LAG)
lags2[peak] = (double)PITCH_MAX_LAG;
/* store lags of highest peak in output array */
lags[0] = lags1[peak];
lags[1] = lags1[peak];
lags[2] = lags2[peak];
lags[3] = lags2[peak];
}
else
{
row = max_ind / (PITCH_LAG_SPAN2+4);
lags1[0] = (double) ((max_ind - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
lags2[0] = (double) (lags1[0] + PITCH_BW - row);
} else {
row = max_ind / (PITCH_LAG_SPAN2 + 4);
lags1[0] = (double)((max_ind - row * (PITCH_LAG_SPAN2 + 4)) +
PITCH_MIN_LAG / 2 - 4);
lags2[0] = (double)(lags1[0] + PITCH_BW - row);
if (lags1[0] < (double) PITCH_MIN_LAG) lags1[0] = (double) PITCH_MIN_LAG;
if (lags2[0] < (double) PITCH_MIN_LAG) lags2[0] = (double) PITCH_MIN_LAG;
if (lags1[0] > (double) PITCH_MAX_LAG) lags1[0] = (double) PITCH_MAX_LAG;
if (lags2[0] > (double) PITCH_MAX_LAG) lags2[0] = (double) PITCH_MAX_LAG;
if (lags1[0] < (double)PITCH_MIN_LAG)
lags1[0] = (double)PITCH_MIN_LAG;
if (lags2[0] < (double)PITCH_MIN_LAG)
lags2[0] = (double)PITCH_MIN_LAG;
if (lags1[0] > (double)PITCH_MAX_LAG)
lags1[0] = (double)PITCH_MAX_LAG;
if (lags2[0] > (double)PITCH_MAX_LAG)
lags2[0] = (double)PITCH_MAX_LAG;
/* store lags of highest peak in output array */
lags[0] = lags1[0];
@ -498,18 +531,20 @@ static void WebRtcIsac_InitializePitch(const double* in,
RTC_POP_IGNORING_WFRAME_LARGER_THAN()
/* create weighting matrix by orthogonalizing a basis of polynomials of increasing order
* t = (0:4)';
* A = [t.^0, t.^1, t.^2, t.^3, t.^4];
* [Q, dummy] = qr(A);
* P.Weight = Q * diag([0, .1, .5, 1, 1]) * Q'; */
/* create weighting matrix by orthogonalizing a basis of polynomials of
* increasing order t = (0:4)'; A = [t.^0, t.^1, t.^2, t.^3, t.^4]; [Q, dummy] =
* qr(A); P.Weight = Q * diag([0, .1, .5, 1, 1]) * Q'; */
static const double kWeight[5][5] = {
{ 0.29714285714286, -0.30857142857143, -0.05714285714286, 0.05142857142857, 0.01714285714286},
{-0.30857142857143, 0.67428571428571, -0.27142857142857, -0.14571428571429, 0.05142857142857},
{-0.05714285714286, -0.27142857142857, 0.65714285714286, -0.27142857142857, -0.05714285714286},
{ 0.05142857142857, -0.14571428571429, -0.27142857142857, 0.67428571428571, -0.30857142857143},
{ 0.01714285714286, 0.05142857142857, -0.05714285714286, -0.30857142857143, 0.29714285714286}
};
{0.29714285714286, -0.30857142857143, -0.05714285714286, 0.05142857142857,
0.01714285714286},
{-0.30857142857143, 0.67428571428571, -0.27142857142857, -0.14571428571429,
0.05142857142857},
{-0.05714285714286, -0.27142857142857, 0.65714285714286, -0.27142857142857,
-0.05714285714286},
{0.05142857142857, -0.14571428571429, -0.27142857142857, 0.67428571428571,
-0.30857142857143},
{0.01714285714286, 0.05142857142857, -0.05714285714286, -0.30857142857143,
0.29714285714286}};
/* second order high-pass filter */
static void WebRtcIsac_Highpass(const double* in,
@ -521,12 +556,12 @@ static void WebRtcIsac_Highpass(const double* in,
* p = 0.94 * exp(j*2*pi*140/8000);
* HP_b = [1, -2*real(z), abs(z)^2];
* HP_a = [1, -2*real(p), abs(p)^2]; */
static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
static const double a_coef[2] = {1.86864659625574, -0.88360000000000};
static const double b_coef[2] = {-1.99524591718270, 0.99600400000000};
size_t k;
for (k=0; k<N; k++) {
for (k = 0; k < N; k++) {
*out = *in + state[1];
state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
@ -535,17 +570,18 @@ static void WebRtcIsac_Highpass(const double* in,
RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN samples */
double *out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
PitchAnalysisStruct *State,
double *lags,
double *gains)
{
void WebRtcIsac_PitchAnalysis(
const double* in, /* PITCH_FRAME_LEN samples */
double* out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
PitchAnalysisStruct* State,
double* lags,
double* gains) {
double HPin[PITCH_FRAME_LEN];
double Weighted[PITCH_FRAME_LEN];
double Whitened[PITCH_FRAME_LEN + QLOOKAHEAD];
double inbuf[PITCH_FRAME_LEN + QLOOKAHEAD];
double out_G[PITCH_FRAME_LEN + QLOOKAHEAD]; // could be removed by using out instead
double out_G[PITCH_FRAME_LEN +
QLOOKAHEAD]; // could be removed by using out instead
double out_dG[4][PITCH_FRAME_LEN + QLOOKAHEAD];
double old_lag, old_gain;
double nrg_wht, tmp;
@ -562,10 +598,12 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
memcpy(Whitened, State->whitened_buf, sizeof(double) * QLOOKAHEAD);
/* compute weighted and whitened signals */
WebRtcIsac_WeightingFilter(HPin, &Weighted[0], &Whitened[QLOOKAHEAD], &(State->Wghtstr));
WebRtcIsac_WeightingFilter(HPin, &Weighted[0], &Whitened[QLOOKAHEAD],
&(State->Wghtstr));
/* copy from buffer into state */
memcpy(State->whitened_buf, Whitened+PITCH_FRAME_LEN, sizeof(double) * QLOOKAHEAD);
memcpy(State->whitened_buf, Whitened + PITCH_FRAME_LEN,
sizeof(double) * QLOOKAHEAD);
old_lag = State->PFstr_wght.oldlagp[0];
old_gain = State->PFstr_wght.oldgainp[0];
@ -573,7 +611,6 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
/* inital pitch estimate */
WebRtcIsac_InitializePitch(Weighted, old_lag, old_gain, State, lags);
/* Iterative optimization of lags - to be done */
/* compute energy of whitened signal */
@ -581,10 +618,10 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
for (k = 0; k < PITCH_FRAME_LEN + QLOOKAHEAD; k++)
nrg_wht += Whitened[k] * Whitened[k];
/* Iterative optimization of gains */
/* set weights for energy, gain fluctiation, and spectral gain penalty functions */
/* set weights for energy, gain fluctiation, and spectral gain penalty
* functions */
Wnrg = 1.0 / nrg_wht;
Wgain = 0.005;
Wfluct = 3.0;
@ -596,9 +633,11 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
/* two iterations should be enough */
for (iter = 0; iter < 2; iter++) {
/* compute Jacobian of pre-filter output towards gains */
WebRtcIsac_PitchfilterPre_gains(Whitened, out_G, out_dG, &(State->PFstr_wght), lags, gains);
WebRtcIsac_PitchfilterPre_gains(Whitened, out_G, out_dG,
&(State->PFstr_wght), lags, gains);
/* gradient and approximate Hessian (lower triangle) for minimizing the filter's output power */
/* gradient and approximate Hessian (lower triangle) for minimizing the
* filter's output power */
for (k = 0; k < 4; k++) {
tmp = 0.0;
for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
@ -614,16 +653,17 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
}
}
/* add gradient and Hessian (lower triangle) for dampening fast gain changes */
/* add gradient and Hessian (lower triangle) for dampening fast gain changes
*/
for (k = 0; k < 4; k++) {
tmp = kWeight[k+1][0] * old_gain;
tmp = kWeight[k + 1][0] * old_gain;
for (m = 0; m < 4; m++)
tmp += kWeight[k+1][m+1] * gains[m];
tmp += kWeight[k + 1][m + 1] * gains[m];
grad[k] += tmp * Wfluct;
}
for (k = 0; k < 4; k++) {
for (m = 0; m <= k; m++) {
H[k][m] += kWeight[k+1][m+1] * Wfluct;
H[k][m] += kWeight[k + 1][m + 1] * Wfluct;
}
}
@ -637,10 +677,10 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
grad[3] += 1.33 * (tmp * tmp * Wgain);
H[3][3] += 2.66 * tmp * (tmp * tmp * Wgain);
/* compute Cholesky factorization of Hessian
* by overwritting the upper triangle; scale factors on diagonal
* (for non pc-platforms store the inverse of the diagonals seperately to minimize divisions) */
* (for non pc-platforms store the inverse of the diagonals seperately to
* minimize divisions) */
H[0][1] = H[1][0] / H[0][0];
H[0][2] = H[2][0] / H[0][0];
H[0][3] = H[3][0] / H[0][0];
@ -648,8 +688,10 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
H[1][2] = (H[2][1] - H[0][1] * H[2][0]) / H[1][1];
H[1][3] = (H[3][1] - H[0][1] * H[3][0]) / H[1][1];
H[2][2] -= H[0][0] * H[0][2] * H[0][2] + H[1][1] * H[1][2] * H[1][2];
H[2][3] = (H[3][2] - H[0][2] * H[3][0] - H[1][2] * H[1][1] * H[1][3]) / H[2][2];
H[3][3] -= H[0][0] * H[0][3] * H[0][3] + H[1][1] * H[1][3] * H[1][3] + H[2][2] * H[2][3] * H[2][3];
H[2][3] =
(H[3][2] - H[0][2] * H[3][0] - H[1][2] * H[1][1] * H[1][3]) / H[2][2];
H[3][3] -= H[0][0] * H[0][3] * H[0][3] + H[1][1] * H[1][3] * H[1][3] +
H[2][2] * H[2][3] * H[2][3];
/* Compute update as delta_gains = -inv(H) * grad */
/* copy and negate */
@ -682,7 +724,7 @@ void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN
/* concatenate previous input's end and current input */
memcpy(inbuf, State->inbuf, sizeof(double) * QLOOKAHEAD);
memcpy(inbuf+QLOOKAHEAD, in, sizeof(double) * PITCH_FRAME_LEN);
memcpy(inbuf + QLOOKAHEAD, in, sizeof(double) * PITCH_FRAME_LEN);
/* lookahead pitch filtering for masking analysis */
WebRtcIsac_PitchfilterPre_la(inbuf, out, &(State->PFstr), lags, gains);

65
modules/audio_coding/codecs/isac/main/source/pitch_filter.c
View File

@ -12,8 +12,8 @@
#include <memory.h>
#include <stdlib.h>
#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
#include "rtc_base/compile_assert_c.h"
/*
@ -58,8 +58,7 @@ static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
-0.01463300534216},
{0.01654127246315, -0.04533310458085, 0.09585268418557, -0.20266133815190,
0.79117042386878, 0.44560418147640, -0.13991265473712, 0.05816126837865,
-0.01985640750433}
};
-0.01985640750433}};
/*
* Enumerating the operation of the filter.
@ -78,7 +77,10 @@ static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
* used to find the optimal gain.
*/
typedef enum {
kPitchFilterPre, kPitchFilterPost, kPitchFilterPreLa, kPitchFilterPreGain
kPitchFilterPre,
kPitchFilterPost,
kPitchFilterPreLa,
kPitchFilterPreGain
} PitchFilterOperation;
/*
@ -104,7 +106,7 @@ typedef enum {
typedef struct {
double buffer[PITCH_INTBUFFSIZE + QLOOKAHEAD];
double damper_state[PITCH_DAMPORDER];
const double *interpol_coeff;
const double* interpol_coeff;
double gain;
double lag;
int lag_offset;
@ -132,7 +134,8 @@ typedef struct {
* where the output of different gain values (differential
* change to gain) is written.
*/
static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
static void FilterSegment(const double* in_data,
PitchFilterParam* parameters,
double* out_data,
double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
int n;
@ -173,15 +176,15 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
for (j = 0; j < parameters->sub_frame + 1; ++j) {
/* Filter for fractional pitch. */
sum2 = 0.0;
for (m = PITCH_FRACORDER-1; m >= m_tmp; --m) {
for (m = PITCH_FRACORDER - 1; m >= m_tmp; --m) {
/* `lag_index + m` is always larger than or equal to zero, see how
* m_tmp is computed. This is equivalent to assume samples outside
* `out_dg[j]` are zero. */
sum2 += out_dg[j][lag_index + m] * parameters->interpol_coeff[m];
}
/* Add the contribution of differential gain change. */
parameters->damper_state_dg[j][0] = parameters->gain_mult[j] * sum +
parameters->gain * sum2;
parameters->damper_state_dg[j][0] =
parameters->gain_mult[j] * sum + parameters->gain * sum2;
}
/* Filter with damping filter, and store the results. */
@ -201,8 +204,8 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
/* Subtract from input and update buffer. */
out_data[parameters->index] = in_data[parameters->index] - sum;
parameters->buffer[pos] = in_data[parameters->index] +
out_data[parameters->index];
parameters->buffer[pos] =
in_data[parameters->index] + out_data[parameters->index];
++parameters->index;
++pos;
@ -216,8 +219,8 @@ static void Update(PitchFilterParam* parameters) {
double fraction;
int fraction_index;
/* Compute integer lag-offset. */
parameters->lag_offset = WebRtcIsac_lrint(parameters->lag + PITCH_FILTDELAY +
0.5);
parameters->lag_offset =
WebRtcIsac_lrint(parameters->lag + PITCH_FILTDELAY + 0.5);
/* Find correct set of coefficients for computing fractional pitch. */
fraction = parameters->lag_offset - (parameters->lag + PITCH_FILTDELAY);
fraction_index = WebRtcIsac_lrint(PITCH_FRACS * fraction - 0.5);
@ -257,8 +260,11 @@ static void Update(PitchFilterParam* parameters) {
* where the output of different gain values (differential
* change to gain) is written.
*/
static void FilterFrame(const double* in_data, PitchFiltstr* filter_state,
double* lags, double* gains, PitchFilterOperation mode,
static void FilterFrame(const double* in_data,
PitchFiltstr* filter_state,
double* lags,
double* gains,
PitchFilterOperation mode,
double* out_data,
double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
PitchFilterParam filter_parameters;
@ -289,7 +295,7 @@ static void FilterFrame(const double* in_data, PitchFiltstr* filter_state,
memset(filter_parameters.damper_state_dg, 0,
sizeof(filter_parameters.damper_state_dg));
for (n = 0; n < PITCH_SUBFRAMES; ++n) {
//memset(out_dg[n], 0, sizeof(double) * (PITCH_FRAME_LEN + QLOOKAHEAD));
// memset(out_dg[n], 0, sizeof(double) * (PITCH_FRAME_LEN + QLOOKAHEAD));
memset(out_dg[n], 0, sizeof(out_dg[n]));
}
} else if (mode == kPitchFilterPost) {
@ -360,29 +366,38 @@ static void FilterFrame(const double* in_data, PitchFiltstr* filter_state,
}
}
void WebRtcIsac_PitchfilterPre(double* in_data, double* out_data,
PitchFiltstr* pf_state, double* lags,
void WebRtcIsac_PitchfilterPre(double* in_data,
double* out_data,
PitchFiltstr* pf_state,
double* lags,
double* gains) {
FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPre, out_data, NULL);
}
void WebRtcIsac_PitchfilterPre_la(double* in_data, double* out_data,
PitchFiltstr* pf_state, double* lags,
void WebRtcIsac_PitchfilterPre_la(double* in_data,
double* out_data,
PitchFiltstr* pf_state,
double* lags,
double* gains) {
FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreLa, out_data,
NULL);
}
void WebRtcIsac_PitchfilterPre_gains(
double* in_data, double* out_data,
double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD], PitchFiltstr *pf_state,
double* lags, double* gains) {
double* in_data,
double* out_data,
double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD],
PitchFiltstr* pf_state,
double* lags,
double* gains) {
FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreGain, out_data,
out_dg);
}
void WebRtcIsac_PitchfilterPost(double* in_data, double* out_data,
PitchFiltstr* pf_state, double* lags,
void WebRtcIsac_PitchfilterPost(double* in_data,
double* out_data,
PitchFiltstr* pf_state,
double* lags,
double* gains) {
FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPost, out_data, NULL);
}