Format the rest of C files in the repo
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}
diff --git a/common_audio/ring_buffer.c b/common_audio/ring_buffer.c
index 590f5f9..a3fabd0 100644
--- a/common_audio/ring_buffer.c
+++ b/common_audio/ring_buffer.c
@@ -28,10 +28,9 @@
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 @@
void** data_ptr,
void* data,
size_t element_count) {
-
if (self == NULL) {
return 0;
}
@@ -112,17 +110,14 @@
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 @@
}
// Update read position
- WebRtc_MoveReadPtr(self, (int) read_count);
+ WebRtc_MoveReadPtr(self, (int)read_count);
return read_count;
}
@@ -152,21 +147,21 @@
{
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 @@
{
// 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 @@
}
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;
}
diff --git a/common_audio/vad/vad_core.c b/common_audio/vad/vad_core.c
index 0872449..9b40f42 100644
--- a/common_audio/vad/vad_core.c
+++ b/common_audio/vad/vad_core.c
@@ -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 kNoiseUpdateConst = 655; // Q15
-static const int16_t kSpeechUpdateConst = 6554; // Q15
-static const int16_t kBackEta = 154; // Q8
+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 @@
// 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 @@
// - 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 @@
// - 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 @@
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 @@
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 @@
// 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 @@
// 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
+ tmp1_s32 = (noise_probability[0] & 0xFFFFF000) << 2; // Q29
+ 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 @@
}
// 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
+ tmp1_s32 = (speech_probability[0] & 0xFFFFF000) << 2; // Q29
+ sgprvec[channel] = (int16_t)WebRtcSpl_DivW32W16(tmp1_s32, h1); // Q14
sgprvec[channel + kNumChannels] = 16384 - sgprvec[channel];
}
}
@@ -277,14 +278,13 @@
// 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 @@
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 @@
// 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).
@@ -394,12 +394,12 @@
// 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
+ tmp_s16 += 32; // Rounding
nsk += tmp_s16 >> 6; // Q13 >> 6 = Q7.
if (nsk < kMinStd) {
nsk = kMinStd;
@@ -419,8 +419,8 @@
// `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 @@
// 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]);
+ 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,
- &kNoiseDataWeights[channel]);
+ 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 @@
}
}
- 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 @@
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 @@
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 @@
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 @@
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 @@
// 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 @@
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
@@ -629,57 +621,57 @@
return vad;
}
-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)
+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);
+ len = frame_length / 2;
- // Downsample signal 32->16->8 before doing VAD
- 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);
+ len /= 2;
- WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
- len /= 2;
+ // Do VAD on an 8 kHz signal
+ vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
- // Do VAD on an 8 kHz signal
- vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
-
- return vad;
+ return vad;
}
-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)
+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);
+ // Wideband: Downsample signal before doing VAD
+ WebRtcVad_Downsampling(speech_frame, speechNB,
+ inst->downsampling_filter_states, frame_length);
- len = frame_length / 2;
- vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
+ len = frame_length / 2;
+ vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
- return vad;
+ return vad;
}
-int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
- size_t frame_length)
-{
- int16_t feature_vector[kNumChannels], total_power;
+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
- total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
- feature_vector);
+ // Get power in the bands
+ total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
+ feature_vector);
- // Make a VAD
- inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
+ // Make a VAD
+ inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
- return inst->vad;
+ return inst->vad;
}
diff --git a/common_audio/vad/vad_filterbank.c b/common_audio/vad/vad_filterbank.c
index aff63f7..32830fa0 100644
--- a/common_audio/vad/vad_filterbank.c
+++ b/common_audio/vad/vad_filterbank.c
@@ -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 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 @@
// - 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 @@
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 @@
// - 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 @@
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.
+ 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 @@
// 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 @@
// 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 @@
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
@@ -205,12 +213,12 @@
// 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.
*log_energy = (int16_t)(((kLogConst * log2_energy) >> 19) +
- ((tot_rshifts * kLogConst) >> 9));
+ ((tot_rshifts * kLogConst) >> 9));
if (*log_energy < 0) {
*log_energy = 0;
@@ -235,13 +243,15 @@
// 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
@@ -256,8 +266,8 @@
// Initialize variables for the first SplitFilter().
int frequency_band = 0;
const int16_t* in_ptr = data_in; // [0 - 4000] Hz.
- int16_t* hp_out_ptr = hp_120; // [2000 - 4000] Hz.
- int16_t* lp_out_ptr = lp_120; // [0 - 2000] Hz.
+ int16_t* hp_out_ptr = hp_120; // [2000 - 4000] Hz.
+ 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`.
@@ -268,7 +278,7 @@
// For the upper band (2000 Hz - 4000 Hz) split at 3000 Hz and downsample.
frequency_band = 1;
- in_ptr = hp_120; // [2000 - 4000] Hz.
+ in_ptr = hp_120; // [2000 - 4000] Hz.
hp_out_ptr = hp_60; // [3000 - 4000] Hz.
lp_out_ptr = lp_60; // [2000 - 3000] Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
@@ -284,9 +294,9 @@
// For the lower band (0 Hz - 2000 Hz) split at 1000 Hz and downsample.
frequency_band = 2;
- in_ptr = lp_120; // [0 - 2000] Hz.
- hp_out_ptr = hp_60; // [1000 - 2000] Hz.
- lp_out_ptr = lp_60; // [0 - 1000] Hz.
+ 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.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
@@ -297,7 +307,7 @@
// For the lower band (0 Hz - 1000 Hz) split at 500 Hz and downsample.
frequency_band = 3;
- in_ptr = lp_60; // [0 - 1000] Hz.
+ in_ptr = lp_60; // [0 - 1000] Hz.
hp_out_ptr = hp_120; // [500 - 1000] Hz.
lp_out_ptr = lp_120; // [0 - 500] Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
@@ -309,7 +319,7 @@
// For the lower band (0 Hz - 500 Hz) split at 250 Hz and downsample.
frequency_band = 4;
- in_ptr = lp_120; // [0 - 500] Hz.
+ in_ptr = lp_120; // [0 - 500] Hz.
hp_out_ptr = hp_60; // [250 - 500] Hz.
lp_out_ptr = lp_60; // [0 - 250] Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
diff --git a/common_audio/vad/vad_gmm.c b/common_audio/vad/vad_gmm.c
index 4a7fe67..46d2de1 100644
--- a/common_audio/vad/vad_gmm.c
+++ b/common_audio/vad/vad_gmm.c
@@ -36,8 +36,8 @@
// 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.
diff --git a/common_audio/vad/vad_sp.c b/common_audio/vad/vad_sp.c
index 3d24cf6..b745465 100644
--- a/common_audio/vad/vad_sp.c
+++ b/common_audio/vad/vad_sp.c
@@ -10,15 +10,15 @@
#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 kSmoothingDown = 6553; // 0.2 in Q15.
-static const int16_t kSmoothingUp = 32439; // 0.99 in Q15.
+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.
// TODO(bjornv): Move this function to vad_filterbank.c.
// Downsampling filter based on splitting filter and allpass functions.
@@ -36,14 +36,14 @@
// 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 @@
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];
}
diff --git a/common_audio/vad/webrtc_vad.c b/common_audio/vad/webrtc_vad.c
index 6dd14d8..d3c8b08 100644
--- a/common_audio/vad/webrtc_vad.c
+++ b/common_audio/vad/webrtc_vad.c
@@ -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 @@
// 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 @@
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;
@@ -73,7 +75,7 @@
}
if (fs == 48000) {
- vad = WebRtcVad_CalcVad48khz(self, audio_frame, frame_length);
+ vad = WebRtcVad_CalcVad48khz(self, audio_frame, frame_length);
} else if (fs == 32000) {
vad = WebRtcVad_CalcVad32khz(self, audio_frame, frame_length);
} else if (fs == 16000) {
@@ -99,7 +101,7 @@
for (i = 0; i < kRatesSize; i++) {
if (kValidRates[i] == rate) {
for (valid_length_ms = 10; valid_length_ms <= kMaxFrameLengthMs;
- valid_length_ms += 10) {
+ valid_length_ms += 10) {
valid_length = (size_t)(kValidRates[i] / 1000 * valid_length_ms);
if (frame_length == valid_length) {
return_value = 0;
diff --git a/modules/audio_coding/codecs/g711/g711_interface.c b/modules/audio_coding/codecs/g711/g711_interface.c
index 5fe1692..84b08cb 100644
--- a/modules/audio_coding/codecs/g711/g711_interface.c
+++ b/modules/audio_coding/codecs/g711/g711_interface.c
@@ -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,
diff --git a/modules/audio_coding/codecs/g722/g722_interface.c b/modules/audio_coding/codecs/g722/g722_interface.c
index 36ee6d9..0744e99 100644
--- a/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/modules/audio_coding/codecs/g722/g722_interface.c
@@ -8,60 +8,56 @@
* 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);
- } else {
- return(-1);
- }
+int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst) {
+ *G722enc_inst = (G722EncInst*)malloc(sizeof(G722EncoderState));
+ if (*G722enc_inst != NULL) {
+ return (0);
+ } else {
+ return (-1);
+ }
}
-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);
- if (G722enc_inst == NULL) {
- return -1;
- } else {
- return 0;
- }
+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);
+ if (G722enc_inst == NULL) {
+ return -1;
+ } else {
+ return 0;
+ }
}
-int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
-{
- // Free encoder memory
- return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
+int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst) {
+ // Free encoder memory
+ 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;
- // Encode the input speech vector
- return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
- speechIn, len);
+ uint8_t* encoded) {
+ unsigned char* codechar = (unsigned char*)encoded;
+ // Encode the input speech vector
+ 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);
- } else {
- return(-1);
- }
+int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst) {
+ *G722dec_inst = (G722DecInst*)malloc(sizeof(G722DecoderState));
+ if (*G722dec_inst != NULL) {
+ return (0);
+ } else {
+ return (-1);
+ }
}
void WebRtcG722_DecoderInit(G722DecInst* inst) {
@@ -70,35 +66,29 @@
WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
}
-int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
-{
- // Free encoder memory
- return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
+int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst) {
+ // Free encoder memory
+ 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)
-{
- // Decode the G.722 encoder stream
- *speechType=G722_WEBRTC_SPEECH;
- return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
- encoded, len);
+ 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);
}
-int16_t WebRtcG722_Version(char *versionStr, short len)
-{
- // Get version string
- char version[30] = "2.0.0\n";
- if (strlen(version) < (unsigned int)len)
- {
- strcpy(versionStr, version);
- return 0;
- }
- else
- {
- return -1;
- }
+int16_t WebRtcG722_Version(char* versionStr, short len) {
+ // Get version string
+ char version[30] = "2.0.0\n";
+ if (strlen(version) < (unsigned int)len) {
+ strcpy(versionStr, version);
+ return 0;
+ } else {
+ return -1;
+ }
}
diff --git a/modules/audio_coding/codecs/isac/main/source/filter_functions.c b/modules/audio_coding/codecs/isac/main/source/filter_functions.c
index a4f297c..d359e8f 100644
--- a/modules/audio_coding/codecs/isac/main/source/filter_functions.c
+++ b/modules/audio_coding/codecs/isac/main/source/filter_functions.c
@@ -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 @@
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 @@
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 @@
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;
+ 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 @@
sum += prod;
r[lag] = sum;
}
-
}
static void WebRtcIsac_BwExpand(double* out,
@@ -116,7 +109,7 @@
double coef,
size_t length) {
size_t i;
- double chirp;
+ double chirp;
chirp = coef;
@@ -131,65 +124,68 @@
double* weiout,
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 whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
- double weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
- double *weo, *who, opol[PITCH_WLPCORDER+1], ext[PITCH_WLPCWINLEN];
- int k, n, endpos, start;
+ 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 whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+ double weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+ 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);
}
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c b/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
index 8a19ac1..157eb19 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
@@ -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,
- 0.60086484101160, -0.12203175715679, 0.02184247643159, -0.00067556028640};
+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 @@
*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 @@
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;
}
@@ -91,27 +92,25 @@
*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];
}
- outcorr += PITCH_LAG_SPAN2 - 1; /* index of last element in array */
+ outcorr += PITCH_LAG_SPAN2 - 1; /* index of last element in array */
*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 @@
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 @@
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 @@
/* 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 @@
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;
@@ -256,7 +263,7 @@
corr = corrvec1[ind1++] + corrvec2[ind2++];
CorrSurfPtr1[k] = corr;
if (corr > corr_max) {
- corr_max = corr; /* update maximum */
+ corr_max = corr; /* update maximum */
max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
}
}
@@ -264,63 +271,66 @@
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) {
- corr_max = corr; /* update maximum */
+ corr_max = corr; /* update maximum */
max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
}
corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
CorrSurfPtr2[k] = corr;
if (corr > corr_max) {
- corr_max = corr; /* update maximum */
+ corr_max = corr; /* update maximum */
max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
}
}
/* 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) {
- corr_max = corr; /* update maximum */
+ corr_max = corr; /* update maximum */
max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
}
corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
CorrSurfPtr2[k] = corr;
if (corr > corr_max) {
- corr_max = corr; /* update maximum */
+ corr_max = corr; /* update maximum */
max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
}
}
/* fill remainder of correlation surface */
for (m = 2; m < PITCH_BW; m++) {
ind1 = 0;
- ind2 = PITCH_BW - m; /* always larger than ind1 */
+ 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) {
- corr_max = corr; /* update maximum */
+ corr_max = corr; /* update maximum */
max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
}
corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
CorrSurfPtr2[k] = corr;
if (corr > corr_max) {
- corr_max = corr; /* update maximum */
+ corr_max = corr; /* update maximum */
max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
}
}
@@ -331,33 +341,41 @@
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 @@
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 @@
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 @@
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];
@@ -498,35 +531,37 @@
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,
- double* out,
- double* state,
- size_t N) {
+ double* out,
+ double* state,
+ size_t N) {
/* create high-pass filter ocefficients
* z = 0.998 * exp(j*2*pi*35/8000);
* 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 b_coef[2] = {-1.99524591718270, 0.99600400000000};
+ 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 @@
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 @@
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 @@
/* 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 @@
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 @@
/* 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 @@
}
}
- /* 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 @@
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 @@
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 @@
/* 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);
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
index bf03dff..494b5b7 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
@@ -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"
/*
@@ -31,35 +31,34 @@
*/
static const double kDampFilter[PITCH_DAMPORDER] = {-0.07, 0.25, 0.64, 0.25,
- -0.07};
+ -0.07};
/* interpolation coefficients; generated by design_pitch_filter.m */
static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
- {-0.02239172458614, 0.06653315052934, -0.16515880017569, 0.60701333734125,
- 0.64671399919202, -0.20249000396417, 0.09926548334755, -0.04765933793109,
+ {-0.02239172458614, 0.06653315052934, -0.16515880017569, 0.60701333734125,
+ 0.64671399919202, -0.20249000396417, 0.09926548334755, -0.04765933793109,
0.01754159521746},
- {-0.01985640750434, 0.05816126837866, -0.13991265473714, 0.44560418147643,
- 0.79117042386876, -0.20266133815188, 0.09585268418555, -0.04533310458084,
+ {-0.01985640750434, 0.05816126837866, -0.13991265473714, 0.44560418147643,
+ 0.79117042386876, -0.20266133815188, 0.09585268418555, -0.04533310458084,
0.01654127246314},
- {-0.01463300534216, 0.04229888475060, -0.09897034715253, 0.28284326017787,
- 0.90385267956632, -0.16976950138649, 0.07704272393639, -0.03584218578311,
+ {-0.01463300534216, 0.04229888475060, -0.09897034715253, 0.28284326017787,
+ 0.90385267956632, -0.16976950138649, 0.07704272393639, -0.03584218578311,
0.01295781500709},
- {-0.00764851320885, 0.02184035544377, -0.04985561057281, 0.13083306574393,
- 0.97545011664662, -0.10177807997561, 0.04400901776474, -0.02010737175166,
+ {-0.00764851320885, 0.02184035544377, -0.04985561057281, 0.13083306574393,
+ 0.97545011664662, -0.10177807997561, 0.04400901776474, -0.02010737175166,
0.00719783432422},
- {-0.00000000000000, 0.00000000000000, -0.00000000000001, 0.00000000000001,
- 0.99999999999999, 0.00000000000001, -0.00000000000001, 0.00000000000000,
+ {-0.00000000000000, 0.00000000000000, -0.00000000000001, 0.00000000000001,
+ 0.99999999999999, 0.00000000000001, -0.00000000000001, 0.00000000000000,
-0.00000000000000},
- {0.00719783432422, -0.02010737175166, 0.04400901776474, -0.10177807997562,
- 0.97545011664663, 0.13083306574393, -0.04985561057280, 0.02184035544377,
+ {0.00719783432422, -0.02010737175166, 0.04400901776474, -0.10177807997562,
+ 0.97545011664663, 0.13083306574393, -0.04985561057280, 0.02184035544377,
-0.00764851320885},
- {0.01295781500710, -0.03584218578312, 0.07704272393640, -0.16976950138650,
- 0.90385267956634, 0.28284326017785, -0.09897034715252, 0.04229888475059,
+ {0.01295781500710, -0.03584218578312, 0.07704272393640, -0.16976950138650,
+ 0.90385267956634, 0.28284326017785, -0.09897034715252, 0.04229888475059,
-0.01463300534216},
- {0.01654127246315, -0.04533310458085, 0.09585268418557, -0.20266133815190,
- 0.79117042386878, 0.44560418147640, -0.13991265473712, 0.05816126837865,
- -0.01985640750433}
-};
+ {0.01654127246315, -0.04533310458085, 0.09585268418557, -0.20266133815190,
+ 0.79117042386878, 0.44560418147640, -0.13991265473712, 0.05816126837865,
+ -0.01985640750433}};
/*
* Enumerating the operation of the filter.
@@ -78,7 +77,10 @@
* used to find the optimal gain.
*/
typedef enum {
- kPitchFilterPre, kPitchFilterPost, kPitchFilterPreLa, kPitchFilterPreGain
+ kPitchFilterPre,
+ kPitchFilterPost,
+ kPitchFilterPreLa,
+ kPitchFilterPreGain
} PitchFilterOperation;
/*
@@ -104,7 +106,7 @@
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 @@
* 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 @@
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 @@
/* 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 @@
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 @@
* 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;
@@ -276,7 +282,7 @@
memcpy(filter_parameters.buffer, filter_state->ubuf,
sizeof(filter_state->ubuf));
RTC_COMPILE_ASSERT(sizeof(filter_parameters.buffer) >=
- sizeof(filter_state->ubuf));
+ sizeof(filter_state->ubuf));
memset(filter_parameters.buffer +
sizeof(filter_state->ubuf) / sizeof(filter_state->ubuf[0]),
0, sizeof(filter_parameters.buffer) - sizeof(filter_state->ubuf));
@@ -289,7 +295,7 @@
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 @@
}
}
-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);
}
