Reland "Avoiding overflow in cross correlation in NetEq."
The original CL is https://codereview.webrtc.org/1908623002/
An error was caused by that and this CL fix that problem and reland the CL.
BUG=
Review-Url: https://codereview.webrtc.org/1931933004
Cr-Commit-Position: refs/heads/master@{#12589}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index a51a73a..544bfa0 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -787,6 +787,8 @@
"neteq/buffer_level_filter.h",
"neteq/comfort_noise.cc",
"neteq/comfort_noise.h",
+ "neteq/cross_correlation.cc",
+ "neteq/cross_correlation.h",
"neteq/decision_logic.cc",
"neteq/decision_logic.h",
"neteq/decision_logic_fax.cc",
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index dc6bbf6..a2ef5b0 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -939,34 +939,34 @@
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
- Run(8000, PlatformChecksum("f34e5c0e4dd4cd6c82b23f6ed006dad0",
- "67a1471049dc87e7498bc19bf130dd35",
- "efb5a07480bad8afb184c4150f4b3f3a",
- "51717ab374871cbfa2c6977ea2aa40f3"),
+ Run(8000, PlatformChecksum("90be25dd9505005aaadf91b77ee31624",
+ "ac6dc4b5bf6d277f693889c4c916882e",
+ "a607f7d0ba98683c9c236217f86aaa6b",
+ "4a54f6ec712bda58484a388e1a332b42"),
std::vector<ExternalDecoder>());
}
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
- Run(16000, PlatformChecksum("5066b412805f3050f65154d676006964",
- "887905a40d37f213b76f64296871473e",
- "f580bfd4e5e29f0399b61b7512d4e3b4",
- "5b2ae32c590b41d0c601179e14eaae96"),
+ Run(16000, PlatformChecksum("2c713197d41becd52c1ceecbd2b9f687",
+ "130cc2a43063c74197122e3760690e7d",
+ "cdc3d88f6d8e497d4e00c62c0e6dbb3c",
+ "83edb67c157d0e3a0fb9f7d7b1ce5dc7"),
std::vector<ExternalDecoder>());
}
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
- Run(32000, PlatformChecksum("2cb4784af507c45b9121e2315def36f2",
- "d2392b3247095d894a49b74a1106f281",
- "fdf5166b98c43235978685e40e28fea6",
- "7f620312f2fa74a10048bbb7739d4bf3"),
+ Run(32000, PlatformChecksum("fe5851d43c13df66a7ad30fdb124e62f",
+ "309d24be4b287dc92c340f10a807a11e",
+ "c4a0e0b2e031d62c693af2a9ff4337ac",
+ "4cbfc6ab4d704f5d9b4f10406437fda9"),
std::vector<ExternalDecoder>());
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
- Run(48000, PlatformChecksum("ce63f874a198621fa35398e412640fcf",
- "2cf0b8fe9784e8c96db307e125beb723",
- "71f89e87ee1bad594f529d6c036289ad",
- "b64c891e99eccc9ff45541ef67c9e9bf"),
+ Run(48000, PlatformChecksum("a9241f426b4bf2ac650b6d287469a550",
+ "30374fd4a932df942c1b1120e7b724ad",
+ "22242dd832824046d48db9ea8a01f84c",
+ "c7f46bf165400b266d9b57aee02d2747"),
std::vector<ExternalDecoder>());
}
@@ -1021,10 +1021,10 @@
std::vector<ExternalDecoder> external_decoders;
external_decoders.push_back(ed);
- Run(48000, PlatformChecksum("ce63f874a198621fa35398e412640fcf",
- "2cf0b8fe9784e8c96db307e125beb723",
- "71f89e87ee1bad594f529d6c036289ad",
- "b64c891e99eccc9ff45541ef67c9e9bf"),
+ Run(48000, PlatformChecksum("a9241f426b4bf2ac650b6d287469a550",
+ "30374fd4a932df942c1b1120e7b724ad",
+ "22242dd832824046d48db9ea8a01f84c",
+ "c7f46bf165400b266d9b57aee02d2747"),
external_decoders);
EXPECT_CALL(mock_decoder, Die());
diff --git a/webrtc/modules/audio_coding/neteq/background_noise.cc b/webrtc/modules/audio_coding/neteq/background_noise.cc
index 7e7a632..c86045e 100644
--- a/webrtc/modules/audio_coding/neteq/background_noise.cc
+++ b/webrtc/modules/audio_coding/neteq/background_noise.cc
@@ -17,6 +17,7 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
namespace webrtc {
@@ -169,15 +170,10 @@
int32_t BackgroundNoise::CalculateAutoCorrelation(
const int16_t* signal, size_t length, int32_t* auto_correlation) const {
- int16_t signal_max = WebRtcSpl_MaxAbsValueW16(signal, length);
- int correlation_scale = kLogVecLen -
- WebRtcSpl_NormW32(signal_max * signal_max);
- correlation_scale = std::max(0, correlation_scale);
-
static const int kCorrelationStep = -1;
- WebRtcSpl_CrossCorrelation(auto_correlation, signal, signal, length,
- kMaxLpcOrder + 1, correlation_scale,
- kCorrelationStep);
+ const int correlation_scale =
+ CrossCorrelationWithAutoShift(signal, signal, length, kMaxLpcOrder + 1,
+ kCorrelationStep, auto_correlation);
// Number of shifts to normalize energy to energy/sample.
int energy_sample_shift = kLogVecLen - correlation_scale;
diff --git a/webrtc/modules/audio_coding/neteq/cross_correlation.cc b/webrtc/modules/audio_coding/neteq/cross_correlation.cc
new file mode 100644
index 0000000..ad89ab8
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/cross_correlation.cc
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
+
+#include <cstdlib>
+#include <limits>
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+// This function decides the overflow-protecting scaling and calls
+// WebRtcSpl_CrossCorrelation.
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+ const int16_t* sequence_2,
+ size_t sequence_1_length,
+ size_t cross_correlation_length,
+ int cross_correlation_step,
+ int32_t* cross_correlation) {
+ // Find the maximum absolute value of sequence_1 and 2.
+ const int16_t max_1 = WebRtcSpl_MaxAbsValueW16(sequence_1, sequence_1_length);
+ const int sequence_2_shift =
+ cross_correlation_step * (static_cast<int>(cross_correlation_length) - 1);
+ const int16_t* sequence_2_start =
+ sequence_2_shift >= 0 ? sequence_2 : sequence_2 + sequence_2_shift;
+ const size_t sequence_2_length =
+ sequence_1_length + std::abs(sequence_2_shift);
+ const int16_t max_2 =
+ WebRtcSpl_MaxAbsValueW16(sequence_2_start, sequence_2_length);
+
+ // In order to avoid overflow when computing the sum we should scale the
+ // samples so that (in_vector_length * max_1 * max_2) will not overflow.
+ // Expected scaling fulfills
+ // 1) sufficient:
+ // sequence_1_length * (max_1 * max_2 >> scaling) <= 0x7fffffff;
+ // 2) necessary:
+ // if (scaling > 0)
+ // sequence_1_length * (max_1 * max_2 >> (scaling - 1)) > 0x7fffffff;
+ // The following calculation fulfills 1) and almost fulfills 2).
+ // There are some corner cases that 2) is not satisfied, e.g.,
+ // max_1 = 17, max_2 = 30848, sequence_1_length = 4095, in such case,
+ // optimal scaling is 0, while the following calculation results in 1.
+ const int32_t factor = (max_1 * max_2) / (std::numeric_limits<int32_t>::max()
+ / static_cast<int32_t>(sequence_1_length));
+ const int scaling = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+
+ WebRtcSpl_CrossCorrelation(cross_correlation, sequence_1, sequence_2,
+ sequence_1_length, cross_correlation_length,
+ scaling, cross_correlation_step);
+
+ return scaling;
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/cross_correlation.h b/webrtc/modules/audio_coding/neteq/cross_correlation.h
new file mode 100644
index 0000000..db14141
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/cross_correlation.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+// The function calculates the cross-correlation between two sequences
+// |sequence_1| and |sequence_2|. |sequence_1| is taken as reference, with
+// |sequence_1_length| as its length. |sequence_2| slides for the calculation of
+// cross-correlation. The result will be saved in |cross_correlation|.
+// |cross_correlation_length| correlation points are calculated.
+// The corresponding lag starts from 0, and increases with a step of
+// |cross_correlation_step|. The result is without normalization. To avoid
+// overflow, the result will be right shifted. The amount of shifts will be
+// returned.
+//
+// Input:
+// - sequence_1 : First sequence (reference).
+// - sequence_2 : Second sequence (sliding during calculation).
+// - sequence_1_length : Length of |sequence_1|.
+// - cross_correlation_length : Number of cross-correlations to calculate.
+// - cross_correlation_step : Step in the lag for the cross-correlation.
+//
+// Output:
+// - cross_correlation : The cross-correlation in Q(-right_shifts)
+//
+// Return:
+// Number of right shifts in cross_correlation.
+
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+ const int16_t* sequence_2,
+ size_t sequence_1_length,
+ size_t cross_correlation_length,
+ int cross_correlation_step,
+ int32_t* cross_correlation);
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index ef7af46..5cb18db 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -19,6 +19,7 @@
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
@@ -379,12 +380,10 @@
InitializeForAnExpandPeriod();
// Calculate correlation in downsampled domain (4 kHz sample rate).
- int correlation_scale;
size_t correlation_length = 51; // TODO(hlundin): Legacy bit-exactness.
// If it is decided to break bit-exactness |correlation_length| should be
// initialized to the return value of Correlation().
- Correlation(audio_history, signal_length, correlation_vector,
- &correlation_scale);
+ Correlation(audio_history, signal_length, correlation_vector);
// Find peaks in correlation vector.
DspHelper::PeakDetection(correlation_vector, correlation_length,
@@ -455,7 +454,7 @@
&audio_history[signal_length - correlation_length - start_index
- correlation_lags],
correlation_length + start_index + correlation_lags - 1);
- correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
+ int correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
(31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
correlation_scale = std::max(0, correlation_scale);
@@ -582,13 +581,6 @@
}
// Calculate the LPC and the gain of the filters.
- // Calculate scale value needed for auto-correlation.
- correlation_scale = WebRtcSpl_MaxAbsValueW16(
- &(audio_history[signal_length - fs_mult_lpc_analysis_len]),
- fs_mult_lpc_analysis_len);
-
- correlation_scale = std::min(16 - WebRtcSpl_NormW32(correlation_scale), 0);
- correlation_scale = std::max(correlation_scale * 2 + 7, 0);
// Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function.
size_t temp_index = signal_length - fs_mult_lpc_analysis_len -
@@ -601,11 +593,9 @@
memcpy(&temp_signal[kUnvoicedLpcOrder],
&audio_history[temp_index + kUnvoicedLpcOrder],
sizeof(int16_t) * fs_mult_lpc_analysis_len);
- WebRtcSpl_CrossCorrelation(auto_correlation,
- &temp_signal[kUnvoicedLpcOrder],
- &temp_signal[kUnvoicedLpcOrder],
- fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1,
- correlation_scale, -1);
+ CrossCorrelationWithAutoShift(
+ &temp_signal[kUnvoicedLpcOrder], &temp_signal[kUnvoicedLpcOrder],
+ fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, -1, auto_correlation);
delete [] temp_signal;
// Verify that variance is positive.
@@ -766,8 +756,7 @@
void Expand::Correlation(const int16_t* input,
size_t input_length,
- int16_t* output,
- int* output_scale) const {
+ int16_t* output) const {
// Set parameters depending on sample rate.
const int16_t* filter_coefficients;
size_t num_coefficients;
@@ -814,13 +803,11 @@
downsampled_input, norm_shift);
int32_t correlation[kNumCorrelationLags];
- static const int kCorrelationShift = 6;
- WebRtcSpl_CrossCorrelation(
- correlation,
+ CrossCorrelationWithAutoShift(
&downsampled_input[kDownsampledLength - kCorrelationLength],
&downsampled_input[kDownsampledLength - kCorrelationLength
- kCorrelationStartLag],
- kCorrelationLength, kNumCorrelationLags, kCorrelationShift, -1);
+ kCorrelationLength, kNumCorrelationLags, -1, correlation);
// Normalize and move data from 32-bit to 16-bit vector.
int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
@@ -829,8 +816,6 @@
std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
norm_shift2);
- // Total scale factor (right shifts) of correlation value.
- *output_scale = 2 * norm_shift + kCorrelationShift + norm_shift2;
}
void Expand::UpdateLagIndex() {
diff --git a/webrtc/modules/audio_coding/neteq/expand.h b/webrtc/modules/audio_coding/neteq/expand.h
index 7f61bf3..44ced0a 100644
--- a/webrtc/modules/audio_coding/neteq/expand.h
+++ b/webrtc/modules/audio_coding/neteq/expand.h
@@ -120,12 +120,10 @@
// Calculate the auto-correlation of |input|, with length |input_length|
// samples. The correlation is calculated from a downsampled version of
- // |input|, and is written to |output|. The scale factor is written to
- // |output_scale|.
+ // |input|, and is written to |output|.
void Correlation(const int16_t* input,
size_t input_length,
- int16_t* output,
- int* output_scale) const;
+ int16_t* output) const;
void UpdateLagIndex();
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index 9aed91f..b62df61 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -18,6 +18,7 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
@@ -63,10 +64,8 @@
for (size_t channel = 0; channel < num_channels_; ++channel) {
int16_t* input_channel = &input_vector[channel][0];
int16_t* expanded_channel = &expanded_[channel][0];
- int16_t expanded_max, input_max;
int16_t new_mute_factor = SignalScaling(
- input_channel, input_length_per_channel, expanded_channel,
- &expanded_max, &input_max);
+ input_channel, input_length_per_channel, expanded_channel);
// Adjust muting factor (product of "main" muting factor and expand muting
// factor).
@@ -89,8 +88,7 @@
// Calculate the lag of the strongest correlation period.
best_correlation_index = CorrelateAndPeakSearch(
- expanded_max, input_max, old_length,
- input_length_per_channel, expand_period);
+ old_length, input_length_per_channel, expand_period);
}
static const int kTempDataSize = 3600;
@@ -204,19 +202,19 @@
}
int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
- const int16_t* expanded_signal,
- int16_t* expanded_max, int16_t* input_max) const {
+ const int16_t* expanded_signal) const {
// Adjust muting factor if new vector is more or less of the BGN energy.
const size_t mod_input_length =
std::min(static_cast<size_t>(64 * fs_mult_), input_length);
- *expanded_max = WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
- *input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
+ const int16_t expanded_max =
+ WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
+ const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
// Calculate energy of expanded signal.
// |log_fs_mult| is log2(fs_mult_), but is not exact for 48000 Hz.
int log_fs_mult = 30 - WebRtcSpl_NormW32(fs_mult_);
int expanded_shift = 6 + log_fs_mult
- - WebRtcSpl_NormW32(*expanded_max * *expanded_max);
+ - WebRtcSpl_NormW32(expanded_max * expanded_max);
expanded_shift = std::max(expanded_shift, 0);
int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
expanded_signal,
@@ -224,8 +222,7 @@
expanded_shift);
// Calculate energy of input signal.
- int input_shift = 6 + log_fs_mult -
- WebRtcSpl_NormW32(*input_max * *input_max);
+ int input_shift = 6 + log_fs_mult - WebRtcSpl_NormW32(input_max * input_max);
input_shift = std::max(input_shift, 0);
int32_t energy_input = WebRtcSpl_DotProductWithScale(input, input,
mod_input_length,
@@ -307,22 +304,17 @@
}
}
-size_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
- size_t start_position, size_t input_length,
+size_t Merge::CorrelateAndPeakSearch(size_t start_position, size_t input_length,
size_t expand_period) const {
// Calculate correlation without any normalization.
const size_t max_corr_length = kMaxCorrelationLength;
size_t stop_position_downsamp =
std::min(max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
- int correlation_shift = 0;
- if (expanded_max * input_max > 26843546) {
- correlation_shift = 3;
- }
int32_t correlation[kMaxCorrelationLength];
- WebRtcSpl_CrossCorrelation(correlation, input_downsampled_,
- expanded_downsampled_, kInputDownsampLength,
- stop_position_downsamp, correlation_shift, 1);
+ CrossCorrelationWithAutoShift(input_downsampled_, expanded_downsampled_,
+ kInputDownsampLength, stop_position_downsamp, 1,
+ correlation);
// Normalize correlation to 14 bits and copy to a 16-bit array.
const size_t pad_length = expand_->overlap_length() - 1;
diff --git a/webrtc/modules/audio_coding/neteq/merge.h b/webrtc/modules/audio_coding/neteq/merge.h
index a168502..95dea5a 100644
--- a/webrtc/modules/audio_coding/neteq/merge.h
+++ b/webrtc/modules/audio_coding/neteq/merge.h
@@ -69,11 +69,10 @@
// of samples that were taken from the |sync_buffer_|.
size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
- // Analyzes |input| and |expanded_signal| to find maximum values. Returns
- // a muting factor (Q14) to be used on the new data.
+ // Analyzes |input| and |expanded_signal| and returns muting factor (Q14) to
+ // be used on the new data.
int16_t SignalScaling(const int16_t* input, size_t input_length,
- const int16_t* expanded_signal,
- int16_t* expanded_max, int16_t* input_max) const;
+ const int16_t* expanded_signal) const;
// Downsamples |input| (|input_length| samples) and |expanded_signal| to
// 4 kHz sample rate. The downsampled signals are written to
@@ -84,8 +83,7 @@
// Calculates cross-correlation between |input_downsampled_| and
// |expanded_downsampled_|, and finds the correlation maximum. The maximizing
// lag is returned.
- size_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
- size_t start_position, size_t input_length,
+ size_t CorrelateAndPeakSearch(size_t start_position, size_t input_length,
size_t expand_period) const;
const int fs_mult_; // fs_hz_ / 8000.
diff --git a/webrtc/modules/audio_coding/neteq/neteq.gypi b/webrtc/modules/audio_coding/neteq/neteq.gypi
index fee51df..509dda0 100644
--- a/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -73,6 +73,8 @@
'buffer_level_filter.h',
'comfort_noise.cc',
'comfort_noise.h',
+ 'cross_correlation.cc',
+ 'cross_correlation.h',
'decision_logic.cc',
'decision_logic.h',
'decision_logic_fax.cc',
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index e982db2..ad7a201 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -460,14 +460,14 @@
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
const std::string output_checksum = PlatformChecksum(
- "f587883b7c371ee8d87dbf1b0f07525af7d959b8",
- "a349bd71dba548029b05d1d2a6dc7caafab9a856",
- "f587883b7c371ee8d87dbf1b0f07525af7d959b8",
- "08266b198e7686b3cd9330813e0d2cd72fc8fdc2");
+ "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
+ "d2a6b6ff54b340cf9f961c7f07768d86b3761073",
+ "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
+ "f9749813dbc3fb59dae761de518fec65b8407c5b");
const std::string network_stats_checksum = PlatformChecksum(
"2cf380a05ee07080bd72471e8ec7777a39644ec9",
- "2853ab577fe571adfc7b18f77bbe58f1253d2019",
+ "01be67dc4c3b8e74743a45cbd8684c0535dec9ad",
"2cf380a05ee07080bd72471e8ec7777a39644ec9",
"2cf380a05ee07080bd72471e8ec7777a39644ec9");
@@ -497,16 +497,16 @@
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string output_checksum = PlatformChecksum(
- "c23004d91ffbe5e7a1f24620fc89b58c0426040f",
- "c23004d91ffbe5e7a1f24620fc89b58c0426040f",
- "c23004d91ffbe5e7a1f24620fc89b58c0426040f",
- "c23004d91ffbe5e7a1f24620fc89b58c0426040f");
+ "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+ "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+ "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+ "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4");
const std::string network_stats_checksum = PlatformChecksum(
- "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
- "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
- "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
- "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb");
+ "6eab76efbde753d4dde38983445ca16b4ce59b39",
+ "6eab76efbde753d4dde38983445ca16b4ce59b39",
+ "6eab76efbde753d4dde38983445ca16b4ce59b39",
+ "6eab76efbde753d4dde38983445ca16b4ce59b39");
const std::string rtcp_stats_checksum = PlatformChecksum(
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
diff --git a/webrtc/modules/audio_coding/neteq/time_stretch.cc b/webrtc/modules/audio_coding/neteq/time_stretch.cc
index 6a91ea4..880b1f8 100644
--- a/webrtc/modules/audio_coding/neteq/time_stretch.cc
+++ b/webrtc/modules/audio_coding/neteq/time_stretch.cc
@@ -16,6 +16,7 @@
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
namespace webrtc {
@@ -158,20 +159,15 @@
}
void TimeStretch::AutoCorrelation() {
- // Set scaling factor for cross correlation to protect against overflow.
- int scaling = kLogCorrelationLen - WebRtcSpl_NormW32(
- max_input_value_ * max_input_value_);
- scaling = std::max(0, scaling);
-
// Calculate correlation from lag kMinLag to lag kMaxLag in 4 kHz domain.
int32_t auto_corr[kCorrelationLen];
- WebRtcSpl_CrossCorrelation(auto_corr, &downsampled_input_[kMaxLag],
- &downsampled_input_[kMaxLag - kMinLag],
- kCorrelationLen, kMaxLag - kMinLag, scaling, -1);
+ CrossCorrelationWithAutoShift(
+ &downsampled_input_[kMaxLag], &downsampled_input_[kMaxLag - kMinLag],
+ kCorrelationLen, kMaxLag - kMinLag, -1, auto_corr);
// Normalize correlation to 14 bits and write to |auto_correlation_|.
int32_t max_corr = WebRtcSpl_MaxAbsValueW32(auto_corr, kCorrelationLen);
- scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
+ int scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
WebRtcSpl_VectorBitShiftW32ToW16(auto_correlation_, kCorrelationLen,
auto_corr, scaling);
}
