mirror of
https://github.com/oxen-io/session-android.git
synced 2024-11-24 18:45:19 +00:00
367 lines
16 KiB
C++
367 lines
16 KiB
C++
/*
|
|
* Copyright (c) 2012 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/merge.h"
|
|
|
|
#include <assert.h>
|
|
#include <string.h> // memmove, memcpy, memset, size_t
|
|
|
|
#include <algorithm> // min, max
|
|
|
|
#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/dsp_helper.h"
|
|
#include "webrtc/modules/audio_coding/neteq/expand.h"
|
|
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
|
|
|
namespace webrtc {
|
|
|
|
int Merge::Process(int16_t* input, size_t input_length,
|
|
int16_t* external_mute_factor_array,
|
|
AudioMultiVector* output) {
|
|
// TODO(hlundin): Change to an enumerator and skip assert.
|
|
assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
|
|
fs_hz_ == 48000);
|
|
assert(fs_hz_ <= kMaxSampleRate); // Should not be possible.
|
|
|
|
int old_length;
|
|
int expand_period;
|
|
// Get expansion data to overlap and mix with.
|
|
int expanded_length = GetExpandedSignal(&old_length, &expand_period);
|
|
|
|
// Transfer input signal to an AudioMultiVector.
|
|
AudioMultiVector input_vector(num_channels_);
|
|
input_vector.PushBackInterleaved(input, input_length);
|
|
size_t input_length_per_channel = input_vector.Size();
|
|
assert(input_length_per_channel == input_length / num_channels_);
|
|
|
|
int16_t best_correlation_index = 0;
|
|
size_t output_length = 0;
|
|
|
|
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, static_cast<int>(input_length_per_channel),
|
|
expanded_channel, &expanded_max, &input_max);
|
|
|
|
// Adjust muting factor (product of "main" muting factor and expand muting
|
|
// factor).
|
|
int16_t* external_mute_factor = &external_mute_factor_array[channel];
|
|
*external_mute_factor =
|
|
(*external_mute_factor * expand_->MuteFactor(channel)) >> 14;
|
|
|
|
// Update |external_mute_factor| if it is lower than |new_mute_factor|.
|
|
if (new_mute_factor > *external_mute_factor) {
|
|
*external_mute_factor = std::min(new_mute_factor,
|
|
static_cast<int16_t>(16384));
|
|
}
|
|
|
|
if (channel == 0) {
|
|
// Downsample, correlate, and find strongest correlation period for the
|
|
// master (i.e., first) channel only.
|
|
// Downsample to 4kHz sample rate.
|
|
Downsample(input_channel, static_cast<int>(input_length_per_channel),
|
|
expanded_channel, expanded_length);
|
|
|
|
// Calculate the lag of the strongest correlation period.
|
|
best_correlation_index = CorrelateAndPeakSearch(
|
|
expanded_max, input_max, old_length,
|
|
static_cast<int>(input_length_per_channel), expand_period);
|
|
}
|
|
|
|
static const int kTempDataSize = 3600;
|
|
int16_t temp_data[kTempDataSize]; // TODO(hlundin) Remove this.
|
|
int16_t* decoded_output = temp_data + best_correlation_index;
|
|
|
|
// Mute the new decoded data if needed (and unmute it linearly).
|
|
// This is the overlapping part of expanded_signal.
|
|
int interpolation_length = std::min(
|
|
kMaxCorrelationLength * fs_mult_,
|
|
expanded_length - best_correlation_index);
|
|
interpolation_length = std::min(interpolation_length,
|
|
static_cast<int>(input_length_per_channel));
|
|
if (*external_mute_factor < 16384) {
|
|
// Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
|
|
// and so on.
|
|
int increment = 4194 / fs_mult_;
|
|
*external_mute_factor = DspHelper::RampSignal(input_channel,
|
|
interpolation_length,
|
|
*external_mute_factor,
|
|
increment);
|
|
DspHelper::UnmuteSignal(&input_channel[interpolation_length],
|
|
input_length_per_channel - interpolation_length,
|
|
external_mute_factor, increment,
|
|
&decoded_output[interpolation_length]);
|
|
} else {
|
|
// No muting needed.
|
|
memmove(
|
|
&decoded_output[interpolation_length],
|
|
&input_channel[interpolation_length],
|
|
sizeof(int16_t) * (input_length_per_channel - interpolation_length));
|
|
}
|
|
|
|
// Do overlap and mix linearly.
|
|
int increment = 16384 / (interpolation_length + 1); // In Q14.
|
|
int16_t mute_factor = 16384 - increment;
|
|
memmove(temp_data, expanded_channel,
|
|
sizeof(int16_t) * best_correlation_index);
|
|
DspHelper::CrossFade(&expanded_channel[best_correlation_index],
|
|
input_channel, interpolation_length,
|
|
&mute_factor, increment, decoded_output);
|
|
|
|
output_length = best_correlation_index + input_length_per_channel;
|
|
if (channel == 0) {
|
|
assert(output->Empty()); // Output should be empty at this point.
|
|
output->AssertSize(output_length);
|
|
} else {
|
|
assert(output->Size() == output_length);
|
|
}
|
|
memcpy(&(*output)[channel][0], temp_data,
|
|
sizeof(temp_data[0]) * output_length);
|
|
}
|
|
|
|
// Copy back the first part of the data to |sync_buffer_| and remove it from
|
|
// |output|.
|
|
sync_buffer_->ReplaceAtIndex(*output, old_length, sync_buffer_->next_index());
|
|
output->PopFront(old_length);
|
|
|
|
// Return new added length. |old_length| samples were borrowed from
|
|
// |sync_buffer_|.
|
|
return static_cast<int>(output_length) - old_length;
|
|
}
|
|
|
|
int Merge::GetExpandedSignal(int* old_length, int* expand_period) {
|
|
// Check how much data that is left since earlier.
|
|
*old_length = static_cast<int>(sync_buffer_->FutureLength());
|
|
// Should never be less than overlap_length.
|
|
assert(*old_length >= static_cast<int>(expand_->overlap_length()));
|
|
// Generate data to merge the overlap with using expand.
|
|
expand_->SetParametersForMergeAfterExpand();
|
|
|
|
if (*old_length >= 210 * kMaxSampleRate / 8000) {
|
|
// TODO(hlundin): Write test case for this.
|
|
// The number of samples available in the sync buffer is more than what fits
|
|
// in expanded_signal. Keep the first 210 * kMaxSampleRate / 8000 samples,
|
|
// but shift them towards the end of the buffer. This is ok, since all of
|
|
// the buffer will be expand data anyway, so as long as the beginning is
|
|
// left untouched, we're fine.
|
|
int16_t length_diff = *old_length - 210 * kMaxSampleRate / 8000;
|
|
sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index());
|
|
*old_length = 210 * kMaxSampleRate / 8000;
|
|
// This is the truncated length.
|
|
}
|
|
// This assert should always be true thanks to the if statement above.
|
|
assert(210 * kMaxSampleRate / 8000 - *old_length >= 0);
|
|
|
|
AudioMultiVector expanded_temp(num_channels_);
|
|
expand_->Process(&expanded_temp);
|
|
*expand_period = static_cast<int>(expanded_temp.Size()); // Samples per
|
|
// channel.
|
|
|
|
expanded_.Clear();
|
|
// Copy what is left since earlier into the expanded vector.
|
|
expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index());
|
|
assert(expanded_.Size() == static_cast<size_t>(*old_length));
|
|
assert(expanded_temp.Size() > 0);
|
|
// Do "ugly" copy and paste from the expanded in order to generate more data
|
|
// to correlate (but not interpolate) with.
|
|
const int required_length = (120 + 80 + 2) * fs_mult_;
|
|
if (expanded_.Size() < static_cast<size_t>(required_length)) {
|
|
while (expanded_.Size() < static_cast<size_t>(required_length)) {
|
|
// Append one more pitch period each time.
|
|
expanded_.PushBack(expanded_temp);
|
|
}
|
|
// Trim the length to exactly |required_length|.
|
|
expanded_.PopBack(expanded_.Size() - required_length);
|
|
}
|
|
assert(expanded_.Size() >= static_cast<size_t>(required_length));
|
|
return required_length;
|
|
}
|
|
|
|
int16_t Merge::SignalScaling(const int16_t* input, int input_length,
|
|
const int16_t* expanded_signal,
|
|
int16_t* expanded_max, int16_t* input_max) const {
|
|
// Adjust muting factor if new vector is more or less of the BGN energy.
|
|
const int mod_input_length = std::min(64 * fs_mult_, input_length);
|
|
*expanded_max = WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
|
|
*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);
|
|
expanded_shift = std::max(expanded_shift, 0);
|
|
int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
|
|
expanded_signal,
|
|
mod_input_length,
|
|
expanded_shift);
|
|
|
|
// Calculate energy of input signal.
|
|
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,
|
|
input_shift);
|
|
|
|
// Align to the same Q-domain.
|
|
if (input_shift > expanded_shift) {
|
|
energy_expanded = energy_expanded >> (input_shift - expanded_shift);
|
|
} else {
|
|
energy_input = energy_input >> (expanded_shift - input_shift);
|
|
}
|
|
|
|
// Calculate muting factor to use for new frame.
|
|
int16_t mute_factor;
|
|
if (energy_input > energy_expanded) {
|
|
// Normalize |energy_input| to 14 bits.
|
|
int16_t temp_shift = WebRtcSpl_NormW32(energy_input) - 17;
|
|
energy_input = WEBRTC_SPL_SHIFT_W32(energy_input, temp_shift);
|
|
// Put |energy_expanded| in a domain 14 higher, so that
|
|
// energy_expanded / energy_input is in Q14.
|
|
energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
|
|
// Calculate sqrt(energy_expanded / energy_input) in Q14.
|
|
mute_factor = WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14);
|
|
} else {
|
|
// Set to 1 (in Q14) when |expanded| has higher energy than |input|.
|
|
mute_factor = 16384;
|
|
}
|
|
|
|
return mute_factor;
|
|
}
|
|
|
|
// TODO(hlundin): There are some parameter values in this method that seem
|
|
// strange. Compare with Expand::Correlation.
|
|
void Merge::Downsample(const int16_t* input, int input_length,
|
|
const int16_t* expanded_signal, int expanded_length) {
|
|
const int16_t* filter_coefficients;
|
|
int num_coefficients;
|
|
int decimation_factor = fs_hz_ / 4000;
|
|
static const int kCompensateDelay = 0;
|
|
int length_limit = fs_hz_ / 100; // 10 ms in samples.
|
|
if (fs_hz_ == 8000) {
|
|
filter_coefficients = DspHelper::kDownsample8kHzTbl;
|
|
num_coefficients = 3;
|
|
} else if (fs_hz_ == 16000) {
|
|
filter_coefficients = DspHelper::kDownsample16kHzTbl;
|
|
num_coefficients = 5;
|
|
} else if (fs_hz_ == 32000) {
|
|
filter_coefficients = DspHelper::kDownsample32kHzTbl;
|
|
num_coefficients = 7;
|
|
} else { // fs_hz_ == 48000
|
|
filter_coefficients = DspHelper::kDownsample48kHzTbl;
|
|
num_coefficients = 7;
|
|
}
|
|
int signal_offset = num_coefficients - 1;
|
|
WebRtcSpl_DownsampleFast(&expanded_signal[signal_offset],
|
|
expanded_length - signal_offset,
|
|
expanded_downsampled_, kExpandDownsampLength,
|
|
filter_coefficients, num_coefficients,
|
|
decimation_factor, kCompensateDelay);
|
|
if (input_length <= length_limit) {
|
|
// Not quite long enough, so we have to cheat a bit.
|
|
int16_t temp_len = input_length - signal_offset;
|
|
// TODO(hlundin): Should |downsamp_temp_len| be corrected for round-off
|
|
// errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
|
|
int16_t downsamp_temp_len = temp_len / decimation_factor;
|
|
WebRtcSpl_DownsampleFast(&input[signal_offset], temp_len,
|
|
input_downsampled_, downsamp_temp_len,
|
|
filter_coefficients, num_coefficients,
|
|
decimation_factor, kCompensateDelay);
|
|
memset(&input_downsampled_[downsamp_temp_len], 0,
|
|
sizeof(int16_t) * (kInputDownsampLength - downsamp_temp_len));
|
|
} else {
|
|
WebRtcSpl_DownsampleFast(&input[signal_offset],
|
|
input_length - signal_offset, input_downsampled_,
|
|
kInputDownsampLength, filter_coefficients,
|
|
num_coefficients, decimation_factor,
|
|
kCompensateDelay);
|
|
}
|
|
}
|
|
|
|
int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
|
|
int start_position, int input_length,
|
|
int expand_period) const {
|
|
// Calculate correlation without any normalization.
|
|
const int max_corr_length = kMaxCorrelationLength;
|
|
int stop_position_downsamp = std::min(
|
|
max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
|
|
int16_t 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);
|
|
|
|
// Normalize correlation to 14 bits and copy to a 16-bit array.
|
|
const int pad_length = static_cast<int>(expand_->overlap_length() - 1);
|
|
const int correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
|
|
scoped_ptr<int16_t[]> correlation16(new int16_t[correlation_buffer_size]);
|
|
memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
|
|
int16_t* correlation_ptr = &correlation16[pad_length];
|
|
int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
|
|
stop_position_downsamp);
|
|
int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
|
|
WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
|
|
correlation, norm_shift);
|
|
|
|
// Calculate allowed starting point for peak finding.
|
|
// The peak location bestIndex must fulfill two criteria:
|
|
// (1) w16_bestIndex + input_length <
|
|
// timestamps_per_call_ + expand_->overlap_length();
|
|
// (2) w16_bestIndex + input_length < start_position.
|
|
int start_index = timestamps_per_call_ +
|
|
static_cast<int>(expand_->overlap_length());
|
|
start_index = std::max(start_position, start_index);
|
|
start_index = std::max(start_index - input_length, 0);
|
|
// Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
|
|
int start_index_downsamp = start_index / (fs_mult_ * 2);
|
|
|
|
// Calculate a modified |stop_position_downsamp| to account for the increased
|
|
// start index |start_index_downsamp| and the effective array length.
|
|
int modified_stop_pos =
|
|
std::min(stop_position_downsamp,
|
|
kMaxCorrelationLength + pad_length - start_index_downsamp);
|
|
int best_correlation_index;
|
|
int16_t best_correlation;
|
|
static const int kNumCorrelationCandidates = 1;
|
|
DspHelper::PeakDetection(&correlation_ptr[start_index_downsamp],
|
|
modified_stop_pos, kNumCorrelationCandidates,
|
|
fs_mult_, &best_correlation_index,
|
|
&best_correlation);
|
|
// Compensate for modified start index.
|
|
best_correlation_index += start_index;
|
|
|
|
// Ensure that underrun does not occur for 10ms case => we have to get at
|
|
// least 10ms + overlap . (This should never happen thanks to the above
|
|
// modification of peak-finding starting point.)
|
|
while ((best_correlation_index + input_length) <
|
|
static_cast<int>(timestamps_per_call_ + expand_->overlap_length()) ||
|
|
best_correlation_index + input_length < start_position) {
|
|
assert(false); // Should never happen.
|
|
best_correlation_index += expand_period; // Jump one lag ahead.
|
|
}
|
|
return best_correlation_index;
|
|
}
|
|
|
|
int Merge::RequiredFutureSamples() {
|
|
return static_cast<int>(fs_hz_ / 100 * num_channels_); // 10 ms.
|
|
}
|
|
|
|
|
|
} // namespace webrtc
|