/*
* 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 "modules/audio_coding/neteq/merge.h"
#include <assert.h>
#include <string.h> // memmove, memcpy, memset, size_t
#include <algorithm> // min, max
#include <memory>
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/cross_correlation.h"
#include "modules/audio_coding/neteq/dsp_helper.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
Merge::Merge(int fs_hz,
size_t num_channels,
Expand* expand,
SyncBuffer* sync_buffer)
: fs_hz_(fs_hz),
num_channels_(num_channels),
fs_mult_(fs_hz_ / 8000),
timestamps_per_call_(static_cast<size_t>(fs_hz_ / 100)),
expand_(expand),
sync_buffer_(sync_buffer),
expanded_(num_channels_) {
assert(num_channels_ > 0);
}
Merge::~Merge() = default;
size_t Merge::Process(int16_t* input,
size_t input_length,
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.
size_t old_length;
size_t expand_period;
// Get expansion data to overlap and mix with.
size_t expanded_length = GetExpandedSignal(&old_length, &expand_period);
// Transfer input signal to an AudioMultiVector.
AudioMultiVector input_vector(num_channels_);
input_vector.PushBackInterleaved(
rtc::ArrayView<const int16_t>(input, input_length));
size_t input_length_per_channel = input_vector.Size();
assert(input_length_per_channel == input_length / num_channels_);
size_t best_correlation_index = 0;
size_t output_length = 0;
std::unique_ptr<int16_t[]> input_channel(
new int16_t[input_length_per_channel]);
std::unique_ptr<int16_t[]> expanded_channel(new int16_t[expanded_length]);
for (size_t channel = 0; channel < num_channels_; ++channel) {
input_vector[channel].CopyTo(input_length_per_channel, 0,
input_channel.get());
expanded_[channel].CopyTo(expanded_length, 0, expanded_channel.get());
const int16_t new_mute_factor = std::min<int16_t>(
16384, SignalScaling(input_channel.get(), input_length_per_channel,
expanded_channel.get()));
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.get(), input_length_per_channel,
expanded_channel.get(), expanded_length);
// Calculate the lag of the strongest correlation period.
best_correlation_index = CorrelateAndPeakSearch(
old_length, input_length_per_channel, expand_period);
}
temp_data_.resize(input_length_per_channel + best_correlation_index);
int16_t* decoded_output = temp_data_.data() + best_correlation_index;
// Mute the new decoded data if needed (and unmute it linearly).
// This is the overlapping part of expanded_signal.
size_t interpolation_length =
std::min(kMaxCorrelationLength * fs_mult_,
expanded_length - best_correlation_index);
interpolation_length =
std::min(interpolation_length, input_length_per_channel);
RTC_DCHECK_LE(new_mute_factor, 16384);
int16_t mute_factor =
std::max(expand_->MuteFactor(channel), new_mute_factor);
RTC_DCHECK_GE(mute_factor, 0);
if (mute_factor < 16384) {
// Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
// and so on, or as fast as it takes to come back to full gain within the
// frame length.
const int back_to_fullscale_inc = static_cast<int>(
((16384 - mute_factor) << 6) / input_length_per_channel);
const int increment = std::max(4194 / fs_mult_, back_to_fullscale_inc);
mute_factor = static_cast<int16_t>(DspHelper::RampSignal(
input_channel.get(), interpolation_length, mute_factor, increment));
DspHelper::UnmuteSignal(&input_channel[interpolation_length],
input_length_per_channel - interpolation_length,
&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.
int16_t increment =
static_cast<int16_t>(16384 / (interpolation_length + 1)); // In Q14.
int16_t local_mute_factor = 16384 - increment;
memmove(temp_data_.data(), expanded_channel.get(),
sizeof(int16_t) * best_correlation_index);
DspHelper::CrossFade(&expanded_channel[best_correlation_index],
input_channel.get(), interpolation_length,
&local_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);
}
(*output)[channel].OverwriteAt(temp_data_.data(), output_length, 0);
}
// 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_|.
RTC_DCHECK_GE(output_length, old_length);
return output_length - old_length;
}
size_t Merge::GetExpandedSignal(size_t* old_length, size_t* expand_period) {
// Check how much data that is left since earlier.
*old_length = sync_buffer_->FutureLength();
// Should never be less than overlap_length.
assert(*old_length >= 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.
size_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);
AudioMultiVector expanded_temp(num_channels_);
expand_->Process(&expanded_temp);
*expand_period = 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() == *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 size_t required_length = static_cast<size_t>((120 + 80 + 2) * fs_mult_);
if (expanded_.Size() < required_length) {
while (expanded_.Size() < 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() >= required_length);
return required_length;
}
int16_t Merge::SignalScaling(const int16_t* input,
size_t input_length,
const int16_t* expanded_signal) const {
// Adjust muting factor if new vector is more or less of the BGN energy.
const auto mod_input_length = rtc::SafeMin<size_t>(
64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
const int16_t expanded_max =
WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
int32_t factor =
(expanded_max * expanded_max) / (std::numeric_limits<int32_t>::max() /
static_cast<int32_t>(mod_input_length));
const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
int32_t energy_expanded = WebRtcSpl_DotProductWithScale(
expanded_signal, expanded_signal, mod_input_length, expanded_shift);
// Calculate energy of input signal.
const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
factor = (input_max * input_max) / (std::numeric_limits<int32_t>::max() /
static_cast<int32_t>(mod_input_length));
const int input_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
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 = static_cast<int16_t>(
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,
size_t input_length,
const int16_t* expanded_signal,
size_t expanded_length) {
const int16_t* filter_coefficients;
size_t num_coefficients;
int decimation_factor = fs_hz_ / 4000;
static const size_t kCompensateDelay = 0;
size_t length_limit = static_cast<size_t>(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;
}
size_t 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.
// If the input is shorter than the offset, we consider the input to be 0
// length. This will cause us to skip the downsampling since it makes no
// sense anyway, and input_downsampled_ will be filled with zeros. This is
// clearly a pathological case, and the signal quality will suffer, but
// there is not much we can do.
const size_t temp_len =
input_length > signal_offset ? input_length - signal_offset : 0;
// TODO(hlundin): Should |downsamp_temp_len| be corrected for round-off
// errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
size_t downsamp_temp_len = temp_len / decimation_factor;
if (downsamp_temp_len > 0) {
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);
}
}
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);
int32_t correlation[kMaxCorrelationLength];
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;
const size_t correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
std::unique_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);
int 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.
size_t start_index = timestamps_per_call_ + expand_->overlap_length();
start_index = std::max(start_position, start_index);
start_index = (input_length > start_index) ? 0 : (start_index - input_length);
// Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
size_t 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.
size_t modified_stop_pos =
std::min(stop_position_downsamp,
kMaxCorrelationLength + pad_length - start_index_downsamp);
size_t best_correlation_index;
int16_t best_correlation;
static const size_t 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) <
(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;
}
size_t Merge::RequiredFutureSamples() {
return fs_hz_ / 100 * num_channels_; // 10 ms.
}
} // namespace webrtc