Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for a hard-coded reference to change. This CL effectively reverts r6257 "Rename neteq4 folder to neteq". BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
366
webrtc/modules/audio_coding/neteq/merge.cc
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366
webrtc/modules/audio_coding/neteq/merge.cc
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/neteq/merge.h"
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#include <assert.h>
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#include <string.h> // memmove, memcpy, memset, size_t
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#include <algorithm> // min, max
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
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#include "webrtc/modules/audio_coding/neteq/expand.h"
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#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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int Merge::Process(int16_t* input, size_t input_length,
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int16_t* external_mute_factor_array,
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AudioMultiVector* output) {
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// TODO(hlundin): Change to an enumerator and skip assert.
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assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
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fs_hz_ == 48000);
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assert(fs_hz_ <= kMaxSampleRate); // Should not be possible.
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int old_length;
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int expand_period;
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// Get expansion data to overlap and mix with.
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int expanded_length = GetExpandedSignal(&old_length, &expand_period);
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// Transfer input signal to an AudioMultiVector.
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AudioMultiVector input_vector(num_channels_);
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input_vector.PushBackInterleaved(input, input_length);
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size_t input_length_per_channel = input_vector.Size();
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assert(input_length_per_channel == input_length / num_channels_);
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int16_t best_correlation_index = 0;
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size_t output_length = 0;
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for (size_t channel = 0; channel < num_channels_; ++channel) {
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int16_t* input_channel = &input_vector[channel][0];
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int16_t* expanded_channel = &expanded_[channel][0];
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int16_t expanded_max, input_max;
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int16_t new_mute_factor = SignalScaling(
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input_channel, static_cast<int>(input_length_per_channel),
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expanded_channel, &expanded_max, &input_max);
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// Adjust muting factor (product of "main" muting factor and expand muting
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// factor).
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int16_t* external_mute_factor = &external_mute_factor_array[channel];
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*external_mute_factor =
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(*external_mute_factor * expand_->MuteFactor(channel)) >> 14;
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// Update |external_mute_factor| if it is lower than |new_mute_factor|.
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if (new_mute_factor > *external_mute_factor) {
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*external_mute_factor = std::min(new_mute_factor,
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static_cast<int16_t>(16384));
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}
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if (channel == 0) {
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// Downsample, correlate, and find strongest correlation period for the
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// master (i.e., first) channel only.
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// Downsample to 4kHz sample rate.
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Downsample(input_channel, static_cast<int>(input_length_per_channel),
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expanded_channel, expanded_length);
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// Calculate the lag of the strongest correlation period.
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best_correlation_index = CorrelateAndPeakSearch(
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expanded_max, input_max, old_length,
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static_cast<int>(input_length_per_channel), expand_period);
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}
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static const int kTempDataSize = 3600;
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int16_t temp_data[kTempDataSize]; // TODO(hlundin) Remove this.
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int16_t* decoded_output = temp_data + best_correlation_index;
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// Mute the new decoded data if needed (and unmute it linearly).
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// This is the overlapping part of expanded_signal.
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int interpolation_length = std::min(
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kMaxCorrelationLength * fs_mult_,
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expanded_length - best_correlation_index);
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interpolation_length = std::min(interpolation_length,
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static_cast<int>(input_length_per_channel));
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if (*external_mute_factor < 16384) {
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// Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
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// and so on.
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int increment = 4194 / fs_mult_;
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*external_mute_factor = DspHelper::RampSignal(input_channel,
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interpolation_length,
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*external_mute_factor,
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increment);
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DspHelper::UnmuteSignal(&input_channel[interpolation_length],
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input_length_per_channel - interpolation_length,
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external_mute_factor, increment,
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&decoded_output[interpolation_length]);
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} else {
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// No muting needed.
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memmove(
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&decoded_output[interpolation_length],
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&input_channel[interpolation_length],
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sizeof(int16_t) * (input_length_per_channel - interpolation_length));
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}
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// Do overlap and mix linearly.
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int increment = 16384 / (interpolation_length + 1); // In Q14.
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int16_t mute_factor = 16384 - increment;
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memmove(temp_data, expanded_channel,
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sizeof(int16_t) * best_correlation_index);
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DspHelper::CrossFade(&expanded_channel[best_correlation_index],
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input_channel, interpolation_length,
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&mute_factor, increment, decoded_output);
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output_length = best_correlation_index + input_length_per_channel;
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if (channel == 0) {
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assert(output->Empty()); // Output should be empty at this point.
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output->AssertSize(output_length);
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} else {
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assert(output->Size() == output_length);
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}
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memcpy(&(*output)[channel][0], temp_data,
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sizeof(temp_data[0]) * output_length);
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}
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// Copy back the first part of the data to |sync_buffer_| and remove it from
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// |output|.
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sync_buffer_->ReplaceAtIndex(*output, old_length, sync_buffer_->next_index());
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output->PopFront(old_length);
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// Return new added length. |old_length| samples were borrowed from
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// |sync_buffer_|.
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return static_cast<int>(output_length) - old_length;
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}
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int Merge::GetExpandedSignal(int* old_length, int* expand_period) {
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// Check how much data that is left since earlier.
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*old_length = static_cast<int>(sync_buffer_->FutureLength());
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// Should never be less than overlap_length.
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assert(*old_length >= static_cast<int>(expand_->overlap_length()));
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// Generate data to merge the overlap with using expand.
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expand_->SetParametersForMergeAfterExpand();
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if (*old_length >= 210 * kMaxSampleRate / 8000) {
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// TODO(hlundin): Write test case for this.
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// The number of samples available in the sync buffer is more than what fits
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// in expanded_signal. Keep the first 210 * kMaxSampleRate / 8000 samples,
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// but shift them towards the end of the buffer. This is ok, since all of
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// the buffer will be expand data anyway, so as long as the beginning is
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// left untouched, we're fine.
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int16_t length_diff = *old_length - 210 * kMaxSampleRate / 8000;
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sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index());
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*old_length = 210 * kMaxSampleRate / 8000;
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// This is the truncated length.
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}
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// This assert should always be true thanks to the if statement above.
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assert(210 * kMaxSampleRate / 8000 - *old_length >= 0);
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AudioMultiVector expanded_temp(num_channels_);
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expand_->Process(&expanded_temp);
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*expand_period = static_cast<int>(expanded_temp.Size()); // Samples per
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// channel.
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expanded_.Clear();
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// Copy what is left since earlier into the expanded vector.
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expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index());
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assert(expanded_.Size() == static_cast<size_t>(*old_length));
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assert(expanded_temp.Size() > 0);
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// Do "ugly" copy and paste from the expanded in order to generate more data
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// to correlate (but not interpolate) with.
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const int required_length = (120 + 80 + 2) * fs_mult_;
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if (expanded_.Size() < static_cast<size_t>(required_length)) {
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while (expanded_.Size() < static_cast<size_t>(required_length)) {
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// Append one more pitch period each time.
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expanded_.PushBack(expanded_temp);
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}
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// Trim the length to exactly |required_length|.
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expanded_.PopBack(expanded_.Size() - required_length);
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}
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assert(expanded_.Size() >= static_cast<size_t>(required_length));
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return required_length;
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}
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int16_t Merge::SignalScaling(const int16_t* input, int input_length,
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const int16_t* expanded_signal,
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int16_t* expanded_max, int16_t* input_max) const {
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// Adjust muting factor if new vector is more or less of the BGN energy.
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const int mod_input_length = std::min(64 * fs_mult_, input_length);
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*expanded_max = WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
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*input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
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// Calculate energy of expanded signal.
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// |log_fs_mult| is log2(fs_mult_), but is not exact for 48000 Hz.
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int log_fs_mult = 30 - WebRtcSpl_NormW32(fs_mult_);
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int expanded_shift = 6 + log_fs_mult
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- WebRtcSpl_NormW32(*expanded_max * *expanded_max);
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expanded_shift = std::max(expanded_shift, 0);
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int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
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expanded_signal,
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mod_input_length,
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expanded_shift);
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// Calculate energy of input signal.
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int input_shift = 6 + log_fs_mult -
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WebRtcSpl_NormW32(*input_max * *input_max);
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input_shift = std::max(input_shift, 0);
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int32_t energy_input = WebRtcSpl_DotProductWithScale(input, input,
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mod_input_length,
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input_shift);
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// Align to the same Q-domain.
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if (input_shift > expanded_shift) {
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energy_expanded = energy_expanded >> (input_shift - expanded_shift);
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} else {
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energy_input = energy_input >> (expanded_shift - input_shift);
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}
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// Calculate muting factor to use for new frame.
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int16_t mute_factor;
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if (energy_input > energy_expanded) {
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// Normalize |energy_input| to 14 bits.
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int16_t temp_shift = WebRtcSpl_NormW32(energy_input) - 17;
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energy_input = WEBRTC_SPL_SHIFT_W32(energy_input, temp_shift);
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// Put |energy_expanded| in a domain 14 higher, so that
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// energy_expanded / energy_input is in Q14.
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energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
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// Calculate sqrt(energy_expanded / energy_input) in Q14.
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mute_factor = WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14);
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} else {
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// Set to 1 (in Q14) when |expanded| has higher energy than |input|.
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mute_factor = 16384;
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}
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return mute_factor;
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}
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// TODO(hlundin): There are some parameter values in this method that seem
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// strange. Compare with Expand::Correlation.
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void Merge::Downsample(const int16_t* input, int input_length,
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const int16_t* expanded_signal, int expanded_length) {
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const int16_t* filter_coefficients;
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int num_coefficients;
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int decimation_factor = fs_hz_ / 4000;
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static const int kCompensateDelay = 0;
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int length_limit = fs_hz_ / 100; // 10 ms in samples.
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if (fs_hz_ == 8000) {
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filter_coefficients = DspHelper::kDownsample8kHzTbl;
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num_coefficients = 3;
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} else if (fs_hz_ == 16000) {
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filter_coefficients = DspHelper::kDownsample16kHzTbl;
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num_coefficients = 5;
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} else if (fs_hz_ == 32000) {
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filter_coefficients = DspHelper::kDownsample32kHzTbl;
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num_coefficients = 7;
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} else { // fs_hz_ == 48000
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filter_coefficients = DspHelper::kDownsample48kHzTbl;
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num_coefficients = 7;
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}
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int signal_offset = num_coefficients - 1;
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WebRtcSpl_DownsampleFast(&expanded_signal[signal_offset],
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expanded_length - signal_offset,
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expanded_downsampled_, kExpandDownsampLength,
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filter_coefficients, num_coefficients,
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decimation_factor, kCompensateDelay);
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if (input_length <= length_limit) {
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// Not quite long enough, so we have to cheat a bit.
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int16_t temp_len = input_length - signal_offset;
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// TODO(hlundin): Should |downsamp_temp_len| be corrected for round-off
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// errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
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int16_t downsamp_temp_len = temp_len / decimation_factor;
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WebRtcSpl_DownsampleFast(&input[signal_offset], temp_len,
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input_downsampled_, downsamp_temp_len,
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filter_coefficients, num_coefficients,
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decimation_factor, kCompensateDelay);
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memset(&input_downsampled_[downsamp_temp_len], 0,
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sizeof(int16_t) * (kInputDownsampLength - downsamp_temp_len));
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} else {
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WebRtcSpl_DownsampleFast(&input[signal_offset],
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input_length - signal_offset, input_downsampled_,
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kInputDownsampLength, filter_coefficients,
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num_coefficients, decimation_factor,
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kCompensateDelay);
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}
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}
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int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
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int start_position, int input_length,
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int expand_period) const {
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// Calculate correlation without any normalization.
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const int max_corr_length = kMaxCorrelationLength;
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int stop_position_downsamp = std::min(
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max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
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int16_t correlation_shift = 0;
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if (expanded_max * input_max > 26843546) {
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correlation_shift = 3;
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}
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int32_t correlation[kMaxCorrelationLength];
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WebRtcSpl_CrossCorrelation(correlation, input_downsampled_,
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expanded_downsampled_, kInputDownsampLength,
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stop_position_downsamp, correlation_shift, 1);
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// Normalize correlation to 14 bits and copy to a 16-bit array.
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const int pad_length = static_cast<int>(expand_->overlap_length() - 1);
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const int correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
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scoped_ptr<int16_t[]> correlation16(new int16_t[correlation_buffer_size]);
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memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
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int16_t* correlation_ptr = &correlation16[pad_length];
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int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
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stop_position_downsamp);
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int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
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WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
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correlation, norm_shift);
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// Calculate allowed starting point for peak finding.
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// The peak location bestIndex must fulfill two criteria:
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// (1) w16_bestIndex + input_length <
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// timestamps_per_call_ + expand_->overlap_length();
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// (2) w16_bestIndex + input_length < start_position.
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int start_index = timestamps_per_call_ +
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static_cast<int>(expand_->overlap_length());
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start_index = std::max(start_position, start_index);
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start_index = std::max(start_index - input_length, 0);
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// Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
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int start_index_downsamp = start_index / (fs_mult_ * 2);
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// Calculate a modified |stop_position_downsamp| to account for the increased
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// start index |start_index_downsamp| and the effective array length.
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int modified_stop_pos =
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std::min(stop_position_downsamp,
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kMaxCorrelationLength + pad_length - start_index_downsamp);
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int best_correlation_index;
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int16_t best_correlation;
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static const int kNumCorrelationCandidates = 1;
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DspHelper::PeakDetection(&correlation_ptr[start_index_downsamp],
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modified_stop_pos, kNumCorrelationCandidates,
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fs_mult_, &best_correlation_index,
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&best_correlation);
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// Compensate for modified start index.
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best_correlation_index += start_index;
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// Ensure that underrun does not occur for 10ms case => we have to get at
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// least 10ms + overlap . (This should never happen thanks to the above
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// modification of peak-finding starting point.)
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while ((best_correlation_index + input_length) <
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static_cast<int>(timestamps_per_call_ + expand_->overlap_length()) ||
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best_correlation_index + input_length < start_position) {
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assert(false); // Should never happen.
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best_correlation_index += expand_period; // Jump one lag ahead.
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}
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return best_correlation_index;
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}
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int Merge::RequiredFutureSamples() {
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return static_cast<int>(fs_hz_ / 100 * num_channels_); // 10 ms.
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}
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} // namespace webrtc
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