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Rename neteq4 folder to neteq

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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
2014-06-09 08:10:28 +00:00
parent 31f967c611
commit 9c55f0f957
158 changed files with 547 additions and 511 deletions
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190
webrtc/modules/audio_coding/neteq/normal.cc Normal file
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/*
* 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/normal.h"
#include <string.h> // memset, memcpy
#include <algorithm> // min
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
namespace webrtc {
int Normal::Process(const int16_t* input,
size_t length,
Modes last_mode,
int16_t* external_mute_factor_array,
AudioMultiVector* output) {
if (length == 0) {
// Nothing to process.
output->Clear();
return static_cast<int>(length);
}
assert(output->Empty());
// Output should be empty at this point.
output->PushBackInterleaved(input, length);
int16_t* signal = &(*output)[0][0];
const unsigned fs_mult = fs_hz_ / 8000;
assert(fs_mult > 0);
// fs_shift = log2(fs_mult), rounded down.
// Note that |fs_shift| is not "exact" for 48 kHz.
// TODO(hlundin): Investigate this further.
const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
// Check if last RecOut call resulted in an Expand. If so, we have to take
// care of some cross-fading and unmuting.
if (last_mode == kModeExpand) {
// Generate interpolation data using Expand.
// First, set Expand parameters to appropriate values.
expand_->SetParametersForNormalAfterExpand();
// Call Expand.
AudioMultiVector expanded(output->Channels());
expand_->Process(&expanded);
expand_->Reset();
for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
// Adjust muting factor (main muting factor times expand muting factor).
external_mute_factor_array[channel_ix] = static_cast<int16_t>(
WEBRTC_SPL_MUL_16_16_RSFT(external_mute_factor_array[channel_ix],
expand_->MuteFactor(channel_ix), 14));
int16_t* signal = &(*output)[channel_ix][0];
size_t length_per_channel = length / output->Channels();
// Find largest absolute value in new data.
int16_t decoded_max = WebRtcSpl_MaxAbsValueW16(
signal, static_cast<int>(length_per_channel));
// Adjust muting factor if needed (to BGN level).
int energy_length = std::min(static_cast<int>(fs_mult * 64),
static_cast<int>(length_per_channel));
int scaling = 6 + fs_shift
- WebRtcSpl_NormW32(decoded_max * decoded_max);
scaling = std::max(scaling, 0); // |scaling| should always be >= 0.
int32_t energy = WebRtcSpl_DotProductWithScale(signal, signal,
energy_length, scaling);
energy = energy / (energy_length >> scaling);
int mute_factor;
if ((energy != 0) &&
(energy > background_noise_.Energy(channel_ix))) {
// Normalize new frame energy to 15 bits.
scaling = WebRtcSpl_NormW32(energy) - 16;
// We want background_noise_.energy() / energy in Q14.
int32_t bgn_energy =
background_noise_.Energy(channel_ix) << (scaling+14);
int16_t energy_scaled = energy << scaling;
int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
mute_factor = WebRtcSpl_SqrtFloor(static_cast<int32_t>(ratio) << 14);
} else {
mute_factor = 16384; // 1.0 in Q14.
}
if (mute_factor > external_mute_factor_array[channel_ix]) {
external_mute_factor_array[channel_ix] = std::min(mute_factor, 16384);
}
// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
int16_t increment = 64 / fs_mult;
for (size_t i = 0; i < length_per_channel; i++) {
// Scale with mute factor.
assert(channel_ix < output->Channels());
assert(i < output->Size());
int32_t scaled_signal = (*output)[channel_ix][i] *
external_mute_factor_array[channel_ix];
// Shift 14 with proper rounding.
(*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
// Increase mute_factor towards 16384.
external_mute_factor_array[channel_ix] =
std::min(external_mute_factor_array[channel_ix] + increment, 16384);
}
// Interpolate the expanded data into the new vector.
// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
assert(fs_shift < 3); // Will always be 0, 1, or, 2.
increment = 4 >> fs_shift;
int fraction = increment;
for (size_t i = 0; i < 8 * fs_mult; i++) {
// TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
// now for legacy bit-exactness.
assert(channel_ix < output->Channels());
assert(i < output->Size());
(*output)[channel_ix][i] =
(fraction * (*output)[channel_ix][i] +
(32 - fraction) * expanded[channel_ix][i] + 8) >> 5;
fraction += increment;
}
}
} else if (last_mode == kModeRfc3389Cng) {
assert(output->Channels() == 1); // Not adapted for multi-channel yet.
static const int kCngLength = 32;
int16_t cng_output[kCngLength];
// Reset mute factor and start up fresh.
external_mute_factor_array[0] = 16384;
AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder) {
CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
// Generate long enough for 32kHz.
if (WebRtcCng_Generate(cng_inst, cng_output, kCngLength, 0) < 0) {
// Error returned; set return vector to all zeros.
memset(cng_output, 0, sizeof(cng_output));
}
} else {
// If no CNG instance is defined, just copy from the decoded data.
// (This will result in interpolating the decoded with itself.)
memcpy(cng_output, signal, fs_mult * 8 * sizeof(int16_t));
}
// Interpolate the CNG into the new vector.
// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
assert(fs_shift < 3); // Will always be 0, 1, or, 2.
int16_t increment = 4 >> fs_shift;
int16_t fraction = increment;
for (size_t i = 0; i < 8 * fs_mult; i++) {
// TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
// for legacy bit-exactness.
signal[i] =
(fraction * signal[i] + (32 - fraction) * cng_output[i] + 8) >> 5;
fraction += increment;
}
} else if (external_mute_factor_array[0] < 16384) {
// Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
// still ramping up from previous muting.
// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
int16_t increment = 64 / fs_mult;
size_t length_per_channel = length / output->Channels();
for (size_t i = 0; i < length_per_channel; i++) {
for (size_t channel_ix = 0; channel_ix < output->Channels();
++channel_ix) {
// Scale with mute factor.
assert(channel_ix < output->Channels());
assert(i < output->Size());
int32_t scaled_signal = (*output)[channel_ix][i] *
external_mute_factor_array[channel_ix];
// Shift 14 with proper rounding.
(*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
// Increase mute_factor towards 16384.
external_mute_factor_array[channel_ix] =
std::min(16384, external_mute_factor_array[channel_ix] + increment);
}
}
}
return static_cast<int>(length);
}
} // namespace webrtc