webrtc/modules/audio_coding/neteq/normal.cc - src - Git at Google

 /*
  *  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/audio_decoder.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"

 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.
   if (length % output->Channels() != 0) {
     // The length does not match the number of channels.
     output->Clear();
     return 0;
   }
   output->PushBackInterleaved(input, length);

   const int 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();

     size_t length_per_channel = length / output->Channels();
     std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
     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>(
           (external_mute_factor_array[channel_ix] *
           expand_->MuteFactor(channel_ix)) >> 14);

       (*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());

       // Find largest absolute value in new data.
       int16_t decoded_max =
           WebRtcSpl_MaxAbsValueW16(signal.get(), length_per_channel);
       // Adjust muting factor if needed (to BGN level).
       size_t energy_length =
           std::min(static_cast<size_t>(fs_mult * 64), 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.get(), signal.get(),
                                                      energy_length, scaling);
       int32_t scaled_energy_length =
           static_cast<int32_t>(energy_length >> scaling);
       if (scaled_energy_length > 0) {
         energy = energy / scaled_energy_length;
       } else {
         energy = 0;
       }

       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 =
             static_cast<int16_t>(WEBRTC_SPL_SHIFT_W32(energy, scaling));
         int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
         mute_factor = WebRtcSpl_SqrtFloor(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] =
             static_cast<int16_t>(std::min(mute_factor, 16384));
       }

       // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
       int 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] =
             static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] = static_cast<int16_t>(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 < static_cast<size_t>(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] =
             static_cast<int16_t>((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 size_t kCngLength = 48;
     RTC_DCHECK_LE(static_cast<size_t>(8 * fs_mult), kCngLength);
     int16_t cng_output[kCngLength];
     // Reset mute factor and start up fresh.
     external_mute_factor_array[0] = 16384;
     ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();

     if (cng_decoder) {
       // Generate long enough for 48kHz.
       if (!cng_decoder->Generate(cng_output, 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.)
       (*output)[0].CopyTo(fs_mult * 8, 0, cng_output);
     }
     // 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 < static_cast<size_t>(8 * fs_mult); i++) {
       // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
       // for legacy bit-exactness.
       (*output)[0][i] = (fraction * (*output)[0][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).
     int 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] =
             static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
             16384, external_mute_factor_array[channel_ix] + increment));
       }
     }
   }

   return static_cast<int>(length);
 }

 }  // namespace webrtc
	/*
	* 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/audio_decoder.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"

	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.
	if (length % output->Channels() != 0) {
	// The length does not match the number of channels.
	output->Clear();
	return 0;
	}
	output->PushBackInterleaved(input, length);

	const int 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();

	size_t length_per_channel = length / output->Channels();
	std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
	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>(
	(external_mute_factor_array[channel_ix] *
	expand_->MuteFactor(channel_ix)) >> 14);

	(*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());

	// Find largest absolute value in new data.
	int16_t decoded_max =
	WebRtcSpl_MaxAbsValueW16(signal.get(), length_per_channel);
	// Adjust muting factor if needed (to BGN level).
	size_t energy_length =
	std::min(static_cast<size_t>(fs_mult * 64), 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.get(), signal.get(),
	energy_length, scaling);
	int32_t scaled_energy_length =
	static_cast<int32_t>(energy_length >> scaling);
	if (scaled_energy_length > 0) {
	energy = energy / scaled_energy_length;
	} else {
	energy = 0;
	}

	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 =
	static_cast<int16_t>(WEBRTC_SPL_SHIFT_W32(energy, scaling));
	int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
	mute_factor = WebRtcSpl_SqrtFloor(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] =
	static_cast<int16_t>(std::min(mute_factor, 16384));
	}

	// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
	int 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] =
	static_cast<int16_t>((scaled_signal + 8192) >> 14);
	// Increase mute_factor towards 16384.
	external_mute_factor_array[channel_ix] = static_cast<int16_t>(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 < static_cast<size_t>(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] =
	static_cast<int16_t>((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 size_t kCngLength = 48;
	RTC_DCHECK_LE(static_cast<size_t>(8 * fs_mult), kCngLength);
	int16_t cng_output[kCngLength];
	// Reset mute factor and start up fresh.
	external_mute_factor_array[0] = 16384;
	ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();

	if (cng_decoder) {
	// Generate long enough for 48kHz.
	if (!cng_decoder->Generate(cng_output, 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.)
	(output)[0].CopyTo(fs_mult 8, 0, cng_output);
	}
	// 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 < static_cast<size_t>(8 * fs_mult); i++) {
	// TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
	// for legacy bit-exactness.
	(output)[0][i] = (fraction (*output)[0][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).
	int 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] =
	static_cast<int16_t>((scaled_signal + 8192) >> 14);
	// Increase mute_factor towards 16384.
	external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
	16384, external_mute_factor_array[channel_ix] + increment));
	}
	}
	}

	return static_cast<int>(length);
	}

	} // namespace webrtc