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 "modules/audio_coding/neteq/normal.h"

 #include <string.h>  // memset, memcpy

 #include <algorithm>  // min

 #include "api/audio_codecs/audio_decoder.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/background_noise.h"
 #include "modules/audio_coding/neteq/decoder_database.h"
 #include "modules/audio_coding/neteq/expand.h"
 #include "rtc_base/checks.h"

 namespace webrtc {

 int Normal::Process(const int16_t* input,
                     size_t length,
                     Modes last_mode,
                     AudioMultiVector* output) {
   if (length == 0) {
     // Nothing to process.
     output->Clear();
     return static_cast<int>(length);
   }

   RTC_DCHECK(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(rtc::ArrayView<const int16_t>(input, length));

   const int fs_mult = fs_hz_ / 8000;
   RTC_DCHECK_GT(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) {
       // Set muting factor to the same as expand muting factor.
       int16_t mute_factor = expand_->MuteFactor(channel_ix);

       (*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 local_mute_factor = 16384;  // 1.0 in Q14.
       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 = WEBRTC_SPL_SHIFT_W32(
             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);
         local_mute_factor =
             std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
       }
       mute_factor = std::max<int16_t>(mute_factor, local_mute_factor);
       RTC_DCHECK_LE(mute_factor, 16384);
       RTC_DCHECK_GE(mute_factor, 0);

       // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14),
       // 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) / length_per_channel);
       const int increment = std::max(64 / fs_mult, back_to_fullscale_inc);
       for (size_t i = 0; i < length_per_channel; i++) {
         // Scale with mute factor.
         RTC_DCHECK_LT(channel_ix, output->Channels());
         RTC_DCHECK_LT(i, output->Size());
         int32_t scaled_signal = (*output)[channel_ix][i] * mute_factor;
         // Shift 14 with proper rounding.
         (*output)[channel_ix][i] =
             static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
         mute_factor =
             static_cast<int16_t>(std::min(mute_factor + increment, 16384));
       }

       // Interpolate the expanded data into the new vector.
       // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
       size_t win_length = samples_per_ms_;
       int16_t win_slope_Q14 = default_win_slope_Q14_;
       RTC_DCHECK_LT(channel_ix, output->Channels());
       if (win_length > output->Size()) {
         win_length = output->Size();
         win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
       }
       int16_t win_up_Q14 = 0;
       for (size_t i = 0; i < win_length; i++) {
         win_up_Q14 += win_slope_Q14;
         (*output)[channel_ix][i] =
             (win_up_Q14 * (*output)[channel_ix][i] +
              ((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
             14;
       }
       RTC_DCHECK_GT(win_up_Q14,
                     (1 << 14) - 32);  // Worst case rouding is a length of 34
     }
   } else if (last_mode == kModeRfc3389Cng) {
     RTC_DCHECK_EQ(output->Channels(), 1);  // Not adapted for multi-channel yet.
     static const size_t kCngLength = 48;
     RTC_DCHECK_LE(8 * fs_mult, kCngLength);
     int16_t cng_output[kCngLength];
     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.)
     size_t win_length = samples_per_ms_;
     int16_t win_slope_Q14 = default_win_slope_Q14_;
     if (win_length > kCngLength) {
       win_length = kCngLength;
       win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
     }
     int16_t win_up_Q14 = 0;
     for (size_t i = 0; i < win_length; i++) {
       win_up_Q14 += win_slope_Q14;
       (*output)[0][i] =
           (win_up_Q14 * (*output)[0][i] +
            ((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
           14;
     }
     RTC_DCHECK_GT(win_up_Q14,
                   (1 << 14) - 32);  // Worst case rouding is a length of 34
   }

   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 "modules/audio_coding/neteq/normal.h"

	#include <string.h> // memset, memcpy

	#include <algorithm> // min

	#include "api/audio_codecs/audio_decoder.h"
	#include "common_audio/signal_processing/include/signal_processing_library.h"
	#include "modules/audio_coding/neteq/audio_multi_vector.h"
	#include "modules/audio_coding/neteq/background_noise.h"
	#include "modules/audio_coding/neteq/decoder_database.h"
	#include "modules/audio_coding/neteq/expand.h"
	#include "rtc_base/checks.h"

	namespace webrtc {

	int Normal::Process(const int16_t* input,
	size_t length,
	Modes last_mode,
	AudioMultiVector* output) {
	if (length == 0) {
	// Nothing to process.
	output->Clear();
	return static_cast<int>(length);
	}

	RTC_DCHECK(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(rtc::ArrayView<const int16_t>(input, length));

	const int fs_mult = fs_hz_ / 8000;
	RTC_DCHECK_GT(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) {
	// Set muting factor to the same as expand muting factor.
	int16_t mute_factor = expand_->MuteFactor(channel_ix);

	(*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 local_mute_factor = 16384; // 1.0 in Q14.
	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 = WEBRTC_SPL_SHIFT_W32(
	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);
	local_mute_factor =
	std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
	}
	mute_factor = std::max<int16_t>(mute_factor, local_mute_factor);
	RTC_DCHECK_LE(mute_factor, 16384);
	RTC_DCHECK_GE(mute_factor, 0);

	// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14),
	// 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) / length_per_channel);
	const int increment = std::max(64 / fs_mult, back_to_fullscale_inc);
	for (size_t i = 0; i < length_per_channel; i++) {
	// Scale with mute factor.
	RTC_DCHECK_LT(channel_ix, output->Channels());
	RTC_DCHECK_LT(i, output->Size());
	int32_t scaled_signal = (output)[channel_ix][i] mute_factor;
	// Shift 14 with proper rounding.
	(*output)[channel_ix][i] =
	static_cast<int16_t>((scaled_signal + 8192) >> 14);
	// Increase mute_factor towards 16384.
	mute_factor =
	static_cast<int16_t>(std::min(mute_factor + increment, 16384));
	}

	// Interpolate the expanded data into the new vector.
	// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
	size_t win_length = samples_per_ms_;
	int16_t win_slope_Q14 = default_win_slope_Q14_;
	RTC_DCHECK_LT(channel_ix, output->Channels());
	if (win_length > output->Size()) {
	win_length = output->Size();
	win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
	}
	int16_t win_up_Q14 = 0;
	for (size_t i = 0; i < win_length; i++) {
	win_up_Q14 += win_slope_Q14;
	(*output)[channel_ix][i] =
	(win_up_Q14 * (*output)[channel_ix][i] +
	((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
	14;
	}
	RTC_DCHECK_GT(win_up_Q14,
	(1 << 14) - 32); // Worst case rouding is a length of 34
	}
	} else if (last_mode == kModeRfc3389Cng) {
	RTC_DCHECK_EQ(output->Channels(), 1); // Not adapted for multi-channel yet.
	static const size_t kCngLength = 48;
	RTC_DCHECK_LE(8 * fs_mult, kCngLength);
	int16_t cng_output[kCngLength];
	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.)
	size_t win_length = samples_per_ms_;
	int16_t win_slope_Q14 = default_win_slope_Q14_;
	if (win_length > kCngLength) {
	win_length = kCngLength;
	win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
	}
	int16_t win_up_Q14 = 0;
	for (size_t i = 0; i < win_length; i++) {
	win_up_Q14 += win_slope_Q14;
	(*output)[0][i] =
	(win_up_Q14 * (*output)[0][i] +
	((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
	14;
	}
	RTC_DCHECK_GT(win_up_Q14,
	(1 << 14) - 32); // Worst case rouding is a length of 34
	}

	return static_cast<int>(length);
	}

	} // namespace webrtc