webrtc/common_audio/signal_processing/splitting_filter.c - src - Git at Google

 /*
  *  Copyright (c) 2011 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.
  */

 /*
  * This file contains the splitting filter functions.
  *
  */

 #include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // Maximum number of samples in a low/high-band frame.
 enum
 {
     kMaxBandFrameLength = 320  // 10 ms at 64 kHz.
 };

 // QMF filter coefficients in Q16.
 static const uint16_t WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
 static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};

 ///////////////////////////////////////////////////////////////////////////////////////////////
 // WebRtcSpl_AllPassQMF(...)
 //
 // Allpass filter used by the analysis and synthesis parts of the QMF filter.
 //
 // Input:
 //    - in_data             : Input data sequence (Q10)
 //    - data_length         : Length of data sequence (>2)
 //    - filter_coefficients : Filter coefficients (length 3, Q16)
 //
 // Input & Output:
 //    - filter_state        : Filter state (length 6, Q10).
 //
 // Output:
 //    - out_data            : Output data sequence (Q10), length equal to
 //                            |data_length|
 //

 void WebRtcSpl_AllPassQMF(int32_t* in_data, size_t data_length,
                           int32_t* out_data, const uint16_t* filter_coefficients,
                           int32_t* filter_state)
 {
     // The procedure is to filter the input with three first order all pass filters
     // (cascade operations).
     //
     //         a_3 + q^-1    a_2 + q^-1    a_1 + q^-1
     // y[n] =  -----------   -----------   -----------   x[n]
     //         1 + a_3q^-1   1 + a_2q^-1   1 + a_1q^-1
     //
     // The input vector |filter_coefficients| includes these three filter coefficients.
     // The filter state contains the in_data state, in_data[-1], followed by
     // the out_data state, out_data[-1]. This is repeated for each cascade.
     // The first cascade filter will filter the |in_data| and store the output in
     // |out_data|. The second will the take the |out_data| as input and make an
     // intermediate storage in |in_data|, to save memory. The third, and final, cascade
     // filter operation takes the |in_data| (which is the output from the previous cascade
     // filter) and store the output in |out_data|.
     // Note that the input vector values are changed during the process.
     size_t k;
     int32_t diff;
     // First all-pass cascade; filter from in_data to out_data.

     // Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
     // vector position n. Then the final output will be y[n] = y_3[n]

     // First loop, use the states stored in memory.
     // "diff" should be safe from wrap around since max values are 2^25
     // diff = (x[0] - y_1[-1])
     diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[1]);
     // y_1[0] =  x[-1] + a_1 * (x[0] - y_1[-1])
     out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);

     // For the remaining loops, use previous values.
     for (k = 1; k < data_length; k++)
     {
         // diff = (x[n] - y_1[n-1])
         diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
         // y_1[n] =  x[n-1] + a_1 * (x[n] - y_1[n-1])
         out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
     }

     // Update states.
     filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
     filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time

     // Second all-pass cascade; filter from out_data to in_data.
     // diff = (y_1[0] - y_2[-1])
     diff = WebRtcSpl_SubSatW32(out_data[0], filter_state[3]);
     // y_2[0] =  y_1[-1] + a_2 * (y_1[0] - y_2[-1])
     in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
     for (k = 1; k < data_length; k++)
     {
         // diff = (y_1[n] - y_2[n-1])
         diff = WebRtcSpl_SubSatW32(out_data[k], in_data[k - 1]);
         // y_2[0] =  y_1[-1] + a_2 * (y_1[0] - y_2[-1])
         in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
     }

     filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
     filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time

     // Third all-pass cascade; filter from in_data to out_data.
     // diff = (y_2[0] - y[-1])
     diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[5]);
     // y[0] =  y_2[-1] + a_3 * (y_2[0] - y[-1])
     out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
     for (k = 1; k < data_length; k++)
     {
         // diff = (y_2[n] - y[n-1])
         diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
         // y[n] =  y_2[n-1] + a_3 * (y_2[n] - y[n-1])
         out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
     }
     filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
     filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
 }

 void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
                            int16_t* low_band, int16_t* high_band,
                            int32_t* filter_state1, int32_t* filter_state2)
 {
     size_t i;
     int16_t k;
     int32_t tmp;
     int32_t half_in1[kMaxBandFrameLength];
     int32_t half_in2[kMaxBandFrameLength];
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
     const size_t band_length = in_data_length / 2;
     RTC_DCHECK_EQ(0, in_data_length % 2);
     RTC_DCHECK_LE(band_length, kMaxBandFrameLength);

     // Split even and odd samples. Also shift them to Q10.
     for (i = 0, k = 0; i < band_length; i++, k += 2)
     {
         half_in2[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)in_data[k], 10);
         half_in1[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)in_data[k + 1], 10);
     }

     // All pass filter even and odd samples, independently.
     WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
                          WebRtcSpl_kAllPassFilter1, filter_state1);
     WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
                          WebRtcSpl_kAllPassFilter2, filter_state2);

     // Take the sum and difference of filtered version of odd and even
     // branches to get upper & lower band.
     for (i = 0; i < band_length; i++)
     {
         tmp = (filter1[i] + filter2[i] + 1024) >> 11;
         low_band[i] = WebRtcSpl_SatW32ToW16(tmp);

         tmp = (filter1[i] - filter2[i] + 1024) >> 11;
         high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
     }
 }

 void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
                             size_t band_length, int16_t* out_data,
                             int32_t* filter_state1, int32_t* filter_state2)
 {
     int32_t tmp;
     int32_t half_in1[kMaxBandFrameLength];
     int32_t half_in2[kMaxBandFrameLength];
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
     size_t i;
     int16_t k;
     RTC_DCHECK_LE(band_length, kMaxBandFrameLength);

     // Obtain the sum and difference channels out of upper and lower-band channels.
     // Also shift to Q10 domain.
     for (i = 0; i < band_length; i++)
     {
         tmp = (int32_t)low_band[i] + (int32_t)high_band[i];
         half_in1[i] = tmp * (1 << 10);
         tmp = (int32_t)low_band[i] - (int32_t)high_band[i];
         half_in2[i] = tmp * (1 << 10);
     }

     // all-pass filter the sum and difference channels
     WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
                          WebRtcSpl_kAllPassFilter2, filter_state1);
     WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
                          WebRtcSpl_kAllPassFilter1, filter_state2);

     // The filtered signals are even and odd samples of the output. Combine
     // them. The signals are Q10 should shift them back to Q0 and take care of
     // saturation.
     for (i = 0, k = 0; i < band_length; i++)
     {
         tmp = (filter2[i] + 512) >> 10;
         out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);

         tmp = (filter1[i] + 512) >> 10;
         out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
     }

 }
	/*
	* Copyright (c) 2011 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.
	*/

	/*
	* This file contains the splitting filter functions.
	*
	*/

	#include "webrtc/base/checks.h"
	#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

	// Maximum number of samples in a low/high-band frame.
	enum
	{
	kMaxBandFrameLength = 320 // 10 ms at 64 kHz.
	};

	// QMF filter coefficients in Q16.
	static const uint16_t WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
	static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};

	///////////////////////////////////////////////////////////////////////////////////////////////
	// WebRtcSpl_AllPassQMF(...)
	//
	// Allpass filter used by the analysis and synthesis parts of the QMF filter.
	//
	// Input:
	// - in_data : Input data sequence (Q10)
	// - data_length : Length of data sequence (>2)
	// - filter_coefficients : Filter coefficients (length 3, Q16)
	//
	// Input & Output:
	// - filter_state : Filter state (length 6, Q10).
	//
	// Output:
	// - out_data : Output data sequence (Q10), length equal to
	// \|data_length\|
	//

	void WebRtcSpl_AllPassQMF(int32_t* in_data, size_t data_length,
	int32_t* out_data, const uint16_t* filter_coefficients,
	int32_t* filter_state)
	{
	// The procedure is to filter the input with three first order all pass filters
	// (cascade operations).
	//
	// a_3 + q^-1 a_2 + q^-1 a_1 + q^-1
	// y[n] = ----------- ----------- ----------- x[n]
	// 1 + a_3q^-1 1 + a_2q^-1 1 + a_1q^-1
	//
	// The input vector \|filter_coefficients\| includes these three filter coefficients.
	// The filter state contains the in_data state, in_data[-1], followed by
	// the out_data state, out_data[-1]. This is repeated for each cascade.
	// The first cascade filter will filter the \|in_data\| and store the output in
	// \|out_data\|. The second will the take the \|out_data\| as input and make an
	// intermediate storage in \|in_data\|, to save memory. The third, and final, cascade
	// filter operation takes the \|in_data\| (which is the output from the previous cascade
	// filter) and store the output in \|out_data\|.
	// Note that the input vector values are changed during the process.
	size_t k;
	int32_t diff;
	// First all-pass cascade; filter from in_data to out_data.

	// Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
	// vector position n. Then the final output will be y[n] = y_3[n]

	// First loop, use the states stored in memory.
	// "diff" should be safe from wrap around since max values are 2^25
	// diff = (x[0] - y_1[-1])
	diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[1]);
	// y_1[0] = x[-1] + a_1 * (x[0] - y_1[-1])
	out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);

	// For the remaining loops, use previous values.
	for (k = 1; k < data_length; k++)
	{
	// diff = (x[n] - y_1[n-1])
	diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
	// y_1[n] = x[n-1] + a_1 * (x[n] - y_1[n-1])
	out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
	}

	// Update states.
	filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
	filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time

	// Second all-pass cascade; filter from out_data to in_data.
	// diff = (y_1[0] - y_2[-1])
	diff = WebRtcSpl_SubSatW32(out_data[0], filter_state[3]);
	// y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
	in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
	for (k = 1; k < data_length; k++)
	{
	// diff = (y_1[n] - y_2[n-1])
	diff = WebRtcSpl_SubSatW32(out_data[k], in_data[k - 1]);
	// y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
	in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
	}

	filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
	filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time

	// Third all-pass cascade; filter from in_data to out_data.
	// diff = (y_2[0] - y[-1])
	diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[5]);
	// y[0] = y_2[-1] + a_3 * (y_2[0] - y[-1])
	out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
	for (k = 1; k < data_length; k++)
	{
	// diff = (y_2[n] - y[n-1])
	diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
	// y[n] = y_2[n-1] + a_3 * (y_2[n] - y[n-1])
	out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
	}
	filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
	filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
	}

	void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
	int16_t* low_band, int16_t* high_band,
	int32_t* filter_state1, int32_t* filter_state2)
	{
	size_t i;
	int16_t k;
	int32_t tmp;
	int32_t half_in1[kMaxBandFrameLength];
	int32_t half_in2[kMaxBandFrameLength];
	int32_t filter1[kMaxBandFrameLength];
	int32_t filter2[kMaxBandFrameLength];
	const size_t band_length = in_data_length / 2;
	RTC_DCHECK_EQ(0, in_data_length % 2);
	RTC_DCHECK_LE(band_length, kMaxBandFrameLength);

	// Split even and odd samples. Also shift them to Q10.
	for (i = 0, k = 0; i < band_length; i++, k += 2)
	{
	half_in2[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)in_data[k], 10);
	half_in1[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)in_data[k + 1], 10);
	}

	// All pass filter even and odd samples, independently.
	WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
	WebRtcSpl_kAllPassFilter1, filter_state1);
	WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
	WebRtcSpl_kAllPassFilter2, filter_state2);

	// Take the sum and difference of filtered version of odd and even
	// branches to get upper & lower band.
	for (i = 0; i < band_length; i++)
	{
	tmp = (filter1[i] + filter2[i] + 1024) >> 11;
	low_band[i] = WebRtcSpl_SatW32ToW16(tmp);

	tmp = (filter1[i] - filter2[i] + 1024) >> 11;
	high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
	}
	}

	void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
	size_t band_length, int16_t* out_data,
	int32_t* filter_state1, int32_t* filter_state2)
	{
	int32_t tmp;
	int32_t half_in1[kMaxBandFrameLength];
	int32_t half_in2[kMaxBandFrameLength];
	int32_t filter1[kMaxBandFrameLength];
	int32_t filter2[kMaxBandFrameLength];
	size_t i;
	int16_t k;
	RTC_DCHECK_LE(band_length, kMaxBandFrameLength);

	// Obtain the sum and difference channels out of upper and lower-band channels.
	// Also shift to Q10 domain.
	for (i = 0; i < band_length; i++)
	{
	tmp = (int32_t)low_band[i] + (int32_t)high_band[i];
	half_in1[i] = tmp * (1 << 10);
	tmp = (int32_t)low_band[i] - (int32_t)high_band[i];
	half_in2[i] = tmp * (1 << 10);
	}

	// all-pass filter the sum and difference channels
	WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
	WebRtcSpl_kAllPassFilter2, filter_state1);
	WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
	WebRtcSpl_kAllPassFilter1, filter_state2);

	// The filtered signals are even and odd samples of the output. Combine
	// them. The signals are Q10 should shift them back to Q0 and take care of
	// saturation.
	for (i = 0, k = 0; i < band_length; i++)
	{
	tmp = (filter2[i] + 512) >> 10;
	out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);

	tmp = (filter1[i] + 512) >> 10;
	out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
	}

	}