1 /*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 /*
12 * This file contains the splitting filter functions.
13 *
14 */
15
16 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
17
18 #include <assert.h>
19
20 // Maximum number of samples in a low/high-band frame.
21 enum
22 {
23 kMaxBandFrameLength = 320 // 10 ms at 64 kHz.
24 };
25
26 // QMF filter coefficients in Q16.
27 static const uint16_t WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
28 static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
29
30 ///////////////////////////////////////////////////////////////////////////////////////////////
31 // WebRtcSpl_AllPassQMF(...)
32 //
33 // Allpass filter used by the analysis and synthesis parts of the QMF filter.
34 //
35 // Input:
36 // - in_data : Input data sequence (Q10)
37 // - data_length : Length of data sequence (>2)
38 // - filter_coefficients : Filter coefficients (length 3, Q16)
39 //
40 // Input & Output:
41 // - filter_state : Filter state (length 6, Q10).
42 //
43 // Output:
44 // - out_data : Output data sequence (Q10), length equal to
45 // |data_length|
46 //
47
WebRtcSpl_AllPassQMF(int32_t * in_data,size_t data_length,int32_t * out_data,const uint16_t * filter_coefficients,int32_t * filter_state)48 void WebRtcSpl_AllPassQMF(int32_t* in_data, size_t data_length,
49 int32_t* out_data, const uint16_t* filter_coefficients,
50 int32_t* filter_state)
51 {
52 // The procedure is to filter the input with three first order all pass filters
53 // (cascade operations).
54 //
55 // a_3 + q^-1 a_2 + q^-1 a_1 + q^-1
56 // y[n] = ----------- ----------- ----------- x[n]
57 // 1 + a_3q^-1 1 + a_2q^-1 1 + a_1q^-1
58 //
59 // The input vector |filter_coefficients| includes these three filter coefficients.
60 // The filter state contains the in_data state, in_data[-1], followed by
61 // the out_data state, out_data[-1]. This is repeated for each cascade.
62 // The first cascade filter will filter the |in_data| and store the output in
63 // |out_data|. The second will the take the |out_data| as input and make an
64 // intermediate storage in |in_data|, to save memory. The third, and final, cascade
65 // filter operation takes the |in_data| (which is the output from the previous cascade
66 // filter) and store the output in |out_data|.
67 // Note that the input vector values are changed during the process.
68 size_t k;
69 int32_t diff;
70 // First all-pass cascade; filter from in_data to out_data.
71
72 // Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
73 // vector position n. Then the final output will be y[n] = y_3[n]
74
75 // First loop, use the states stored in memory.
76 // "diff" should be safe from wrap around since max values are 2^25
77 // diff = (x[0] - y_1[-1])
78 diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[1]);
79 // y_1[0] = x[-1] + a_1 * (x[0] - y_1[-1])
80 out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);
81
82 // For the remaining loops, use previous values.
83 for (k = 1; k < data_length; k++)
84 {
85 // diff = (x[n] - y_1[n-1])
86 diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
87 // y_1[n] = x[n-1] + a_1 * (x[n] - y_1[n-1])
88 out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
89 }
90
91 // Update states.
92 filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
93 filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
94
95 // Second all-pass cascade; filter from out_data to in_data.
96 // diff = (y_1[0] - y_2[-1])
97 diff = WebRtcSpl_SubSatW32(out_data[0], filter_state[3]);
98 // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
99 in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
100 for (k = 1; k < data_length; k++)
101 {
102 // diff = (y_1[n] - y_2[n-1])
103 diff = WebRtcSpl_SubSatW32(out_data[k], in_data[k - 1]);
104 // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
105 in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
106 }
107
108 filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
109 filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
110
111 // Third all-pass cascade; filter from in_data to out_data.
112 // diff = (y_2[0] - y[-1])
113 diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[5]);
114 // y[0] = y_2[-1] + a_3 * (y_2[0] - y[-1])
115 out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
116 for (k = 1; k < data_length; k++)
117 {
118 // diff = (y_2[n] - y[n-1])
119 diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
120 // y[n] = y_2[n-1] + a_3 * (y_2[n] - y[n-1])
121 out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
122 }
123 filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
124 filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
125 }
126
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)127 void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
128 int16_t* low_band, int16_t* high_band,
129 int32_t* filter_state1, int32_t* filter_state2)
130 {
131 size_t i;
132 int16_t k;
133 int32_t tmp;
134 int32_t half_in1[kMaxBandFrameLength];
135 int32_t half_in2[kMaxBandFrameLength];
136 int32_t filter1[kMaxBandFrameLength];
137 int32_t filter2[kMaxBandFrameLength];
138 const size_t band_length = in_data_length / 2;
139 assert(in_data_length % 2 == 0);
140 assert(band_length <= kMaxBandFrameLength);
141
142 // Split even and odd samples. Also shift them to Q10.
143 for (i = 0, k = 0; i < band_length; i++, k += 2)
144 {
145 half_in2[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)in_data[k], 10);
146 half_in1[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)in_data[k + 1], 10);
147 }
148
149 // All pass filter even and odd samples, independently.
150 WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
151 WebRtcSpl_kAllPassFilter1, filter_state1);
152 WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
153 WebRtcSpl_kAllPassFilter2, filter_state2);
154
155 // Take the sum and difference of filtered version of odd and even
156 // branches to get upper & lower band.
157 for (i = 0; i < band_length; i++)
158 {
159 tmp = (filter1[i] + filter2[i] + 1024) >> 11;
160 low_band[i] = WebRtcSpl_SatW32ToW16(tmp);
161
162 tmp = (filter1[i] - filter2[i] + 1024) >> 11;
163 high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
164 }
165 }
166
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)167 void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
168 size_t band_length, int16_t* out_data,
169 int32_t* filter_state1, int32_t* filter_state2)
170 {
171 int32_t tmp;
172 int32_t half_in1[kMaxBandFrameLength];
173 int32_t half_in2[kMaxBandFrameLength];
174 int32_t filter1[kMaxBandFrameLength];
175 int32_t filter2[kMaxBandFrameLength];
176 size_t i;
177 int16_t k;
178 assert(band_length <= kMaxBandFrameLength);
179
180 // Obtain the sum and difference channels out of upper and lower-band channels.
181 // Also shift to Q10 domain.
182 for (i = 0; i < band_length; i++)
183 {
184 tmp = (int32_t)low_band[i] + (int32_t)high_band[i];
185 half_in1[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
186 tmp = (int32_t)low_band[i] - (int32_t)high_band[i];
187 half_in2[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
188 }
189
190 // all-pass filter the sum and difference channels
191 WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
192 WebRtcSpl_kAllPassFilter2, filter_state1);
193 WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
194 WebRtcSpl_kAllPassFilter1, filter_state2);
195
196 // The filtered signals are even and odd samples of the output. Combine
197 // them. The signals are Q10 should shift them back to Q0 and take care of
198 // saturation.
199 for (i = 0, k = 0; i < band_length; i++)
200 {
201 tmp = (filter2[i] + 512) >> 10;
202 out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
203
204 tmp = (filter1[i] + 512) >> 10;
205 out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
206 }
207
208 }
209