avcodec/g728_template: do_hybrid_window() template

intended for use by RealAudio 2.0 (28.8k) and G.728 decoders.

Peter Ross committed 1 year ago

Commit 93368021 parent b1172b82 files · +68 −47Browse files

Unified · Split

2 files

libavcodec/g728_template.c +65 −0
libavcodec/ra288.c +3 −47

@@ -0,0 +1,65 @@
	1	+/*
	2	+ * G.728 / RealAudio 2.0 (28.8K) decoder
	3	+ *
	4	+ * This file is part of FFmpeg.
	5	+ *
	6	+ * FFmpeg is free software; you can redistribute it and/or
	7	+ * modify it under the terms of the GNU Lesser General Public
	8	+ * License as published by the Free Software Foundation; either
	9	+ * version 2.1 of the License, or (at your option) any later version.
	10	+ *
	11	+ * FFmpeg is distributed in the hope that it will be useful,
	12	+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
	13	+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	14	+ * Lesser General Public License for more details.
	15	+ *
	16	+ * You should have received a copy of the GNU Lesser General Public
	17	+ * License along with FFmpeg; if not, write to the Free Software
	18	+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	19	+ */
	20	+
	21	+static void convolve(float tgt, const float src, int len, int n)
	22	+{
	23	+ for (; n >= 0; n--)
	24	+ tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
	25	+
	26	+}
	27	+
	28	+/**
	29	+ * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
	30	+ *
	31	+ * @param order filter order
	32	+ * @param n input length
	33	+ * @param non_rec number of non-recursive samples
	34	+ * @param out filter output
	35	+ * @param hist pointer to the input history of the filter
	36	+ * @param out pointer to the non-recursive part of the output
	37	+ * @param out2 pointer to the recursive part of the output
	38	+ * @param window pointer to the windowing function table
	39	+ */
	40	+static void do_hybrid_window(void (vector_fmul)(float dst, const float src0, const float src1, int len),
	41	+ int order, int n, int non_rec, float *out,
	42	+ float hist, float out2, const float *window)
	43	+{
	44	+ int i;
	45	+ float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
	46	+ float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
	47	+ LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
	48	+ MAX_BACKWARD_FILTER_LEN +
	49	+ MAX_BACKWARD_FILTER_NONREC, 16)]);
	50	+
	51	+ av_assert2(order>=0);
	52	+
	53	+ vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
	54	+
	55	+ convolve(buffer1, work + order , n , order);
	56	+ convolve(buffer2, work + order + n, non_rec, order);
	57	+
	58	+ for (i=0; i <= order; i++) {
	59	+ out2[i] = out2[i] * ATTEN + buffer1[i];
	60	+ out [i] = out2[i] + buffer2[i];
	61	+ }
	62	+
	63	+ /* Multiply by the white noise correcting factor (WNCF). */
	64	+ out = 257.0 / 256.0;
	65	+}

libavcodec/ra288.c+3−47

@@ -37,6 +37,8 @@ · 36 unmodified lines
37	37	#define MAX_BACKWARD_FILTER_ORDER 36
38	38	#define MAX_BACKWARD_FILTER_LEN 40
39	39	#define MAX_BACKWARD_FILTER_NONREC 35
	40	+#define ATTEN 0.5625
	41	+#include "g728_template.c"
40	42
41	43	#define RA288_BLOCK_SIZE 5
42	44	#define RA288_BLOCKS_PER_FRAME 32
@@ -87,13 +89,6 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx) · 44 unmodified lines
87	89	return 0;
88	90	}
89	91
90		−static void convolve(float tgt, const float src, int len, int n)
91		−{
92		− for (; n >= 0; n--)
93		− tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
94		−
95		−}
96		−
97	92	static void decode(RA288Context *ractx, float gain, int cb_coef)
98	93	{
99	94	int i;
@@ -131,45 +126,6 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) · 31 unmodified lines
131	126	ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
132	127	}
133	128
134		−/**
135		− * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
136		− *
137		− * @param order filter order
138		− * @param n input length
139		− * @param non_rec number of non-recursive samples
140		− * @param out filter output
141		− * @param hist pointer to the input history of the filter
142		− * @param out pointer to the non-recursive part of the output
143		− * @param out2 pointer to the recursive part of the output
144		− * @param window pointer to the windowing function table
145		− */
146		−static void do_hybrid_window(RA288Context *ractx,
147		− int order, int n, int non_rec, float *out,
148		− float hist, float out2, const float *window)
149		−{
150		− int i;
151		− float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
152		− float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
153		− LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
154		− MAX_BACKWARD_FILTER_LEN +
155		− MAX_BACKWARD_FILTER_NONREC, 16)]);
156		−
157		− av_assert2(order>=0);
158		−
159		− ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
160		−
161		− convolve(buffer1, work + order , n , order);
162		− convolve(buffer2, work + order + n, non_rec, order);
163		−
164		− for (i=0; i <= order; i++) {
165		− out2[i] = out2[i] * 0.5625 + buffer1[i];
166		− out [i] = out2[i] + buffer2[i];
167		− }
168		−
169		− /* Multiply by the white noise correcting factor (WNCF). */
170		− out = 257.0 / 256.0;
171		−}
172		−
173	129	/**
174	130	* Backward synthesis filter, find the LPC coefficients from past speech data.
175	131	*/
@@ -180,7 +136,7 @@ static void backward_filter(RA288Context *ractx, · 4 unmodified lines
180	136	{
181	137	float temp[MAX_BACKWARD_FILTER_ORDER+1];
182	138
183		− do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
	139	+ do_hybrid_window(ractx->vector_fmul, order, n, non_rec, temp, hist, rec, window);
184	140
185	141	if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
186	142	ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));

avcodec/g728_template: do_hybrid_window() template

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