FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavcodec/ra288.c	Lines:	72	79	91.1 %
Date:	2020-10-23 17:01:47	Branches:	21	26	80.8 %


/*
 * RealAudio 2.0 (28.8K)
 * Copyright (c) 2003 The FFmpeg project
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/internal.h"

#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "celp_filters.h"
#include "get_bits.h"
#include "internal.h"
#include "lpc.h"
#include "ra288.h"

#define MAX_BACKWARD_FILTER_ORDER  36
#define MAX_BACKWARD_FILTER_LEN    40
#define MAX_BACKWARD_FILTER_NONREC 35

#define RA288_BLOCK_SIZE        5
#define RA288_BLOCKS_PER_FRAME 32

typedef struct RA288Context {
    void (*vector_fmul)(float *dst, const float *src0, const float *src1,
                        int len);
    DECLARE_ALIGNED(32, float,   sp_lpc)[FFALIGN(36, 16)];   ///< LPC coefficients for speech data (spec: A)
    DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)];   ///< LPC coefficients for gain        (spec: GB)

    /** speech data history                                      (spec: SB).
     *  Its first 70 coefficients are updated only at backward filtering.
     */
    float sp_hist[111];

    /// speech part of the gain autocorrelation                  (spec: REXP)
    float sp_rec[37];

    /** log-gain history                                         (spec: SBLG).
     *  Its first 28 coefficients are updated only at backward filtering.
     */
    float gain_hist[38];

    /// recursive part of the gain autocorrelation               (spec: REXPLG)
    float gain_rec[11];
} RA288Context;

static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
    RA288Context *ractx = avctx->priv_data;
    AVFloatDSPContext *fdsp;

    avctx->channels       = 1;
    avctx->channel_layout = AV_CH_LAYOUT_MONO;
    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

    if (avctx->block_align != 38) {
        av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
        return AVERROR_PATCHWELCOME;
    }

    fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
    if (!fdsp)
        return AVERROR(ENOMEM);
    ractx->vector_fmul = fdsp->vector_fmul;
    av_free(fdsp);

    return 0;
}

static void convolve(float *tgt, const float *src, int len, int n)
{
    for (; n >= 0; n--)
        tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);

}

static void decode(RA288Context *ractx, float gain, int cb_coef)
{
    int i;
    double sumsum;
    float sum, buffer[5];
    float *block = ractx->sp_hist + 70 + 36; // current block
    float *gain_block = ractx->gain_hist + 28;

    memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));

    /* block 46 of G.728 spec */
    sum = 32.0;
    for (i=0; i < 10; i++)
        sum -= gain_block[9-i] * ractx->gain_lpc[i];

    /* block 47 of G.728 spec */
    sum = av_clipf(sum, 0, 60);

    /* block 48 of G.728 spec */
    /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
    sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));

    for (i=0; i < 5; i++)
        buffer[i] = codetable[cb_coef][i] * sumsum;

    sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);

    sum = FFMAX(sum, 5.0 / (1<<24));

    /* shift and store */
    memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));

    gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);

    ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
}

/**
 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
 *
 * @param order   filter order
 * @param n       input length
 * @param non_rec number of non-recursive samples
 * @param out     filter output
 * @param hist    pointer to the input history of the filter
 * @param out     pointer to the non-recursive part of the output
 * @param out2    pointer to the recursive part of the output
 * @param window  pointer to the windowing function table
 */
static void do_hybrid_window(RA288Context *ractx,
                             int order, int n, int non_rec, float *out,
                             float *hist, float *out2, const float *window)
{
    int i;
    float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
    float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
    LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
                                            MAX_BACKWARD_FILTER_LEN   +
                                            MAX_BACKWARD_FILTER_NONREC, 16)]);

    av_assert2(order>=0);

    ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));

    convolve(buffer1, work + order    , n      , order);
    convolve(buffer2, work + order + n, non_rec, order);

    for (i=0; i <= order; i++) {
        out2[i] = out2[i] * 0.5625 + buffer1[i];
        out [i] = out2[i]          + buffer2[i];
    }

    /* Multiply by the white noise correcting factor (WNCF). */
    *out *= 257.0 / 256.0;
}

/**
 * Backward synthesis filter, find the LPC coefficients from past speech data.
 */
static void backward_filter(RA288Context *ractx,
                            float *hist, float *rec, const float *window,
                            float *lpc, const float *tab,
                            int order, int n, int non_rec, int move_size)
{
    float temp[MAX_BACKWARD_FILTER_ORDER+1];

    do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);

    if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
        ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));

    memmove(hist, hist + n, move_size*sizeof(*hist));
}

static int ra288_decode_frame(AVCodecContext * avctx, void *data,
                              int *got_frame_ptr, AVPacket *avpkt)
{
    AVFrame *frame     = data;
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    float *out;
    int i, ret;
    RA288Context *ractx = avctx->priv_data;
    GetBitContext gb;

    if (buf_size < avctx->block_align) {
        av_log(avctx, AV_LOG_ERROR,
               "Error! Input buffer is too small [%d<%d]\n",
               buf_size, avctx->block_align);
        return AVERROR_INVALIDDATA;
    }

    ret = init_get_bits8(&gb, buf, avctx->block_align);
    if (ret < 0)
        return ret;

    /* get output buffer */
    frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;
    out = (float *)frame->data[0];

    for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
        float gain = amptable[get_bits(&gb, 3)];
        int cb_coef = get_bits(&gb, 6 + (i&1));

        decode(ractx, gain, cb_coef);

        memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
        out += RA288_BLOCK_SIZE;

        if ((i & 7) == 3) {
            backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
                            ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);

            backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
                            ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
        }
    }

    *got_frame_ptr = 1;

    return avctx->block_align;
}

AVCodec ff_ra_288_decoder = {
    .name           = "real_288",
    .long_name      = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_RA_288,
    .priv_data_size = sizeof(RA288Context),
    .init           = ra288_decode_init,
    .decode         = ra288_decode_frame,
    .capabilities   = AV_CODEC_CAP_DR1,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* RealAudio 2.0 (28.8K)
3			* Copyright (c) 2003 The FFmpeg project
4			*
5			* This file is part of FFmpeg.
6			*
7			* FFmpeg is free software; you can redistribute it and/or
8			* modify it under the terms of the GNU Lesser General Public
9			* License as published by the Free Software Foundation; either
10			* version 2.1 of the License, or (at your option) any later version.
11			*
12			* FFmpeg is distributed in the hope that it will be useful,
13			* but WITHOUT ANY WARRANTY; without even the implied warranty of
14			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15			* Lesser General Public License for more details.
16			*
17			* You should have received a copy of the GNU Lesser General Public
18			* License along with FFmpeg; if not, write to the Free Software
19			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20			*/
21
22			#include "libavutil/channel_layout.h"
23			#include "libavutil/float_dsp.h"
24			#include "libavutil/internal.h"
25
26			#define BITSTREAM_READER_LE
27			#include "avcodec.h"
28			#include "celp_filters.h"
29			#include "get_bits.h"
30			#include "internal.h"
31			#include "lpc.h"
32			#include "ra288.h"
33
34			#define MAX_BACKWARD_FILTER_ORDER 36
35			#define MAX_BACKWARD_FILTER_LEN 40
36			#define MAX_BACKWARD_FILTER_NONREC 35
37
38			#define RA288_BLOCK_SIZE 5
39			#define RA288_BLOCKS_PER_FRAME 32
40
41			typedef struct RA288Context {
42			void (vector_fmul)(float dst, const float src0, const float src1,
43			int len);
44			DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
45			DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
46
47			/** speech data history (spec: SB).
48			* Its first 70 coefficients are updated only at backward filtering.
49			*/
50			float sp_hist[111];
51
52			/// speech part of the gain autocorrelation (spec: REXP)
53			float sp_rec[37];
54
55			/** log-gain history (spec: SBLG).
56			* Its first 28 coefficients are updated only at backward filtering.
57			*/
58			float gain_hist[38];
59
60			/// recursive part of the gain autocorrelation (spec: REXPLG)
61			float gain_rec[11];
62			} RA288Context;
63
64		2	static av_cold int ra288_decode_init(AVCodecContext *avctx)
65			{
66		2	RA288Context *ractx = avctx->priv_data;
67			AVFloatDSPContext *fdsp;
68
69		2	avctx->channels = 1;
70		2	avctx->channel_layout = AV_CH_LAYOUT_MONO;
71		2	avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
72
73	✗✓	2	if (avctx->block_align != 38) {
74			av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
75			return AVERROR_PATCHWELCOME;
76			}
77
78		2	fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
79	✗✓	2	if (!fdsp)
80			return AVERROR(ENOMEM);
81		2	ractx->vector_fmul = fdsp->vector_fmul;
82		2	av_free(fdsp);
83
84		2	return 0;
85			}
86
87		39168	static void convolve(float tgt, const float src, int len, int n)
88			{
89	✓✓	979200	for (; n >= 0; n--)
90		940032	tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
91
92		39168	}
93
94		78336	static void decode(RA288Context *ractx, float gain, int cb_coef)
95			{
96			int i;
97			double sumsum;
98			float sum, buffer[5];
99		78336	float *block = ractx->sp_hist + 70 + 36; // current block
100		78336	float *gain_block = ractx->gain_hist + 28;
101
102		78336	memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36sizeof(block));
103
104			/* block 46 of G.728 spec */
105		78336	sum = 32.0;
106	✓✓	861696	for (i=0; i < 10; i++)
107		783360	sum -= gain_block[9-i] * ractx->gain_lpc[i];
108
109			/* block 47 of G.728 spec */
110		78336	sum = av_clipf(sum, 0, 60);
111
112			/* block 48 of G.728 spec */
113			/* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
114		78336	sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
115
116	✓✓	470016	for (i=0; i < 5; i++)
117		391680	buffer[i] = codetable[cb_coef][i] * sumsum;
118
119		78336	sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
120
121	✓✓	78336	sum = FFMAX(sum, 5.0 / (1<<24));
122
123			/* shift and store */
124		78336	memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
125
126		78336	gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
127
128		78336	ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
129		78336	}
130
131			/**
132			* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
133			*
134			* @param order filter order
135			* @param n input length
136			* @param non_rec number of non-recursive samples
137			* @param out filter output
138			* @param hist pointer to the input history of the filter
139			* @param out pointer to the non-recursive part of the output
140			* @param out2 pointer to the recursive part of the output
141			* @param window pointer to the windowing function table
142			*/
143		19584	static void do_hybrid_window(RA288Context *ractx,
144			int order, int n, int non_rec, float *out,
145			float hist, float out2, const float *window)
146			{
147			int i;
148			float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
149			float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
150		19584	LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
151			MAX_BACKWARD_FILTER_LEN +
152			MAX_BACKWARD_FILTER_NONREC, 16)]);
153
154			av_assert2(order>=0);
155
156		19584	ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
157
158		19584	convolve(buffer1, work + order , n , order);
159		19584	convolve(buffer2, work + order + n, non_rec, order);
160
161	✓✓	489600	for (i=0; i <= order; i++) {
162		470016	out2[i] = out2[i] * 0.5625 + buffer1[i];
163		470016	out [i] = out2[i] + buffer2[i];
164			}
165
166			/* Multiply by the white noise correcting factor (WNCF). */
167		19584	out = 257.0 / 256.0;
168		19584	}
169
170			/**
171			* Backward synthesis filter, find the LPC coefficients from past speech data.
172			*/
173		19584	static void backward_filter(RA288Context *ractx,
174			float hist, float rec, const float *window,
175			float lpc, const float tab,
176			int order, int n, int non_rec, int move_size)
177			{
178			float temp[MAX_BACKWARD_FILTER_ORDER+1];
179
180		19584	do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
181
182	✓✓	19584	if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
183		19582	ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
184
185		19584	memmove(hist, hist + n, move_sizesizeof(hist));
186		19584	}
187
188		2448	static int ra288_decode_frame(AVCodecContext * avctx, void *data,
189			int got_frame_ptr, AVPacket avpkt)
190			{
191		2448	AVFrame *frame = data;
192		2448	const uint8_t *buf = avpkt->data;
193		2448	int buf_size = avpkt->size;
194			float *out;
195			int i, ret;
196		2448	RA288Context *ractx = avctx->priv_data;
197			GetBitContext gb;
198
199	✗✓	2448	if (buf_size < avctx->block_align) {
200			av_log(avctx, AV_LOG_ERROR,
201			"Error! Input buffer is too small [%d<%d]\n",
202			buf_size, avctx->block_align);
203			return AVERROR_INVALIDDATA;
204			}
205
206		2448	ret = init_get_bits8(&gb, buf, avctx->block_align);
207	✗✓	2448	if (ret < 0)
208			return ret;
209
210			/* get output buffer */
211		2448	frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
212	✗✓	2448	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
213			return ret;
214		2448	out = (float *)frame->data[0];
215
216	✓✓	80784	for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
217		78336	float gain = amptable[get_bits(&gb, 3)];
218		78336	int cb_coef = get_bits(&gb, 6 + (i&1));
219
220		78336	decode(ractx, gain, cb_coef);
221
222		78336	memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
223		78336	out += RA288_BLOCK_SIZE;
224
225	✓✓	78336	if ((i & 7) == 3) {
226		9792	backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
227		9792	ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
228
229		9792	backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
230		9792	ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
231			}
232			}
233
234		2448	*got_frame_ptr = 1;
235
236		2448	return avctx->block_align;
237			}
238
239			AVCodec ff_ra_288_decoder = {
240			.name = "real_288",
241			.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
242			.type = AVMEDIA_TYPE_AUDIO,
243			.id = AV_CODEC_ID_RA_288,
244			.priv_data_size = sizeof(RA288Context),
245			.init = ra288_decode_init,
246			.decode = ra288_decode_frame,
247			.capabilities = AV_CODEC_CAP_DR1,
248			};