[libav.git] / libavcodec / resample.c

/*
 * Sample rate convertion for both audio and video
 * Copyright (c) 2000 Fabrice Bellard.
 *
 * This library 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 of the License, or (at your option) any later version.
 *
 * This library 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 this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

/**
 * @file resample.c
 * Sample rate convertion for both audio and video.
 */

#include "avcodec.h"

typedef struct {
    /* fractional resampling */
    uint32_t incr; /* fractional increment */
    uint32_t frac;
    int last_sample;
    /* integer down sample */
    int iratio;  /* integer divison ratio */
    int icount, isum;
    int inv;
} ReSampleChannelContext;

struct ReSampleContext {
    ReSampleChannelContext channel_ctx[2];
    float ratio;
    /* channel convert */
    int input_channels, output_channels, filter_channels;
};


#define FRAC_BITS 16
#define FRAC (1 << FRAC_BITS)

static void init_mono_resample(ReSampleChannelContext *s, float ratio)
{
    ratio = 1.0 / ratio;
    s->iratio = (int)floorf(ratio);
    if (s->iratio == 0)
        s->iratio = 1;
    s->incr = (int)((ratio / s->iratio) * FRAC);
    s->frac = FRAC;
    s->last_sample = 0;
    s->icount = s->iratio;
    s->isum = 0;
    s->inv = (FRAC / s->iratio);
}

/* fractional audio resampling */
static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    unsigned int frac, incr;
    int l0, l1;
    short *q, *p, *pend;

    l0 = s->last_sample;
    incr = s->incr;
    frac = s->frac;

    p = input;
    pend = input + nb_samples;
    q = output;

    l1 = *p++;
    for(;;) {
        /* interpolate */
        *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
        frac = frac + s->incr;
        while (frac >= FRAC) {
            frac -= FRAC;
            if (p >= pend)
                goto the_end;
            l0 = l1;
            l1 = *p++;
        }
    }
 the_end:
    s->last_sample = l1;
    s->frac = frac;
    return q - output;
}

static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    short *q, *p, *pend;
    int c, sum;

    p = input;
    pend = input + nb_samples;
    q = output;

    c = s->icount;
    sum = s->isum;

    for(;;) {
        sum += *p++;
        if (--c == 0) {
            *q++ = (sum * s->inv) >> FRAC_BITS;
            c = s->iratio;
            sum = 0;
        }
        if (p >= pend)
            break;
    }
    s->isum = sum;
    s->icount = c;
    return q - output;
}

/* n1: number of samples */
static void stereo_to_mono(short *output, short *input, int n1)
{
    short *p, *q;
    int n = n1;

    p = input;
    q = output;
    while (n >= 4) {
        q[0] = (p[0] + p[1]) >> 1;
        q[1] = (p[2] + p[3]) >> 1;
        q[2] = (p[4] + p[5]) >> 1;
        q[3] = (p[6] + p[7]) >> 1;
        q += 4;
        p += 8;
        n -= 4;
    }
    while (n > 0) {
        q[0] = (p[0] + p[1]) >> 1;
        q++;
        p += 2;
        n--;
    }
}

/* n1: number of samples */
static void mono_to_stereo(short *output, short *input, int n1)
{
    short *p, *q;
    int n = n1;
    int v;

    p = input;
    q = output;
    while (n >= 4) {
        v = p[0]; q[0] = v; q[1] = v;
        v = p[1]; q[2] = v; q[3] = v;
        v = p[2]; q[4] = v; q[5] = v;
        v = p[3]; q[6] = v; q[7] = v;
        q += 8;
        p += 4;
        n -= 4;
    }
    while (n > 0) {
        v = p[0]; q[0] = v; q[1] = v;
        q += 2;
        p += 1;
        n--;
    }
}

/* XXX: should use more abstract 'N' channels system */
static void stereo_split(short *output1, short *output2, short *input, int n)
{
    int i;

    for(i=0;i<n;i++) {
        *output1++ = *input++;
        *output2++ = *input++;
    }
}

static void stereo_mux(short *output, short *input1, short *input2, int n)
{
    int i;

    for(i=0;i<n;i++) {
        *output++ = *input1++;
        *output++ = *input2++;
    }
}

static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
{
    int i;
    short l,r;

    for(i=0;i<n;i++) {
      l=*input1++;
      r=*input2++;
      *output++ = l;           /* left */
      *output++ = (l/2)+(r/2); /* center */
      *output++ = r;           /* right */
      *output++ = 0;           /* left surround */
      *output++ = 0;           /* right surroud */
      *output++ = 0;           /* low freq */
    }
}

static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    short *buf1;
    short *buftmp;

    buf1= (short*)av_malloc( nb_samples * sizeof(short) );

    /* first downsample by an integer factor with averaging filter */
    if (s->iratio > 1) {
        buftmp = buf1;
        nb_samples = integer_downsample(s, buftmp, input, nb_samples);
    } else {
        buftmp = input;
    }

    /* then do a fractional resampling with linear interpolation */
    if (s->incr != FRAC) {
        nb_samples = fractional_resample(s, output, buftmp, nb_samples);
    } else {
        memcpy(output, buftmp, nb_samples * sizeof(short));
    }
    av_free(buf1);
    return nb_samples;
}

ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
                                      int output_rate, int input_rate)
{
    ReSampleContext *s;
    int i;
    
    if ( input_channels > 2)
      {
	av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
	return NULL;
      }

    s = av_mallocz(sizeof(ReSampleContext));
    if (!s)
      {
	av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
	return NULL;
      }

    s->ratio = (float)output_rate / (float)input_rate;
    
    s->input_channels = input_channels;
    s->output_channels = output_channels;
    
    s->filter_channels = s->input_channels;
    if (s->output_channels < s->filter_channels)
        s->filter_channels = s->output_channels;

/*
 * ac3 output is the only case where filter_channels could be greater than 2.
 * input channels can't be greater than 2, so resample the 2 channels and then
 * expand to 6 channels after the resampling.
 */
    if(s->filter_channels>2)
      s->filter_channels = 2;

    for(i=0;i<s->filter_channels;i++) {
        init_mono_resample(&s->channel_ctx[i], s->ratio);
    }
    return s;
}

/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
/* XXX: do it with polyphase filters, since the quality here is
   HORRIBLE. Return the number of samples available in output */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
{
    int i, nb_samples1;
    short *bufin[2];
    short *bufout[2];
    short *buftmp2[2], *buftmp3[2];
    int lenout;

    if (s->input_channels == s->output_channels && s->ratio == 1.0) {
        /* nothing to do */
        memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
        return nb_samples;
    }

    /* XXX: move those malloc to resample init code */
    bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
    bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
    
    /* make some zoom to avoid round pb */
    lenout= (int)(nb_samples * s->ratio) + 16;
    bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
    bufout[1]= (short*) av_malloc( lenout * sizeof(short) );

    if (s->input_channels == 2 &&
        s->output_channels == 1) {
        buftmp2[0] = bufin[0];
        buftmp3[0] = output;
        stereo_to_mono(buftmp2[0], input, nb_samples);
    } else if (s->output_channels >= 2 && s->input_channels == 1) {
        buftmp2[0] = input;
        buftmp3[0] = bufout[0];
    } else if (s->output_channels >= 2) {
        buftmp2[0] = bufin[0];
        buftmp2[1] = bufin[1];
        buftmp3[0] = bufout[0];
        buftmp3[1] = bufout[1];
        stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
    } else {
        buftmp2[0] = input;
        buftmp3[0] = output;
    }

    /* resample each channel */
    nb_samples1 = 0; /* avoid warning */
    for(i=0;i<s->filter_channels;i++) {
        nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
    }

    if (s->output_channels == 2 && s->input_channels == 1) {
        mono_to_stereo(output, buftmp3[0], nb_samples1);
    } else if (s->output_channels == 2) {
        stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
    } else if (s->output_channels == 6) {
        ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
    }

    av_free(bufin[0]);
    av_free(bufin[1]);

    av_free(bufout[0]);
    av_free(bufout[1]);
    return nb_samples1;
}

void audio_resample_close(ReSampleContext *s)
{
    av_free(s);
}
Commit	Line	Data
de6d9b64 FB	1	/*
de6d9b64 FB	2	* Sample rate convertion for both audio and video
ff4ec49e	3	* Copyright (c) 2000 Fabrice Bellard.
de6d9b64	4	*
ff4ec49e FB	5	* This library is free software; you can redistribute it and/or
	6	* modify it under the terms of the GNU Lesser General Public
	7	* License as published by the Free Software Foundation; either
	8	* version 2 of the License, or (at your option) any later version.
de6d9b64	9	*
ff4ec49e	10	* This library is distributed in the hope that it will be useful,
de6d9b64	11	* but WITHOUT ANY WARRANTY; without even the implied warranty of
ff4ec49e FB	12	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
ff4ec49e FB	13	* Lesser General Public License for more details.
de6d9b64	14	*
ff4ec49e FB	15	* You should have received a copy of the GNU Lesser General Public
	16	* License along with this library; if not, write to the Free Software
	17	* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
de6d9b64	18	*/
983e3246 MN	19
	20	/**
	21	* @file resample.c
	22	* Sample rate convertion for both audio and video.
	23	*/
	24
de6d9b64	25	#include "avcodec.h"
69db4e10	26
de6d9b64 FB	27	typedef struct {
de6d9b64 FB	28	/* fractional resampling */
0c1a9eda ZK	29	uint32_t incr; /* fractional increment */
0c1a9eda ZK	30	uint32_t frac;
de6d9b64 FB	31	int last_sample;
	32	/* integer down sample */
	33	int iratio; /* integer divison ratio */
	34	int icount, isum;
	35	int inv;
	36	} ReSampleChannelContext;
	37
	38	struct ReSampleContext {
	39	ReSampleChannelContext channel_ctx[2];
	40	float ratio;
	41	/* channel convert */
	42	int input_channels, output_channels, filter_channels;
	43	};
	44
	45
	46	#define FRAC_BITS 16
	47	#define FRAC (1 << FRAC_BITS)
	48
	49	static void init_mono_resample(ReSampleChannelContext *s, float ratio)
	50	{
	51	ratio = 1.0 / ratio;
5c91a675	52	s->iratio = (int)floorf(ratio);
de6d9b64 FB	53	if (s->iratio == 0)
	54	s->iratio = 1;
	55	s->incr = (int)((ratio / s->iratio) * FRAC);
8170f3dc	56	s->frac = FRAC;
de6d9b64 FB	57	s->last_sample = 0;
	58	s->icount = s->iratio;
	59	s->isum = 0;
	60	s->inv = (FRAC / s->iratio);
	61	}
	62
	63	/* fractional audio resampling */
	64	static int fractional_resample(ReSampleChannelContext s, short output, short *input, int nb_samples)
	65	{
	66	unsigned int frac, incr;
	67	int l0, l1;
	68	short q, p, *pend;
	69
	70	l0 = s->last_sample;
	71	incr = s->incr;
	72	frac = s->frac;
	73
	74	p = input;
	75	pend = input + nb_samples;
	76	q = output;
	77
	78	l1 = *p++;
	79	for(;;) {
	80	/* interpolate */
	81	q++ = (l0 (FRAC - frac) + l1 * frac) >> FRAC_BITS;
	82	frac = frac + s->incr;
	83	while (frac >= FRAC) {
9c89585a	84	frac -= FRAC;
de6d9b64 FB	85	if (p >= pend)
de6d9b64 FB	86	goto the_end;
de6d9b64 FB	87	l0 = l1;
	88	l1 = *p++;
	89	}
	90	}
	91	the_end:
	92	s->last_sample = l1;
	93	s->frac = frac;
	94	return q - output;
	95	}
	96
	97	static int integer_downsample(ReSampleChannelContext s, short output, short *input, int nb_samples)
	98	{
	99	short q, p, *pend;
	100	int c, sum;
	101
	102	p = input;
	103	pend = input + nb_samples;
	104	q = output;
	105
	106	c = s->icount;
	107	sum = s->isum;
	108
	109	for(;;) {
	110	sum += *p++;
	111	if (--c == 0) {
	112	q++ = (sum s->inv) >> FRAC_BITS;
	113	c = s->iratio;
	114	sum = 0;
	115	}
	116	if (p >= pend)
	117	break;
	118	}
	119	s->isum = sum;
	120	s->icount = c;
	121	return q - output;
	122	}
	123
	124	/* n1: number of samples */
	125	static void stereo_to_mono(short output, short input, int n1)
	126	{
	127	short p, q;
	128	int n = n1;
	129
	130	p = input;
	131	q = output;
	132	while (n >= 4) {
	133	q[0] = (p[0] + p[1]) >> 1;
	134	q[1] = (p[2] + p[3]) >> 1;
	135	q[2] = (p[4] + p[5]) >> 1;
	136	q[3] = (p[6] + p[7]) >> 1;
	137	q += 4;
	138	p += 8;
	139	n -= 4;
	140	}
	141	while (n > 0) {
	142	q[0] = (p[0] + p[1]) >> 1;
	143	q++;
	144	p += 2;
	145	n--;
	146	}
	147	}
	148
	149	/* n1: number of samples */
	150	static void mono_to_stereo(short output, short input, int n1)
151	{
152	short p, q;
153	int n = n1;
154	int v;
155
156	p = input;
157	q = output;
158	while (n >= 4) {
159	v = p[0]; q[0] = v; q[1] = v;
160	v = p[1]; q[2] = v; q[3] = v;
161	v = p[2]; q[4] = v; q[5] = v;
162	v = p[3]; q[6] = v; q[7] = v;
163	q += 8;
164	p += 4;
165	n -= 4;
166	}
167	while (n > 0) {
168	v = p[0]; q[0] = v; q[1] = v;
169	q += 2;
170	p += 1;
171	n--;
172	}
173	}
174
175	/* XXX: should use more abstract 'N' channels system */
176	static void stereo_split(short output1, short output2, short *input, int n)
177	{
178	int i;
179
180	for(i=0;i<n;i++) {
181	output1++ = input++;
182	output2++ = input++;
183	}
184	}
185
186	static void stereo_mux(short output, short input1, short *input2, int n)
187	{
188	int i;
189
190	for(i=0;i<n;i++) {
191	output++ = input1++;
192	output++ = input2++;
193	}
194	}
195
743739d2 MN	196	static void ac3_5p1_mux(short output, short input1, short *input2, int n)
	197	{
	198	int i;
	199	short l,r;
	200
	201	for(i=0;i<n;i++) {
	202	l=*input1++;
	203	r=*input2++;
	204	output++ = l; / left */
	205	output++ = (l/2)+(r/2); / center */
	206	output++ = r; / right */
	207	output++ = 0; / left surround */
	208	output++ = 0; / right surroud */
	209	output++ = 0; / low freq */
	210	}
	211	}
	212
de6d9b64 FB	213	static int mono_resample(ReSampleChannelContext s, short output, short *input, int nb_samples)
de6d9b64 FB	214	{
1a565432	215	short *buf1;
de6d9b64 FB	216	short *buftmp;
de6d9b64 FB	217
6000abfa	218	buf1= (short)av_malloc( nb_samples sizeof(short) );
1a565432	219
de6d9b64 FB	220	/* first downsample by an integer factor with averaging filter */
	221	if (s->iratio > 1) {
	222	buftmp = buf1;
	223	nb_samples = integer_downsample(s, buftmp, input, nb_samples);
	224	} else {
	225	buftmp = input;
	226	}
	227
	228	/* then do a fractional resampling with linear interpolation */
	229	if (s->incr != FRAC) {
	230	nb_samples = fractional_resample(s, output, buftmp, nb_samples);
	231	} else {
	232	memcpy(output, buftmp, nb_samples * sizeof(short));
	233	}
6000abfa	234	av_free(buf1);
de6d9b64 FB	235	return nb_samples;
	236	}
	237
	238	ReSampleContext *audio_resample_init(int output_channels, int input_channels,
	239	int output_rate, int input_rate)
	240	{
	241	ReSampleContext *s;
	242	int i;
	243
743739d2 MN	244	if ( input_channels > 2)
743739d2 MN	245	{
9b879566	246	av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
743739d2 MN	247	return NULL;
743739d2 MN	248	}
de6d9b64 FB	249
	250	s = av_mallocz(sizeof(ReSampleContext));
	251	if (!s)
743739d2	252	{
9b879566	253	av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
743739d2 MN	254	return NULL;
743739d2 MN	255	}
de6d9b64 FB	256
	257	s->ratio = (float)output_rate / (float)input_rate;
	258
	259	s->input_channels = input_channels;
	260	s->output_channels = output_channels;
	261
	262	s->filter_channels = s->input_channels;
	263	if (s->output_channels < s->filter_channels)
	264	s->filter_channels = s->output_channels;
	265
743739d2 MN	266	/*
	267	* ac3 output is the only case where filter_channels could be greater than 2.
	268	* input channels can't be greater than 2, so resample the 2 channels and then
	269	* expand to 6 channels after the resampling.
	270	*/
	271	if(s->filter_channels>2)
	272	s->filter_channels = 2;
	273
de6d9b64 FB	274	for(i=0;i<s->filter_channels;i++) {
	275	init_mono_resample(&s->channel_ctx[i], s->ratio);
	276	}
	277	return s;
	278	}
	279
	280	/* resample audio. 'nb_samples' is the number of input samples */
	281	/* XXX: optimize it ! */
	282	/* XXX: do it with polyphase filters, since the quality here is
	283	HORRIBLE. Return the number of samples available in output */
	284	int audio_resample(ReSampleContext s, short output, short *input, int nb_samples)
	285	{
	286	int i, nb_samples1;
1a565432 FB	287	short *bufin[2];
1a565432 FB	288	short *bufout[2];
de6d9b64	289	short buftmp2[2], buftmp3[2];
1a565432	290	int lenout;
de6d9b64 FB	291
	292	if (s->input_channels == s->output_channels && s->ratio == 1.0) {
	293	/* nothing to do */
	294	memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
	295	return nb_samples;
	296	}
	297
1a565432	298	/* XXX: move those malloc to resample init code */
6000abfa FB	299	bufin[0]= (short) av_malloc( nb_samples sizeof(short) );
6000abfa FB	300	bufin[1]= (short) av_malloc( nb_samples sizeof(short) );
1a565432 FB	301
	302	/* make some zoom to avoid round pb */
	303	lenout= (int)(nb_samples * s->ratio) + 16;
6000abfa FB	304	bufout[0]= (short) av_malloc( lenout sizeof(short) );
6000abfa FB	305	bufout[1]= (short) av_malloc( lenout sizeof(short) );
1a565432	306
de6d9b64 FB	307	if (s->input_channels == 2 &&
	308	s->output_channels == 1) {
	309	buftmp2[0] = bufin[0];
	310	buftmp3[0] = output;
	311	stereo_to_mono(buftmp2[0], input, nb_samples);
743739d2	312	} else if (s->output_channels >= 2 && s->input_channels == 1) {
de6d9b64 FB	313	buftmp2[0] = input;
de6d9b64 FB	314	buftmp3[0] = bufout[0];
743739d2	315	} else if (s->output_channels >= 2) {
de6d9b64 FB	316	buftmp2[0] = bufin[0];
	317	buftmp2[1] = bufin[1];
	318	buftmp3[0] = bufout[0];
	319	buftmp3[1] = bufout[1];
	320	stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
	321	} else {
	322	buftmp2[0] = input;
	323	buftmp3[0] = output;
	324	}
	325
	326	/* resample each channel */
	327	nb_samples1 = 0; /* avoid warning */
	328	for(i=0;i<s->filter_channels;i++) {
	329	nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
	330	}
	331
	332	if (s->output_channels == 2 && s->input_channels == 1) {
	333	mono_to_stereo(output, buftmp3[0], nb_samples1);
	334	} else if (s->output_channels == 2) {
	335	stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
743739d2 MN	336	} else if (s->output_channels == 6) {
743739d2 MN	337	ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
de6d9b64 FB	338	}
de6d9b64 FB	339
6000abfa FB	340	av_free(bufin[0]);
6000abfa FB	341	av_free(bufin[1]);
1a565432	342
6000abfa FB	343	av_free(bufout[0]);
6000abfa FB	344	av_free(bufout[1]);
de6d9b64 FB	345	return nb_samples1;
	346	}
	347
	348	void audio_resample_close(ReSampleContext *s)
	349	{
6000abfa	350	av_free(s);
de6d9b64	351	}