[libav.git] / libavcodec / resample.c

/*
 * Sample rate convertion for both audio and video
 * Copyright (c) 2000 Gerard Lantau.
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
 */
#include "avcodec.h"
#include <math.h>

typedef struct {
    /* fractional resampling */
    UINT32 incr; /* fractional increment */
    UINT32 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)floor(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) {
            if (p >= pend)
                goto the_end;
            frac -= FRAC;
            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 int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    short *buf1;
    short *buftmp;

    buf1= (short*) 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));
    }
    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 (output_channels > 2 || input_channels > 2)
        return NULL;

    s = av_mallocz(sizeof(ReSampleContext));
    if (!s)
        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;

    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*) malloc( nb_samples * sizeof(short) );
    bufin[1]= (short*) malloc( nb_samples * sizeof(short) );
    
    /* make some zoom to avoid round pb */
    lenout= (int)(nb_samples * s->ratio) + 16;
    bufout[0]= (short*) malloc( lenout * sizeof(short) );
    bufout[1]= (short*) 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);
    }

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

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

void audio_resample_close(ReSampleContext *s)
{
    free(s);
}
Commit	Line	Data
de6d9b64 FB	1	/*
	2	* Sample rate convertion for both audio and video
	3	* Copyright (c) 2000 Gerard Lantau.
	4	*
	5	* This program is free software; you can redistribute it and/or modify
	6	* it under the terms of the GNU General Public License as published by
	7	* the Free Software Foundation; either version 2 of the License, or
	8	* (at your option) any later version.
	9	*
	10	* This program is distributed in the hope that it will be useful,
	11	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	12	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	13	* GNU General Public License for more details.
	14	*
	15	* You should have received a copy of the GNU General Public License
	16	* along with this program; if not, write to the Free Software
	17	* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	18	*/
de6d9b64	19	#include "avcodec.h"
1a565432	20	#include <math.h>
de6d9b64 FB	21
	22	typedef struct {
	23	/* fractional resampling */
	24	UINT32 incr; /* fractional increment */
	25	UINT32 frac;
	26	int last_sample;
	27	/* integer down sample */
	28	int iratio; /* integer divison ratio */
	29	int icount, isum;
	30	int inv;
	31	} ReSampleChannelContext;
	32
	33	struct ReSampleContext {
	34	ReSampleChannelContext channel_ctx[2];
	35	float ratio;
	36	/* channel convert */
	37	int input_channels, output_channels, filter_channels;
	38	};
	39
	40
	41	#define FRAC_BITS 16
	42	#define FRAC (1 << FRAC_BITS)
	43
	44	static void init_mono_resample(ReSampleChannelContext *s, float ratio)
	45	{
	46	ratio = 1.0 / ratio;
	47	s->iratio = (int)floor(ratio);
	48	if (s->iratio == 0)
	49	s->iratio = 1;
	50	s->incr = (int)((ratio / s->iratio) * FRAC);
8170f3dc	51	s->frac = FRAC;
de6d9b64 FB	52	s->last_sample = 0;
	53	s->icount = s->iratio;
	54	s->isum = 0;
	55	s->inv = (FRAC / s->iratio);
	56	}
	57
	58	/* fractional audio resampling */
	59	static int fractional_resample(ReSampleChannelContext s, short output, short *input, int nb_samples)
	60	{
	61	unsigned int frac, incr;
	62	int l0, l1;
	63	short q, p, *pend;
	64
	65	l0 = s->last_sample;
	66	incr = s->incr;
	67	frac = s->frac;
	68
	69	p = input;
	70	pend = input + nb_samples;
	71	q = output;
	72
	73	l1 = *p++;
	74	for(;;) {
	75	/* interpolate */
	76	q++ = (l0 (FRAC - frac) + l1 * frac) >> FRAC_BITS;
	77	frac = frac + s->incr;
	78	while (frac >= FRAC) {
	79	if (p >= pend)
	80	goto the_end;
	81	frac -= FRAC;
	82	l0 = l1;
	83	l1 = *p++;
	84	}
	85	}
	86	the_end:
	87	s->last_sample = l1;
	88	s->frac = frac;
	89	return q - output;
	90	}
	91
	92	static int integer_downsample(ReSampleChannelContext s, short output, short *input, int nb_samples)
	93	{
	94	short q, p, *pend;
	95	int c, sum;
	96
	97	p = input;
	98	pend = input + nb_samples;
	99	q = output;
	100
	101	c = s->icount;
	102	sum = s->isum;
	103
	104	for(;;) {
	105	sum += *p++;
	106	if (--c == 0) {
	107	q++ = (sum s->inv) >> FRAC_BITS;
	108	c = s->iratio;
	109	sum = 0;
	110	}
	111	if (p >= pend)
	112	break;
	113	}
	114	s->isum = sum;
	115	s->icount = c;
116	return q - output;
117	}
118
119	/* n1: number of samples */
120	static void stereo_to_mono(short output, short input, int n1)
121	{
122	short p, q;
123	int n = n1;
124
125	p = input;
126	q = output;
127	while (n >= 4) {
128	q[0] = (p[0] + p[1]) >> 1;
129	q[1] = (p[2] + p[3]) >> 1;
130	q[2] = (p[4] + p[5]) >> 1;
131	q[3] = (p[6] + p[7]) >> 1;
132	q += 4;
133	p += 8;
134	n -= 4;
135	}
136	while (n > 0) {
137	q[0] = (p[0] + p[1]) >> 1;
138	q++;
139	p += 2;
140	n--;
141	}
142	}
143
144	/* n1: number of samples */
145	static void mono_to_stereo(short output, short input, int n1)
146	{
147	short p, q;
148	int n = n1;
149	int v;
150
151	p = input;
152	q = output;
153	while (n >= 4) {
154	v = p[0]; q[0] = v; q[1] = v;
155	v = p[1]; q[2] = v; q[3] = v;
156	v = p[2]; q[4] = v; q[5] = v;
157	v = p[3]; q[6] = v; q[7] = v;
158	q += 8;
159	p += 4;
160	n -= 4;
161	}
162	while (n > 0) {
163	v = p[0]; q[0] = v; q[1] = v;
164	q += 2;
165	p += 1;
166	n--;
167	}
168	}
169
170	/* XXX: should use more abstract 'N' channels system */
171	static void stereo_split(short output1, short output2, short *input, int n)
172	{
173	int i;
174
175	for(i=0;i<n;i++) {
176	output1++ = input++;
177	output2++ = input++;
178	}
179	}
180
181	static void stereo_mux(short output, short input1, short *input2, int n)
182	{
183	int i;
184
185	for(i=0;i<n;i++) {
186	output++ = input1++;
187	output++ = input2++;
188	}
189	}
190
191	static int mono_resample(ReSampleChannelContext s, short output, short *input, int nb_samples)
192	{
1a565432	193	short *buf1;
de6d9b64 FB	194	short *buftmp;
de6d9b64 FB	195
1a565432 FB	196	buf1= (short) malloc( nb_samples sizeof(short) );
1a565432 FB	197
de6d9b64 FB	198	/* first downsample by an integer factor with averaging filter */
	199	if (s->iratio > 1) {
	200	buftmp = buf1;
	201	nb_samples = integer_downsample(s, buftmp, input, nb_samples);
	202	} else {
	203	buftmp = input;
	204	}
	205
	206	/* then do a fractional resampling with linear interpolation */
	207	if (s->incr != FRAC) {
	208	nb_samples = fractional_resample(s, output, buftmp, nb_samples);
	209	} else {
	210	memcpy(output, buftmp, nb_samples * sizeof(short));
	211	}
1a565432	212	free(buf1);
de6d9b64 FB	213	return nb_samples;
	214	}
	215
	216	ReSampleContext *audio_resample_init(int output_channels, int input_channels,
	217	int output_rate, int input_rate)
	218	{
	219	ReSampleContext *s;
	220	int i;
	221
	222	if (output_channels > 2 \|\| input_channels > 2)
	223	return NULL;
	224
	225	s = av_mallocz(sizeof(ReSampleContext));
	226	if (!s)
	227	return NULL;
	228
	229	s->ratio = (float)output_rate / (float)input_rate;
	230
	231	s->input_channels = input_channels;
	232	s->output_channels = output_channels;
	233
	234	s->filter_channels = s->input_channels;
	235	if (s->output_channels < s->filter_channels)
	236	s->filter_channels = s->output_channels;
	237
	238	for(i=0;i<s->filter_channels;i++) {
	239	init_mono_resample(&s->channel_ctx[i], s->ratio);
	240	}
	241	return s;
	242	}
	243
	244	/* resample audio. 'nb_samples' is the number of input samples */
	245	/* XXX: optimize it ! */
	246	/* XXX: do it with polyphase filters, since the quality here is
	247	HORRIBLE. Return the number of samples available in output */
	248	int audio_resample(ReSampleContext s, short output, short *input, int nb_samples)
	249	{
	250	int i, nb_samples1;
1a565432 FB	251	short *bufin[2];
1a565432 FB	252	short *bufout[2];
de6d9b64	253	short buftmp2[2], buftmp3[2];
1a565432	254	int lenout;
de6d9b64 FB	255
	256	if (s->input_channels == s->output_channels && s->ratio == 1.0) {
	257	/* nothing to do */
	258	memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
	259	return nb_samples;
	260	}
	261
1a565432 FB	262	/* XXX: move those malloc to resample init code */
	263	bufin[0]= (short) malloc( nb_samples sizeof(short) );
	264	bufin[1]= (short) malloc( nb_samples sizeof(short) );
	265
	266	/* make some zoom to avoid round pb */
	267	lenout= (int)(nb_samples * s->ratio) + 16;
	268	bufout[0]= (short) malloc( lenout sizeof(short) );
	269	bufout[1]= (short) malloc( lenout sizeof(short) );
	270
de6d9b64 FB	271	if (s->input_channels == 2 &&
	272	s->output_channels == 1) {
	273	buftmp2[0] = bufin[0];
	274	buftmp3[0] = output;
	275	stereo_to_mono(buftmp2[0], input, nb_samples);
	276	} else if (s->output_channels == 2 && s->input_channels == 1) {
	277	buftmp2[0] = input;
	278	buftmp3[0] = bufout[0];
	279	} else if (s->output_channels == 2) {
	280	buftmp2[0] = bufin[0];
	281	buftmp2[1] = bufin[1];
	282	buftmp3[0] = bufout[0];
	283	buftmp3[1] = bufout[1];
	284	stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
	285	} else {
	286	buftmp2[0] = input;
	287	buftmp3[0] = output;
	288	}
	289
	290	/* resample each channel */
	291	nb_samples1 = 0; /* avoid warning */
	292	for(i=0;i<s->filter_channels;i++) {
	293	nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
	294	}
	295
	296	if (s->output_channels == 2 && s->input_channels == 1) {
	297	mono_to_stereo(output, buftmp3[0], nb_samples1);
	298	} else if (s->output_channels == 2) {
	299	stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
	300	}
	301
1a565432 FB	302	free(bufin[0]);
	303	free(bufin[1]);
	304
	305	free(bufout[0]);
	306	free(bufout[1]);
de6d9b64 FB	307	return nb_samples1;
	308	}
	309
	310	void audio_resample_close(ReSampleContext *s)
	311	{
	312	free(s);
	313	}