* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "adx.h"
+#include "bytestream.h"
+#include "internal.h"
+#include "put_bits.h"
/**
* @file
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
-/* 18 bytes <-> 32 samples */
-
-static void adx_encode(unsigned char *adx,const short *wav,PREV *prev)
+static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
+ ADXChannelState *prev, int channels)
{
+ PutBitContext pb;
int scale;
- int i;
- int s0,s1,s2,d;
- int max=0;
- int min=0;
- int data[32];
+ int i, j;
+ int s0, s1, s2, d;
+ int max = 0;
+ int min = 0;
+ int data[BLOCK_SAMPLES];
s1 = prev->s1;
s2 = prev->s2;
- for(i=0;i<32;i++) {
+ for (i = 0, j = 0; j < 32; i += channels, j++) {
s0 = wav[i];
- d = ((s0<<14) - SCALE1*s1 + SCALE2*s2)/BASEVOL;
- data[i]=d;
- if (max<d) max=d;
- if (min>d) min=d;
+ d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
+ data[j] = d;
+ if (max < d)
+ max = d;
+ if (min > d)
+ min = d;
s2 = s1;
s1 = s0;
}
prev->s1 = s1;
prev->s2 = s2;
- /* -8..+7 */
-
- if (max==0 && min==0) {
- memset(adx,0,18);
+ if (max == 0 && min == 0) {
+ memset(adx, 0, BLOCK_SIZE);
return;
}
- if (max/7>-min/8) scale = max/7;
- else scale = -min/8;
+ if (max / 7 > -min / 8)
+ scale = max / 7;
+ else
+ scale = -min / 8;
- if (scale==0) scale=1;
+ if (scale == 0)
+ scale = 1;
AV_WB16(adx, scale);
- for(i=0;i<16;i++) {
- adx[i+2] = ((data[i*2]/scale)<<4) | ((data[i*2+1]/scale)&0xf);
- }
+ init_put_bits(&pb, adx + 2, 16);
+ for (i = 0; i < BLOCK_SAMPLES; i++)
+ put_sbits(&pb, 4, av_clip(data[i] / scale, -8, 7));
+ flush_put_bits(&pb);
}
-static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t bufsize)
+#define HEADER_SIZE 36
+
+static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
{
-#if 0
- struct {
- uint32_t offset; /* 0x80000000 + sample start - 4 */
- unsigned char unknown1[3]; /* 03 12 04 */
- unsigned char channel; /* 1 or 2 */
- uint32_t freq;
- uint32_t size;
- uint32_t unknown2; /* 01 f4 03 00 */
- uint32_t unknown3; /* 00 00 00 00 */
- uint32_t unknown4; /* 00 00 00 00 */
-
- /* if loop
- unknown3 00 15 00 01
- unknown4 00 00 00 01
- long loop_start_sample;
- long loop_start_byte;
- long loop_end_sample;
- long loop_end_byte;
- long
- */
- } adxhdr; /* big endian */
- /* offset-6 "(c)CRI" */
-#endif
- AV_WB32(buf+0x00,0x80000000|0x20);
- AV_WB32(buf+0x04,0x03120400|avctx->channels);
- AV_WB32(buf+0x08,avctx->sample_rate);
- AV_WB32(buf+0x0c,0); /* FIXME: set after */
- AV_WB32(buf+0x10,0x01040300);
- AV_WB32(buf+0x14,0x00000000);
- AV_WB32(buf+0x18,0x00000000);
- memcpy(buf+0x1c,"\0\0(c)CRI",8);
- return 0x20+4;
+ ADXContext *c = avctx->priv_data;
+
+ bytestream_put_be16(&buf, 0x8000); /* header signature */
+ bytestream_put_be16(&buf, HEADER_SIZE - 4); /* copyright offset */
+ bytestream_put_byte(&buf, 3); /* encoding */
+ bytestream_put_byte(&buf, BLOCK_SIZE); /* block size */
+ bytestream_put_byte(&buf, 4); /* sample size */
+ bytestream_put_byte(&buf, avctx->channels); /* channels */
+ bytestream_put_be32(&buf, avctx->sample_rate); /* sample rate */
+ bytestream_put_be32(&buf, 0); /* total sample count */
+ bytestream_put_be16(&buf, c->cutoff); /* cutoff frequency */
+ bytestream_put_byte(&buf, 3); /* version */
+ bytestream_put_byte(&buf, 0); /* flags */
+ bytestream_put_be32(&buf, 0); /* unknown */
+ bytestream_put_be32(&buf, 0); /* loop enabled */
+ bytestream_put_be16(&buf, 0); /* padding */
+ bytestream_put_buffer(&buf, "(c)CRI", 6); /* copyright signature */
+
+ return HEADER_SIZE;
}
static av_cold int adx_encode_init(AVCodecContext *avctx)
{
- if (avctx->channels > 2)
- return -1; /* only stereo or mono =) */
- avctx->frame_size = 32;
-
- avctx->coded_frame= avcodec_alloc_frame();
- avctx->coded_frame->key_frame= 1;
-
-// avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
+ ADXContext *c = avctx->priv_data;
- av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");
+ if (avctx->channels > 2) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
+ return AVERROR(EINVAL);
+ }
+ avctx->frame_size = BLOCK_SAMPLES;
- return 0;
-}
+#if FF_API_OLD_ENCODE_AUDIO
+ avcodec_get_frame_defaults(&c->frame);
+ avctx->coded_frame = &c->frame;
+#endif
-static av_cold int adx_encode_close(AVCodecContext *avctx)
-{
- av_freep(&avctx->coded_frame);
+ /* the cutoff can be adjusted, but this seems to work pretty well */
+ c->cutoff = 500;
+ ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
return 0;
}
-static int adx_encode_frame(AVCodecContext *avctx,
- uint8_t *frame, int buf_size, void *data)
+static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
- ADXContext *c = avctx->priv_data;
- const short *samples = data;
- unsigned char *dst = frame;
- int rest = avctx->frame_size;
-
-/*
- input data size =
- ffmpeg.c: do_audio_out()
- frame_bytes = enc->frame_size * 2 * enc->channels;
-*/
+ ADXContext *c = avctx->priv_data;
+ const int16_t *samples = (const int16_t *)frame->data[0];
+ uint8_t *dst;
+ int ch, out_size, ret;
+
+ out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
+ if ((ret = ff_alloc_packet(avpkt, out_size)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+ dst = avpkt->data;
-// printf("sz=%d ",buf_size); fflush(stdout);
if (!c->header_parsed) {
- int hdrsize = adx_encode_header(avctx,dst,buf_size);
- dst+=hdrsize;
+ int hdrsize;
+ if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
+ return AVERROR(EINVAL);
+ }
+ dst += hdrsize;
c->header_parsed = 1;
}
- if (avctx->channels==1) {
- while(rest>=32) {
- adx_encode(dst,samples,c->prev);
- dst+=18;
- samples+=32;
- rest-=32;
- }
- } else {
- while(rest>=32*2) {
- short tmpbuf[32*2];
- int i;
-
- for(i=0;i<32;i++) {
- tmpbuf[i] = samples[i*2];
- tmpbuf[i+32] = samples[i*2+1];
- }
-
- adx_encode(dst,tmpbuf,c->prev);
- adx_encode(dst+18,tmpbuf+32,c->prev+1);
- dst+=18*2;
- samples+=32*2;
- rest-=32*2;
- }
+ for (ch = 0; ch < avctx->channels; ch++) {
+ adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
+ dst += BLOCK_SIZE;
}
- return dst-frame;
+
+ *got_packet_ptr = 1;
+ return 0;
}
AVCodec ff_adpcm_adx_encoder = {
- "adpcm_adx",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_ADPCM_ADX,
- sizeof(ADXContext),
- adx_encode_init,
- adx_encode_frame,
- adx_encode_close,
- NULL,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
- .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
+ .name = "adpcm_adx",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_ADPCM_ADX,
+ .priv_data_size = sizeof(ADXContext),
+ .init = adx_encode_init,
+ .encode2 = adx_encode_frame,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
};
projects / ffmpeg / blobdiff