X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fadxenc.c;h=e730811744563a245b2323835aacb04fee88cd29;hb=380146924ecad2e05e9dcc5c3c2e1b5ba47c51e8;hp=b0847f43efb4e181093c30a01f2a19de21c940bf;hpb=2912e87a6c9264d556734e2bf94a99c64cf9b102;p=ffmpeg diff --git a/libavcodec/adxenc.c b/libavcodec/adxenc.c index b0847f43efb..e7308117445 100644 --- a/libavcodec/adxenc.c +++ b/libavcodec/adxenc.c @@ -19,9 +19,11 @@ * 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 @@ -32,166 +34,138 @@ * 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 (maxd) 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; - - av_log(avctx, AV_LOG_DEBUG, "adx encode init\n"); + ADXContext *c = avctx->priv_data; - return 0; -} + if (avctx->channels > 2) { + av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); + return AVERROR(EINVAL); + } + avctx->frame_size = BLOCK_SAMPLES; -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", + .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"), + .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 }, };