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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

adpcmenc: update to AVCodec.encode2()

This commit is contained in:
Justin Ruggles 2012-01-31 15:54:45 -05:00
parent aa872af5e3
commit 24e74f0a0f

View File

@ -24,6 +24,7 @@
#include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
#include "internal.h"
/**
* @file
@ -141,8 +142,10 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
goto error;
#endif
return 0;
error:
@ -156,7 +159,9 @@ error:
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
@ -470,23 +475,31 @@ static void adpcm_compress_trellis(AVCodecContext *avctx,
c->idelta = nodes[0]->step;
}
static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int n, i, st;
int16_t *samples;
int n, i, st, pkt_size, ret;
const int16_t *samples;
uint8_t *dst;
ADPCMEncodeContext *c = avctx->priv_data;
uint8_t *buf;
dst = frame;
samples = data;
samples = (const int16_t *)frame->data[0];
st = avctx->channels == 2;
/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
if (avctx->codec_id == CODEC_ID_ADPCM_SWF)
pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
else
pkt_size = avctx->block_align;
if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
dst = avpkt->data;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_WAV:
n = avctx->frame_size / 8;
n = frame->nb_samples / 8;
c->status[0].prev_sample = samples[0];
/* c->status[0].step_index = 0;
XXX: not sure how to init the state machine */
@ -554,7 +567,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
{
int ch, i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size * 8);
init_put_bits(&pb, dst, pkt_size * 8);
for (ch = 0; ch < avctx->channels; ch++) {
put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
@ -578,16 +591,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
flush_put_bits(&pb);
dst += put_bits_count(&pb) >> 3;
break;
}
case CODEC_ID_ADPCM_SWF:
{
int i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size * 8);
init_put_bits(&pb, dst, pkt_size * 8);
n = avctx->frame_size - 1;
n = frame->nb_samples - 1;
// store AdpcmCodeSize
put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
@ -614,7 +626,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
av_free(buf);
} else {
for (i = 1; i < avctx->frame_size; i++) {
for (i = 1; i < frame->nb_samples; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
samples[avctx->channels * i]));
if (avctx->channels == 2)
@ -623,7 +635,6 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
}
flush_put_bits(&pb);
dst += put_bits_count(&pb) >> 3;
break;
}
case CODEC_ID_ADPCM_MS:
@ -671,7 +682,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
break;
case CODEC_ID_ADPCM_YAMAHA:
n = avctx->frame_size / 2;
n = frame->nb_samples / 2;
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
n *= 2;
@ -697,7 +708,10 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
default:
return AVERROR(EINVAL);
}
return dst - frame;
avpkt->size = pkt_size;
*got_packet_ptr = 1;
return 0;
error:
return AVERROR(ENOMEM);
}
@ -710,7 +724,7 @@ AVCodec ff_ ## name_ ## _encoder = { \
.id = id_, \
.priv_data_size = sizeof(ADPCMEncodeContext), \
.init = adpcm_encode_init, \
.encode = adpcm_encode_frame, \
.encode2 = adpcm_encode_frame, \
.close = adpcm_encode_close, \
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, \
AV_SAMPLE_FMT_NONE}, \