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Code Issues Releases Activity

avcodec/adpcmenc: remove FF_ALLOC_OR_GOTO macros and gotos lable

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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
This commit is contained in:
Limin Wang 2020-05-28 23:43:41 +08:00
parent 0a1dc81723
commit 3240121509
1 changed files with 16 additions and 22 deletions
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38
libavcodec/adpcmenc.c
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@ -65,7 +65,6 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
ADPCMEncodeContext *s = avctx->priv_data;
uint8_t *extradata;
int i;
int ret = AVERROR(ENOMEM);
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
@ -89,14 +88,11 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
if (avctx->trellis) {
int frontier = 1 << avctx->trellis;
int max_paths = frontier * FREEZE_INTERVAL;
FF_ALLOC_OR_GOTO(avctx, s->paths,
max_paths * sizeof(*s->paths), error);
FF_ALLOC_OR_GOTO(avctx, s->node_buf,
2 * frontier * sizeof(*s->node_buf), error);
FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
2 * frontier * sizeof(*s->nodep_buf), error);
FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
65536 * sizeof(*s->trellis_hash), error);
if (!FF_ALLOC_TYPED_ARRAY(s->paths, max_paths) ||
!FF_ALLOC_TYPED_ARRAY(s->node_buf, 2 * frontier) ||
!FF_ALLOC_TYPED_ARRAY(s->nodep_buf, 2 * frontier) ||
!FF_ALLOC_TYPED_ARRAY(s->trellis_hash, 65536))
return AVERROR(ENOMEM);
}
avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
@ -123,7 +119,7 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
avctx->bits_per_coded_sample = 4;
avctx->block_align = BLKSIZE;
if (!(avctx->extradata = av_malloc(32 + AV_INPUT_BUFFER_PADDING_SIZE)))
goto error;
return AVERROR(ENOMEM);
avctx->extradata_size = 32;
extradata = avctx->extradata;
bytestream_put_le16(&extradata, avctx->frame_size);
@ -143,8 +139,7 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
avctx->sample_rate != 44100) {
av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
"22050 or 44100\n");
ret = AVERROR(EINVAL);
goto error;
return AVERROR(EINVAL);
}
avctx->frame_size = 512 * (avctx->sample_rate / 11025);
break;
@ -153,13 +148,10 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
avctx->block_align = BLKSIZE;
break;
default:
ret = AVERROR(EINVAL);
goto error;
return AVERROR(EINVAL);
}
return 0;
error:
return ret;
}
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
@ -523,7 +515,8 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
/* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
if (avctx->trellis > 0) {
FF_ALLOC_ARRAY_OR_GOTO(avctx, buf, avctx->channels, blocks * 8, error);
if (!FF_ALLOC_TYPED_ARRAY(buf, avctx->channels * blocks * 8))
return AVERROR(ENOMEM);
for (ch = 0; ch < avctx->channels; ch++) {
adpcm_compress_trellis(avctx, &samples_p[ch][1],
buf + ch * blocks * 8, &c->status[ch],
@ -618,7 +611,8 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
if (!(buf = av_malloc(2 * n)))
return AVERROR(ENOMEM);
adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
&c->status[0], n, avctx->channels);
if (avctx->channels == 2)
@ -666,7 +660,8 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
if (avctx->trellis > 0) {
n = avctx->block_align - 7 * avctx->channels;
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
if (!(buf = av_malloc(2 * n)))
return AVERROR(ENOMEM);
if (avctx->channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
avctx->channels);
@ -693,7 +688,8 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
case AV_CODEC_ID_ADPCM_YAMAHA:
n = frame->nb_samples / 2;
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
if (!(buf = av_malloc(2 * n * 2)))
return AVERROR(ENOMEM);
n *= 2;
if (avctx->channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
@ -724,8 +720,6 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
avpkt->size = pkt_size;
*got_packet_ptr = 1;
return 0;
error:
return AVERROR(ENOMEM);
}
static const enum AVSampleFormat sample_fmts[] = {