[FFmpeg-devel] [PATCH v4 04/24] avcodec/adpcmenc: remove FF_ALLOC_OR_GOTO macros for gotos lable
lance.lmwang at gmail.com
lance.lmwang at gmail.com
Tue Jun 2 18:26:38 EEST 2020
From: Limin Wang <lance.lmwang at gmail.com>
Signed-off-by: Limin Wang <lance.lmwang at gmail.com>
---
libavcodec/adpcmenc.c | 38 ++++++++++++++++----------------------
1 file changed, 16 insertions(+), 22 deletions(-)
diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c
index d5fbc0b..1fe1aef 100644
--- a/libavcodec/adpcmenc.c
+++ b/libavcodec/adpcmenc.c
@@ -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[] = {
--
1.8.3.1
More information about the ffmpeg-devel
mailing list