From 3240121509662ba1dfd5f918d8e94e89918e8082 Mon Sep 17 00:00:00 2001
From: Limin Wang <lance.lmwang@gmail.com>
Date: Thu, 28 May 2020 23:43:41 +0800
Subject: [PATCH] avcodec/adpcmenc: remove FF_ALLOC_OR_GOTO macros and gotos
 lable

Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
---
 libavcodec/adpcmenc.c | 38 ++++++++++++++++----------------------
 1 file changed, 16 insertions(+), 22 deletions(-)

diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c
index d5fbc0b9a7..1fe1aef6f5 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[] = {