openharmony/third_party_ffmpeg
1
0
Fork 0
mirror of https://gitee.com/openharmony/third_party_ffmpeg synced 2024-11-23 11:19:55 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity

Merge commit '45ee556d51ef04d79d52bf6b0b7f28a4d231cb0c'

Browse Source 
* commit '45ee556d51ef04d79d52bf6b0b7f28a4d231cb0c':
  qdm2: Whitespace cosmetics
  flac: use meaningful return values

Conflicts:
	libavcodec/flacdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2013-07-29 11:34:07 +02:00
commit 5f312139df
3 changed files with 17 additions and 17 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

16
libavcodec/flac.c
View File

@ -55,7 +55,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
/* frame sync code */
if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n");
return -1;
return AVERROR_INVALIDDATA;
}
/* variable block size stream code */
@ -76,7 +76,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
} else {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"invalid channel mode: %d\n", fi->ch_mode);
return -1;
return AVERROR_INVALIDDATA;
}
/* bits per sample */
@ -85,7 +85,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"invalid sample size code (%d)\n",
bps_code);
return -1;
return AVERROR_INVALIDDATA;
}
fi->bps = sample_size_table[bps_code];
@ -93,7 +93,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
if (get_bits1(gb)) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"broken stream, invalid padding\n");
return -1;
return AVERROR_INVALIDDATA;
}
/* sample or frame count */
@ -101,14 +101,14 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
if (fi->frame_or_sample_num < 0) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"sample/frame number invalid; utf8 fscked\n");
return -1;
return AVERROR_INVALIDDATA;
}
/* blocksize */
if (bs_code == 0) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"reserved blocksize code: 0\n");
return -1;
return AVERROR_INVALIDDATA;
} else if (bs_code == 6) {
fi->blocksize = get_bits(gb, 8) + 1;
} else if (bs_code == 7) {
@ -130,7 +130,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"illegal sample rate code %d\n",
sr_code);
return -1;
return AVERROR_INVALIDDATA;
}
/* header CRC-8 check */
@ -139,7 +139,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
get_bits_count(gb)/8)) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"header crc mismatch\n");
return -1;
return AVERROR_INVALIDDATA;
}
return 0;

10
libavcodec/flacdec.c
View File

@ -409,9 +409,9 @@ static int decode_frame(FLACContext *s)
GetBitContext *gb = &s->gb;
FLACFrameInfo fi;
if (ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) {
if ((ret = ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) < 0) {
av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n");
return AVERROR_INVALIDDATA;
return ret;
}
if (s->channels && fi.channels != s->channels && s->got_streaminfo) {
@ -435,7 +435,7 @@ static int decode_frame(FLACContext *s)
} else if (s->bps && fi.bps != s->bps) {
av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
"supported\n");
return -1;
return AVERROR_INVALIDDATA;
}
if (!s->bps) {
@ -523,9 +523,9 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data,
/* check for inline header */
if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
if (!s->got_streaminfo && parse_streaminfo(s, buf, buf_size)) {
if (!s->got_streaminfo && (ret = parse_streaminfo(s, buf, buf_size))) {
av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
return AVERROR_INVALIDDATA;
return ret;
}
return get_metadata_size(buf, buf_size);
}

8
libavcodec/qdm2.c
View File

@ -828,7 +828,7 @@ static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
if (length == 0) {
// If no data use noise
for (sb=sb_min; sb < sb_max; sb++)
build_sb_samples_from_noise (q, sb);
build_sb_samples_from_noise(q, sb);
return 0;
}
@ -841,12 +841,12 @@ static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
else if (sb >= 24)
joined_stereo = 1;
else
joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
if (joined_stereo) {
if (get_bits_left(gb) >= 16)
for (j = 0; j < 16; j++)
sign_bits[j] = get_bits1 (gb);
sign_bits[j] = get_bits1(gb);
for (j = 0; j < 64; j++)
if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
@ -1071,7 +1071,7 @@ static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
* @param q context
* @param gb bitreader context
*/
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
{
int sb, j, k, n, ch;