Commit 5f312139 authored by Michael Niedermayer's avatar Michael Niedermayer

Merge commit '45ee556d'

* commit '45ee556d':
  qdm2: Whitespace cosmetics
  flac: use meaningful return values

Conflicts:
	libavcodec/flacdec.c
Merged-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents 03853b10 45ee556d
...@@ -55,7 +55,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, ...@@ -55,7 +55,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
/* frame sync code */ /* frame sync code */
if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) { if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n"); av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n");
return -1; return AVERROR_INVALIDDATA;
} }
/* variable block size stream code */ /* variable block size stream code */
...@@ -76,7 +76,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, ...@@ -76,7 +76,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
} else { } else {
av_log(avctx, AV_LOG_ERROR + log_level_offset, av_log(avctx, AV_LOG_ERROR + log_level_offset,
"invalid channel mode: %d\n", fi->ch_mode); "invalid channel mode: %d\n", fi->ch_mode);
return -1; return AVERROR_INVALIDDATA;
} }
/* bits per sample */ /* bits per sample */
...@@ -85,7 +85,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, ...@@ -85,7 +85,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
av_log(avctx, AV_LOG_ERROR + log_level_offset, av_log(avctx, AV_LOG_ERROR + log_level_offset,
"invalid sample size code (%d)\n", "invalid sample size code (%d)\n",
bps_code); bps_code);
return -1; return AVERROR_INVALIDDATA;
} }
fi->bps = sample_size_table[bps_code]; fi->bps = sample_size_table[bps_code];
...@@ -93,7 +93,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, ...@@ -93,7 +93,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
if (get_bits1(gb)) { if (get_bits1(gb)) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, av_log(avctx, AV_LOG_ERROR + log_level_offset,
"broken stream, invalid padding\n"); "broken stream, invalid padding\n");
return -1; return AVERROR_INVALIDDATA;
} }
/* sample or frame count */ /* sample or frame count */
...@@ -101,14 +101,14 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, ...@@ -101,14 +101,14 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
if (fi->frame_or_sample_num < 0) { if (fi->frame_or_sample_num < 0) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, av_log(avctx, AV_LOG_ERROR + log_level_offset,
"sample/frame number invalid; utf8 fscked\n"); "sample/frame number invalid; utf8 fscked\n");
return -1; return AVERROR_INVALIDDATA;
} }
/* blocksize */ /* blocksize */
if (bs_code == 0) { if (bs_code == 0) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, av_log(avctx, AV_LOG_ERROR + log_level_offset,
"reserved blocksize code: 0\n"); "reserved blocksize code: 0\n");
return -1; return AVERROR_INVALIDDATA;
} else if (bs_code == 6) { } else if (bs_code == 6) {
fi->blocksize = get_bits(gb, 8) + 1; fi->blocksize = get_bits(gb, 8) + 1;
} else if (bs_code == 7) { } else if (bs_code == 7) {
...@@ -130,7 +130,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, ...@@ -130,7 +130,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
av_log(avctx, AV_LOG_ERROR + log_level_offset, av_log(avctx, AV_LOG_ERROR + log_level_offset,
"illegal sample rate code %d\n", "illegal sample rate code %d\n",
sr_code); sr_code);
return -1; return AVERROR_INVALIDDATA;
} }
/* header CRC-8 check */ /* header CRC-8 check */
...@@ -139,7 +139,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, ...@@ -139,7 +139,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
get_bits_count(gb)/8)) { get_bits_count(gb)/8)) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, av_log(avctx, AV_LOG_ERROR + log_level_offset,
"header crc mismatch\n"); "header crc mismatch\n");
return -1; return AVERROR_INVALIDDATA;
} }
return 0; return 0;
......
...@@ -409,9 +409,9 @@ static int decode_frame(FLACContext *s) ...@@ -409,9 +409,9 @@ static int decode_frame(FLACContext *s)
GetBitContext *gb = &s->gb; GetBitContext *gb = &s->gb;
FLACFrameInfo fi; 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"); 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) { if (s->channels && fi.channels != s->channels && s->got_streaminfo) {
...@@ -435,7 +435,7 @@ static int decode_frame(FLACContext *s) ...@@ -435,7 +435,7 @@ static int decode_frame(FLACContext *s)
} else if (s->bps && fi.bps != s->bps) { } else if (s->bps && fi.bps != s->bps) {
av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not " av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
"supported\n"); "supported\n");
return -1; return AVERROR_INVALIDDATA;
} }
if (!s->bps) { if (!s->bps) {
...@@ -523,9 +523,9 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, ...@@ -523,9 +523,9 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data,
/* check for inline header */ /* check for inline header */
if (AV_RB32(buf) == MKBETAG('f','L','a','C')) { 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"); av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
return AVERROR_INVALIDDATA; return ret;
} }
return get_metadata_size(buf, buf_size); return get_metadata_size(buf, buf_size);
} }
......
...@@ -828,7 +828,7 @@ static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, ...@@ -828,7 +828,7 @@ static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
if (length == 0) { if (length == 0) {
// If no data use noise // If no data use noise
for (sb=sb_min; sb < sb_max; sb++) for (sb=sb_min; sb < sb_max; sb++)
build_sb_samples_from_noise (q, sb); build_sb_samples_from_noise(q, sb);
return 0; return 0;
} }
...@@ -841,12 +841,12 @@ static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, ...@@ -841,12 +841,12 @@ static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
else if (sb >= 24) else if (sb >= 24)
joined_stereo = 1; joined_stereo = 1;
else 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 (joined_stereo) {
if (get_bits_left(gb) >= 16) if (get_bits_left(gb) >= 16)
for (j = 0; j < 16; j++) for (j = 0; j < 16; j++)
sign_bits[j] = get_bits1 (gb); sign_bits[j] = get_bits1(gb);
for (j = 0; j < 64; j++) for (j = 0; j < 64; j++)
if (q->coding_method[1][sb][j] > q->coding_method[0][sb][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, ...@@ -1071,7 +1071,7 @@ static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
* @param q context * @param q context
* @param gb bitreader 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; int sb, j, k, n, ch;
......
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