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author | Reimar Döffinger <Reimar.Doeffinger@gmx.de> | 2011-04-30 23:19:04 +0200 |
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committer | Reimar Döffinger <Reimar.Doeffinger@gmx.de> | 2011-05-01 19:13:01 +0200 |
commit | 636ee66f1c188de9d92cb794f8765390d8177c42 (patch) | |
tree | cafa7581984e703e786453efe855297d3fc93d87 | |
parent | 35fe66abbc9a6d151cedbc8d0261dc007aa71fe2 (diff) | |
download | ffmpeg-636ee66f1c188de9d92cb794f8765390d8177c42.tar.gz |
Fix data_size handling for AC3 and dca decoders.
They use now code identical to the AAC decoder.
The AC3 decoder previously did not check the data_size and
the dca decoder checked against and set wrong values for float.
-rw-r--r-- | libavcodec/ac3dec.c | 9 | ||||
-rw-r--r-- | libavcodec/dca.c | 8 |
2 files changed, 12 insertions, 5 deletions
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 431f67dc23..b4aae2263a 100644 --- a/libavcodec/ac3dec.c +++ b/libavcodec/ac3dec.c @@ -1298,6 +1298,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, float *out_samples_flt = (float *)data; int16_t *out_samples = (int16_t *)data; int blk, ch, err; + int data_size_orig, data_size_tmp; const uint8_t *channel_map; const float *output[AC3_MAX_CHANNELS]; @@ -1314,6 +1315,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, init_get_bits(&s->gbc, buf, buf_size * 8); /* parse the syncinfo */ + data_size_orig = *data_size; *data_size = 0; err = parse_frame_header(s); @@ -1397,6 +1399,11 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on]; for (ch = 0; ch < s->out_channels; ch++) output[ch] = s->output[channel_map[ch]]; + data_size_tmp = s->num_blocks * 256 * avctx->channels; + data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples); + if (data_size_orig < data_size_tmp) + return -1; + *data_size = data_size_tmp; for (blk = 0; blk < s->num_blocks; blk++) { if (!err && decode_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n"); @@ -1410,8 +1417,6 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, out_samples += 256 * s->out_channels; } } - *data_size = s->num_blocks * 256 * avctx->channels; - *data_size *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples); return FFMIN(buf_size, s->frame_size); } diff --git a/libavcodec/dca.c b/libavcodec/dca.c index 1e26eedc3c..03bf7f7b26 100644 --- a/libavcodec/dca.c +++ b/libavcodec/dca.c @@ -1622,6 +1622,7 @@ static int dca_decode_frame(AVCodecContext * avctx, { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; + int data_size_tmp; int lfe_samples; int num_core_channels = 0; @@ -1813,10 +1814,11 @@ static int dca_decode_frame(AVCodecContext * avctx, return -1; } - /* ffdshow custom code */ - if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(samples[0]) * channels) + data_size_tmp = (s->sample_blocks / 8) * 256 * channels; + data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*samples_flt) : sizeof(*samples); + if (*data_size < data_size_tmp) return -1; - *data_size = 256 / 8 * s->sample_blocks * sizeof(samples[0]) * channels; + *data_size = data_size_tmp; /* filter to get final output */ for (i = 0; i < (s->sample_blocks / 8); i++) { |