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author | Carl Eugen Hoyos <cehoyos@ag.or.at> | 2011-05-03 01:09:30 +0200 |
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committer | Carl Eugen Hoyos <cehoyos@ag.or.at> | 2011-05-03 01:09:30 +0200 |
commit | 7d70d1937a5c1a3fda8b988e3bccbb4a04ff3ae7 (patch) | |
tree | 9badfdae42f15fc1eedda265495e31d2262a5b8d /libavcodec | |
parent | 4cf2e30f056f092f4b04d516ee45ceb1ec7f268e (diff) | |
download | ffmpeg-7d70d1937a5c1a3fda8b988e3bccbb4a04ff3ae7.tar.gz |
Support decoding of 24 bit True Audio samples-
Many samples will fail because "Output buffer size is too small."
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/tta.c | 9 |
1 files changed, 8 insertions, 1 deletions
diff --git a/libavcodec/tta.c b/libavcodec/tta.c index b00c54d63f..dccca46132 100644 --- a/libavcodec/tta.c +++ b/libavcodec/tta.c @@ -265,7 +265,7 @@ static av_cold int tta_decode_init(AVCodecContext * avctx) else switch(s->bps) { case 1: avctx->sample_fmt = AV_SAMPLE_FMT_U8; break; case 2: avctx->sample_fmt = AV_SAMPLE_FMT_S16; break; -// case 3: avctx->sample_fmt = AV_SAMPLE_FMT_S24; break; + case 3: avctx->bits_per_coded_sample = 24; case 4: avctx->sample_fmt = AV_SAMPLE_FMT_S32; break; default: av_log_ask_for_sample(s->avctx, @@ -459,6 +459,13 @@ static int tta_decode_frame(AVCodecContext *avctx, *data_size = (uint8_t *)samples - (uint8_t *)data; break; } + case 3: { + int32_t *samples = data; + for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++) + *samples++ = AV_RN32(p) << 8; + *data_size = (uint8_t *)samples - (uint8_t *)data; + break; + } default: av_log(s->avctx, AV_LOG_ERROR, "Error, only 16bit samples supported!\n"); } |