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| author | Carl Eugen Hoyos <cehoyos@ag.or.at> | 2011-05-03 01:06:18 +0200 |
|---|---|---|
| committer | Carl Eugen Hoyos <cehoyos@ag.or.at> | 2011-05-03 01:06:18 +0200 |
| commit | 4cf2e30f056f092f4b04d516ee45ceb1ec7f268e (patch) | |
| tree | a54fbc97b350c4a4c8abdcd920a792e3e86ca936 | |
| parent | fba418e6708a5271e933be27f346759e3d616e7c (diff) | |
| download | ffmpeg-4cf2e30f056f092f4b04d516ee45ceb1ec7f268e.tar.gz | |
Support decoding of 8 bit True Audio samples-
Many samples will fail because "Output buffer size is too small."
| -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 217354c863..b00c54d63f 100644 --- a/libavcodec/tta.c +++ b/libavcodec/tta.c @@ -263,7 +263,7 @@ static av_cold int tta_decode_init(AVCodecContext * avctx) return -1; } else switch(s->bps) { -// case 1: avctx->sample_fmt = AV_SAMPLE_FMT_U8; break; + 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 4: avctx->sample_fmt = AV_SAMPLE_FMT_S32; break; @@ -442,6 +442,13 @@ static int tta_decode_frame(AVCodecContext *avctx, // convert to output buffer switch(s->bps) { + case 1: { + uint8_t *samples = data; + for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++) + *samples++ = *p + 0x80; + *data_size = samples - (uint8_t *)data; + break; + } case 2: { uint16_t *samples = data; for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++) { |