[FFmpeg-cvslog] Support decoding of 24 bit True Audio samples-

[Carl Eugen Hoyos]

ffmpeg | branch: master | Carl Eugen Hoyos <cehoyos at ag.or.at> | Tue May  3 01:09:30 2011 +0200| [7d70d1937a5c1a3fda8b988e3bccbb4a04ff3ae7] | committer: Carl Eugen Hoyos

Support decoding of 24 bit True Audio samples-

Many samples will fail because "Output buffer size is too small."

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7d70d1937a5c1a3fda8b988e3bccbb4a04ff3ae7
---

 libavcodec/tta.c |    9 ++++++++-
 1 files changed, 8 insertions(+), 1 deletions(-)

diff --git a/libavcodec/tta.c b/libavcodec/tta.c
index b00c54d..dccca46 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");
         }