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authorJustin Ruggles <justin.ruggles@gmail.com>2008-09-14 20:00:36 +0000
committerJustin Ruggles <justin.ruggles@gmail.com>2008-09-14 20:00:36 +0000
commite1a12934c5fd97e1e41887aa907ed61dd3d18620 (patch)
tree3753487564d7eca83e766dcd8e04231967e9f46e /libavcodec
parent8588e1484eff061b1e2dba12ebc2377d48d966ee (diff)
downloadffmpeg-e1a12934c5fd97e1e41887aa907ed61dd3d18620.tar.gz
write number of samples in FLAC extradata.
based on a patch by Mathieu Velten (matmaul gmail com). Originally committed as revision 15324 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec')
-rw-r--r--libavcodec/flacenc.c15
1 files changed, 13 insertions, 2 deletions
diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index 1bba6f84cd..d1002f3814 100644
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -96,6 +96,7 @@ typedef struct FlacEncodeContext {
int sr_code[2];
int max_framesize;
uint32_t frame_count;
+ uint64_t sample_count;
FlacFrame frame;
CompressionOptions options;
AVCodecContext *avctx;
@@ -134,8 +135,10 @@ static void write_streaminfo(FlacEncodeContext *s, uint8_t *header)
put_bits(&pb, 20, s->samplerate);
put_bits(&pb, 3, s->channels-1);
put_bits(&pb, 5, 15); /* bits per sample - 1 */
+ /* write 36-bit sample count in 2 put_bits() calls */
+ put_bits(&pb, 24, (s->sample_count & 0xFFFFFF000LL) >> 12);
+ put_bits(&pb, 12, s->sample_count & 0x000000FFFLL);
flush_put_bits(&pb);
- /* total samples = 0 */
/* MD5 signature = 0 */
}
@@ -1251,6 +1254,12 @@ static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
return 0;
}
+ /* when the last block is reached, update the header in extradata */
+ if (!data) {
+ write_streaminfo(s, avctx->extradata);
+ return 0;
+ }
+
init_frame(s);
copy_samples(s, samples);
@@ -1284,6 +1293,8 @@ write_frame:
}
s->frame_count++;
+ s->sample_count += avctx->frame_size;
+
return out_bytes;
}
@@ -1304,7 +1315,7 @@ AVCodec flac_encoder = {
flac_encode_frame,
flac_encode_close,
NULL,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};