[SCM] libav/experimental: Simplify the FLAC decoder now that it no longer has to parse arbitrary raw FLAC data thanks to the recently added FLAC parser. Patch by Michael Chinen [mchinen at gmail]
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 17:19:55 UTC 2013
The following commit has been merged in the experimental branch:
commit 60a684939ab61dbfdf55105907808e015e5f6ec2
Author: Michael Chinen <mchinen at gmail.com>
Date: Tue Dec 7 14:57:02 2010 +0000
Simplify the FLAC decoder now that it no longer has to parse arbitrary raw FLAC
data thanks to the recently added FLAC parser.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25915 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index 36f41de..71a2c58 100644
--- a/libavcodec/flacdec.c
+++ b/libavcodec/flacdec.c
@@ -61,10 +61,6 @@ typedef struct FLACContext {
int got_streaminfo; ///< indicates if the STREAMINFO has been read
int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
- uint8_t *bitstream;
- unsigned int bitstream_size;
- unsigned int bitstream_index;
- unsigned int allocated_bitstream_size;
} FLACContext;
static void allocate_buffers(FLACContext *s);
@@ -140,20 +136,10 @@ static void allocate_buffers(FLACContext *s)
assert(s->max_blocksize);
- if (s->max_framesize == 0 && s->max_blocksize) {
- s->max_framesize = ff_flac_get_max_frame_size(s->max_blocksize,
- s->channels, s->bps);
- }
-
for (i = 0; i < s->channels; i++) {
s->decoded[i] = av_realloc(s->decoded[i],
sizeof(int32_t)*s->max_blocksize);
}
-
- if (s->allocated_bitstream_size < s->max_framesize)
- s->bitstream= av_fast_realloc(s->bitstream,
- &s->allocated_bitstream_size,
- s->max_framesize);
}
void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
@@ -563,7 +549,7 @@ static int flac_decode_frame(AVCodecContext *avctx,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
FLACContext *s = avctx->priv_data;
- int i, j = 0, input_buf_size = 0, bytes_read = 0;
+ int i, j = 0, bytes_read = 0;
int16_t *samples_16 = data;
int32_t *samples_32 = data;
int alloc_data_size= *data_size;
@@ -575,41 +561,13 @@ static int flac_decode_frame(AVCodecContext *avctx,
s->max_framesize =
ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE,
FLAC_MAX_CHANNELS, 32);
- s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
- }
-
- if (1 && s->max_framesize) { //FIXME truncated
- if (s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C'))
- buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize));
- input_buf_size= buf_size;
-
- if (s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index)
- return -1;
-
- if (s->allocated_bitstream_size < s->bitstream_size + buf_size)
- s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_size + buf_size);
-
- if (s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size) {
- memmove(s->bitstream, &s->bitstream[s->bitstream_index],
- s->bitstream_size);
- s->bitstream_index=0;
- }
- memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size],
- buf, buf_size);
- buf= &s->bitstream[s->bitstream_index];
- buf_size += s->bitstream_size;
- s->bitstream_size= buf_size;
-
- if (buf_size < s->max_framesize && input_buf_size) {
- return input_buf_size;
- }
}
/* check that there is at least the smallest decodable amount of data.
this amount corresponds to the smallest valid FLAC frame possible.
FF F8 69 02 00 00 9A 00 00 34 46 */
if (buf_size < 11)
- goto end;
+ return buf_size;
/* check for inline header */
if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
@@ -617,26 +575,13 @@ static int flac_decode_frame(AVCodecContext *avctx,
av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
return -1;
}
- bytes_read = get_metadata_size(buf, buf_size);
- goto end;
- }
-
- /* check for frame sync code and resync stream if necessary */
- if ((AV_RB16(buf) & 0xFFFE) != 0xFFF8) {
- const uint8_t *buf_end = buf + buf_size;
- av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
- while (buf+2 < buf_end && (AV_RB16(buf) & 0xFFFE) != 0xFFF8)
- buf++;
- bytes_read = buf_size - (buf_end - buf);
- goto end; // we may not have enough bits left to decode a frame, so try next time
+ return get_metadata_size(buf, buf_size);
}
/* decode frame */
init_get_bits(&s->gb, buf, buf_size*8);
if (decode_frame(s) < 0) {
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
- s->bitstream_size=0;
- s->bitstream_index=0;
return -1;
}
bytes_read = (get_bits_count(&s->gb)+7)/8;
@@ -646,7 +591,7 @@ static int flac_decode_frame(AVCodecContext *avctx,
if (output_size > alloc_data_size) {
av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
"allocated data size\n");
- goto end;
+ return -1;
}
*data_size = output_size;
@@ -684,20 +629,16 @@ static int flac_decode_frame(AVCodecContext *avctx,
DECORRELATE( (a-=b>>1) + b, a)
}
-end:
if (bytes_read > buf_size) {
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
- s->bitstream_size=0;
- s->bitstream_index=0;
return -1;
}
+ if (bytes_read < buf_size) {
+ av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n",
+ buf_size - bytes_read, buf_size);
+ }
- if (s->bitstream_size) {
- s->bitstream_index += bytes_read;
- s->bitstream_size -= bytes_read;
- return input_buf_size;
- } else
- return bytes_read;
+ return bytes_read;
}
static av_cold int flac_decode_close(AVCodecContext *avctx)
@@ -708,19 +649,10 @@ static av_cold int flac_decode_close(AVCodecContext *avctx)
for (i = 0; i < s->channels; i++) {
av_freep(&s->decoded[i]);
}
- av_freep(&s->bitstream);
return 0;
}
-static void flac_flush(AVCodecContext *avctx)
-{
- FLACContext *s = avctx->priv_data;
-
- s->bitstream_size=
- s->bitstream_index= 0;
-}
-
AVCodec flac_decoder = {
"flac",
AVMEDIA_TYPE_AUDIO,
@@ -730,9 +662,5 @@ AVCodec flac_decoder = {
NULL,
flac_decode_close,
flac_decode_frame,
- CODEC_CAP_DELAY | CODEC_CAP_SUBFRAMES, /* FIXME: add a FLAC parser so that
- we will not need to use either
- of these capabilities */
- .flush= flac_flush,
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};
--
Libav/FFmpeg packaging
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