[FFmpeg-cvslog] adpcmdec: use planar sample format for adpcm_xa
Justin Ruggles
git at videolan.org
Wed Oct 10 13:05:22 CEST 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Thu Aug 30 18:08:59 2012 -0400| [d2b6ae02aa4d80fb19137ec11f3cefb9f71b9b81] | committer: Justin Ruggles
adpcmdec: use planar sample format for adpcm_xa
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d2b6ae02aa4d80fb19137ec11f3cefb9f71b9b81
---
libavcodec/adpcm.c | 44 ++++++++++++++++++++++++++++----------------
1 file changed, 28 insertions(+), 16 deletions(-)
diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c
index fca5f1b..92c7943 100644
--- a/libavcodec/adpcm.c
+++ b/libavcodec/adpcm.c
@@ -139,6 +139,7 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
case AV_CODEC_ID_ADPCM_IMA_QT:
case AV_CODEC_ID_ADPCM_IMA_WAV:
case AV_CODEC_ID_ADPCM_4XM:
+ case AV_CODEC_ID_ADPCM_XA:
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
break;
case AV_CODEC_ID_ADPCM_IMA_WS:
@@ -277,17 +278,22 @@ static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned c
return c->predictor;
}
-static int xa_decode(AVCodecContext *avctx,
- short *out, const unsigned char *in,
- ADPCMChannelStatus *left, ADPCMChannelStatus *right, int inc)
+static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
+ const uint8_t *in, ADPCMChannelStatus *left,
+ ADPCMChannelStatus *right, int channels, int sample_offset)
{
int i, j;
int shift,filter,f0,f1;
int s_1,s_2;
int d,s,t;
- for(i=0;i<4;i++) {
+ out0 += sample_offset;
+ if (channels == 1)
+ out1 = out0 + 28;
+ else
+ out1 += sample_offset;
+ for(i=0;i<4;i++) {
shift = 12 - (in[4+i*2] & 15);
filter = in[4+i*2] >> 4;
if (filter > 4) {
@@ -309,16 +315,14 @@ static int xa_decode(AVCodecContext *avctx,
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
- *out = s_1;
- out += inc;
+ out0[j] = s_1;
}
- if (inc==2) { /* stereo */
+ if (channels == 2) {
left->sample1 = s_1;
left->sample2 = s_2;
s_1 = right->sample1;
s_2 = right->sample2;
- out = out + 1 - 28*2;
}
shift = 12 - (in[5+i*2] & 15);
@@ -339,18 +343,19 @@ static int xa_decode(AVCodecContext *avctx,
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
- *out = s_1;
- out += inc;
+ out1[j] = s_1;
}
- if (inc==2) { /* stereo */
+ if (channels == 2) {
right->sample1 = s_1;
right->sample2 = s_2;
- out -= 1;
} else {
left->sample1 = s_1;
left->sample2 = s_2;
}
+
+ out0 += 28 * (3 - channels);
+ out1 += 28 * (3 - channels);
}
return 0;
@@ -887,14 +892,21 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
bytestream2_seek(&gb, 0, SEEK_END);
break;
case AV_CODEC_ID_ADPCM_XA:
+ {
+ int16_t *out0 = samples_p[0];
+ int16_t *out1 = samples_p[1];
+ int samples_per_block = 28 * (3 - avctx->channels) * 4;
+ int sample_offset = 0;
while (bytestream2_get_bytes_left(&gb) >= 128) {
- if ((ret = xa_decode(avctx, samples, buf + bytestream2_tell(&gb), &c->status[0],
- &c->status[1], avctx->channels)) < 0)
+ if ((ret = xa_decode(avctx, out0, out1, buf + bytestream2_tell(&gb),
+ &c->status[0], &c->status[1],
+ avctx->channels, sample_offset)) < 0)
return ret;
bytestream2_skipu(&gb, 128);
- samples += 28 * 8;
+ sample_offset += samples_per_block;
}
break;
+ }
case AV_CODEC_ID_ADPCM_IMA_EA_EACS:
for (i=0; i<=st; i++) {
c->status[i].step_index = bytestream2_get_le32u(&gb);
@@ -1318,5 +1330,5 @@ ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_3, sample_fmts_s16, adpcm_sbpro_3,
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_4, sample_fmts_s16, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SWF, sample_fmts_s16, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_THP, sample_fmts_s16, adpcm_thp, "ADPCM Nintendo Gamecube THP");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, sample_fmts_s16, adpcm_xa, "ADPCM CDROM XA");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, sample_fmts_s16p, adpcm_xa, "ADPCM CDROM XA");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_YAMAHA, sample_fmts_s16, adpcm_yamaha, "ADPCM Yamaha");
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