diff options
| author | Alex Beregszaszi <alex@rtfs.hu> | 2007-01-19 22:12:59 +0000 |
|---|---|---|
| committer | Alex Beregszaszi <alex@rtfs.hu> | 2007-01-19 22:12:59 +0000 |
| commit | fead30d4440bc7b75006ae60f2742c63a05168b3 (patch) | |
| tree | cead68d924846c1bfb756bb46c86184d889d10db /libavcodec/wavpack.c | |
| parent | 50eaa857c099abda803c803927276d81c2d74edb (diff) | |
| download | ffmpeg-fead30d4440bc7b75006ae60f2742c63a05168b3.tar.gz | |
rename BE/LE_8/16/32 to AV_RL/B_8/16/32
Originally committed as revision 7587 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/wavpack.c')
| -rw-r--r-- | libavcodec/wavpack.c | 24 |
1 files changed, 12 insertions, 12 deletions
diff --git a/libavcodec/wavpack.c b/libavcodec/wavpack.c index 18544831e8..6f5d84bd0c 100644 --- a/libavcodec/wavpack.c +++ b/libavcodec/wavpack.c @@ -387,15 +387,15 @@ static int wavpack_decode_frame(AVCodecContext *avctx, memset(s->decorr, 0, MAX_TERMS * sizeof(Decorr)); - s->samples = LE_32(buf); buf += 4; + s->samples = AV_RL32(buf); buf += 4; if(!s->samples) return buf_size; /* should not happen but who knows */ if(s->samples * 2 * avctx->channels > AVCODEC_MAX_AUDIO_FRAME_SIZE){ av_log(avctx, AV_LOG_ERROR, "Packet size is too big to be handled in lavc!\n"); return -1; } - s->joint = LE_32(buf) & WV_JOINT; buf += 4; - s->CRC = LE_32(buf); buf += 4; + s->joint = AV_RL32(buf) & WV_JOINT; buf += 4; + s->CRC = AV_RL32(buf); buf += 4; // parse metadata blocks while(buf < buf_end){ id = *buf++; @@ -467,23 +467,23 @@ static int wavpack_decode_frame(AVCodecContext *avctx, t = 0; for(i = s->terms - 1; (i >= 0) && (t < size); i--) { if(s->decorr[i].value > 8){ - s->decorr[i].samplesA[0] = wp_exp2(LE_16(buf)); buf += 2; - s->decorr[i].samplesA[1] = wp_exp2(LE_16(buf)); buf += 2; + s->decorr[i].samplesA[0] = wp_exp2(AV_RL16(buf)); buf += 2; + s->decorr[i].samplesA[1] = wp_exp2(AV_RL16(buf)); buf += 2; if(s->stereo){ - s->decorr[i].samplesB[0] = wp_exp2(LE_16(buf)); buf += 2; - s->decorr[i].samplesB[1] = wp_exp2(LE_16(buf)); buf += 2; + s->decorr[i].samplesB[0] = wp_exp2(AV_RL16(buf)); buf += 2; + s->decorr[i].samplesB[1] = wp_exp2(AV_RL16(buf)); buf += 2; t += 4; } t += 4; }else if(s->decorr[i].value < 0){ - s->decorr[i].samplesA[0] = wp_exp2(LE_16(buf)); buf += 2; - s->decorr[i].samplesB[0] = wp_exp2(LE_16(buf)); buf += 2; + s->decorr[i].samplesA[0] = wp_exp2(AV_RL16(buf)); buf += 2; + s->decorr[i].samplesB[0] = wp_exp2(AV_RL16(buf)); buf += 2; t += 4; }else{ for(j = 0; j < s->decorr[i].value; j++){ - s->decorr[i].samplesA[j] = wp_exp2(LE_16(buf)); buf += 2; + s->decorr[i].samplesA[j] = wp_exp2(AV_RL16(buf)); buf += 2; if(s->stereo){ - s->decorr[i].samplesB[j] = wp_exp2(LE_16(buf)); buf += 2; + s->decorr[i].samplesB[j] = wp_exp2(AV_RL16(buf)); buf += 2; } } t += s->decorr[i].value * 2 * avctx->channels; @@ -498,7 +498,7 @@ static int wavpack_decode_frame(AVCodecContext *avctx, continue; } for(i = 0; i < 3 * avctx->channels; i++){ - s->median[i] = wp_exp2(LE_16(buf)); + s->median[i] = wp_exp2(AV_RL16(buf)); buf += 2; } got_entropy = 1; |