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author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-11-20 14:21:32 -0500 |
---|---|---|
committer | Justin Ruggles <justin.ruggles@gmail.com> | 2011-11-26 16:25:06 -0500 |
commit | b237248e297760648307356efa5fcbfe16844829 (patch) | |
tree | 76ac03216c87ddc68904f9abe3e1e5a89c25604b /libavcodec/adxenc.c | |
parent | 954d94dd5e13ba7a5e9e049d0f980bddced9644c (diff) | |
download | ffmpeg-b237248e297760648307356efa5fcbfe16844829.tar.gz |
adx: calculate correct LPC coeffs
Instead of using fixed coefficients, the correct way is to calculate the
coefficients using the highpass cutoff frequency from the ADX stream header
and the sample rate.
Diffstat (limited to 'libavcodec/adxenc.c')
-rw-r--r-- | libavcodec/adxenc.c | 27 |
1 files changed, 18 insertions, 9 deletions
diff --git a/libavcodec/adxenc.c b/libavcodec/adxenc.c index 7225c3159d..b85a70d7b2 100644 --- a/libavcodec/adxenc.c +++ b/libavcodec/adxenc.c @@ -34,7 +34,7 @@ /* 18 bytes <-> 32 samples */ -static void adx_encode(unsigned char *adx,const short *wav, +static void adx_encode(ADXContext *c, unsigned char *adx, const short *wav, ADXChannelState *prev) { int scale; @@ -48,7 +48,7 @@ static void adx_encode(unsigned char *adx,const short *wav, s2 = prev->s2; for(i=0;i<32;i++) { s0 = wav[i]; - d = ((s0 << COEFF_BITS) - COEFF1 * s1 + COEFF2 * s2) >> COEFF_BITS; + d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS; data[i]=d; if (max<d) max=d; if (min>d) min=d; @@ -102,19 +102,24 @@ static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t buf } adxhdr; /* big endian */ /* offset-6 "(c)CRI" */ #endif + ADXContext *c = avctx->priv_data; + AV_WB32(buf+0x00,0x80000000|0x20); AV_WB32(buf+0x04,0x03120400|avctx->channels); AV_WB32(buf+0x08,avctx->sample_rate); AV_WB32(buf+0x0c,0); /* FIXME: set after */ - AV_WB32(buf+0x10,0x01040300); - AV_WB32(buf+0x14,0x00000000); - AV_WB32(buf+0x18,0x00000000); - memcpy(buf+0x1c,"\0\0(c)CRI",8); + AV_WB16(buf + 0x10, c->cutoff); + AV_WB32(buf + 0x12, 0x03000000); + AV_WB32(buf + 0x16, 0x00000000); + AV_WB32(buf + 0x1a, 0x00000000); + memcpy (buf + 0x1e, "(c)CRI", 6); return 0x20+4; } static av_cold int adx_encode_init(AVCodecContext *avctx) { + ADXContext *c = avctx->priv_data; + if (avctx->channels > 2) return -1; /* only stereo or mono =) */ avctx->frame_size = 32; @@ -124,6 +129,10 @@ static av_cold int adx_encode_init(AVCodecContext *avctx) // avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32; + /* the cutoff can be adjusted, but this seems to work pretty well */ + c->cutoff = 500; + ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff); + av_log(avctx, AV_LOG_DEBUG, "adx encode init\n"); return 0; @@ -159,7 +168,7 @@ static int adx_encode_frame(AVCodecContext *avctx, if (avctx->channels==1) { while(rest>=32) { - adx_encode(dst,samples,c->prev); + adx_encode(c, dst, samples, c->prev); dst+=18; samples+=32; rest-=32; @@ -174,8 +183,8 @@ static int adx_encode_frame(AVCodecContext *avctx, tmpbuf[i+32] = samples[i*2+1]; } - adx_encode(dst,tmpbuf,c->prev); - adx_encode(dst+18,tmpbuf+32,c->prev+1); + adx_encode(c, dst, tmpbuf, c->prev); + adx_encode(c, dst + 18, tmpbuf + 32, c->prev + 1); dst+=18*2; samples+=32*2; rest-=32*2; |