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author | Fabrice Bellard <fabrice@bellard.org> | 2003-01-27 09:21:30 +0000 |
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committer | Fabrice Bellard <fabrice@bellard.org> | 2003-01-27 09:21:30 +0000 |
commit | 425ed6e223b8dd190e9fbeb4a41c6479f3fc8df0 (patch) | |
tree | 57fbdc23c600efc2cc0f99df119b9a2752b9ac5a /libavcodec/dv.c | |
parent | 850742d785cd566204b112d77c018e9f2b701239 (diff) | |
download | ffmpeg-streaming-425ed6e223b8dd190e9fbeb4a41c6479f3fc8df0.zip ffmpeg-streaming-425ed6e223b8dd190e9fbeb4a41c6479f3fc8df0.tar.gz |
DV audio decoder by Roman Shaposhnick
Originally committed as revision 1514 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/dv.c')
-rw-r--r-- | libavcodec/dv.c | 118 |
1 files changed, 115 insertions, 3 deletions
diff --git a/libavcodec/dv.c b/libavcodec/dv.c index f312c02..554b593 100644 --- a/libavcodec/dv.c +++ b/libavcodec/dv.c @@ -634,7 +634,6 @@ AVCodec dvvideo_decoder = { typedef struct DVAudioDecodeContext { AVCodecContext *avctx; GetBitContext gb; - } DVAudioDecodeContext; static int dvaudio_decode_init(AVCodecContext *avctx) @@ -643,13 +642,126 @@ static int dvaudio_decode_init(AVCodecContext *avctx) return 0; } +static UINT16 dv_audio_12to16(UINT16 sample) +{ + UINT16 shift, result; + + sample = (sample < 0x800) ? sample : sample | 0xf000; + shift = (sample & 0xf00) >> 8; + + if (shift < 0x2 || shift > 0xd) { + result = sample; + } else if (shift < 0x8) { + shift--; + result = (sample - (256 * shift)) << shift; + } else { + shift = 0xe - shift; + result = ((sample + ((256 * shift) + 1)) << shift) - 1; + } + + return result; +} + /* NOTE: exactly one frame must be given (120000 bytes for NTSC, - 144000 bytes for PAL) */ + 144000 bytes for PAL) + + There's a couple of assumptions being made here: + 1. We don't do any kind of audio error correction. It means, + that erroneous samples 0x8000 are being passed upwards. + Do we need to silence erroneous samples ? Average them ? + 2. We don't do software emphasis. + 3. We are not checking for 'speed' argument being valid. + 4. Audio is always returned as 16bit linear samples: 12bit + nonlinear samples are converted into 16bit linear ones. +*/ static int dvaudio_decode_frame(AVCodecContext *avctx, void *data, int *data_size, UINT8 *buf, int buf_size) { - // DVAudioDecodeContext *s = avctx->priv_data; + DVVideoDecodeContext *s = avctx->priv_data; + const UINT16 (*unshuffle)[9]; + int smpls, freq, quant, sys, stride, difseg, ad, dp, nb_dif_segs, i; + UINT16 lc, rc; + UINT8 *buf_ptr; + + /* parse id */ + init_get_bits(&s->gb, &buf[AAUX_OFFSET], 5*8); + i = get_bits(&s->gb, 8); + if (i != 0x50) { /* No audio ? */ + *data_size = 0; + return buf_size; + } + + get_bits(&s->gb, 1); /* 0 - locked audio, 1 - unlocked audio */ + skip_bits(&s->gb, 1); + smpls = get_bits(&s->gb, 6); /* samples in this frame - min. samples */ + + skip_bits(&s->gb, 8); + + skip_bits(&s->gb, 2); + sys = get_bits(&s->gb, 1); /* 0 - 60 fields, 1 = 50 fields */ + skip_bits(&s->gb, 5); + + get_bits(&s->gb, 1); /* 0 - emphasis on, 1 - emphasis off */ + get_bits(&s->gb, 1); /* 0 - reserved, 1 - emphasis time constant 50/15us */ + freq = get_bits(&s->gb, 3); /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */ + quant = get_bits(&s->gb, 3); /* 0 - 16bit linear, 1 - 12bit nonlinear */ + + if (quant > 1) + return -1; /* Unsupported quantization */ + + avctx->sample_rate = dv_audio_frequency[freq]; + // What about: + // avctx->bit_rate = + // avctx->frame_size = + + *data_size = (dv_audio_min_samples[sys][freq] + smpls) * + avctx->channels * 2; + + if (sys) { + nb_dif_segs = 12; + stride = 108; + unshuffle = dv_place_audio50; + } else { + nb_dif_segs = 10; + stride = 90; + unshuffle = dv_place_audio60; + } + + /* for each DIF segment */ + buf_ptr = buf; + for (difseg = 0; difseg < nb_dif_segs; difseg++) { + buf_ptr += 6 * 80; /* skip DIF segment header */ + for (ad = 0; ad < 9; ad++) { + + for (dp = 8; dp < 80; dp+=2) { + if (quant == 0) { /* 16bit quantization */ + i = unshuffle[difseg][ad] + (dp - 8)/2 * stride; + ((short *)data)[i] = (buf_ptr[dp] << 8) | buf_ptr[dp+1]; + } else { /* 12bit quantization */ + if (difseg >= nb_dif_segs/2) + goto out; /* We're not doing 4ch at this time */ + + lc = ((UINT16)buf_ptr[dp] << 4) | + ((UINT16)buf_ptr[dp+2] >> 4); + rc = ((UINT16)buf_ptr[dp+1] << 4) | + ((UINT16)buf_ptr[dp+2] & 0x0f); + lc = dv_audio_12to16(lc); + rc = dv_audio_12to16(rc); + + i = unshuffle[difseg][ad] + (dp - 8)/3 * stride; + ((short *)data)[i] = lc; + i = unshuffle[difseg+nb_dif_segs/2][ad] + (dp - 8)/3 * stride; + ((short *)data)[i] = rc; + ++dp; + } + } + + buf_ptr += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */ + } + } + +out: return buf_size; } |