From 0187178e07218772553767da0cef12e0c0b149a6 Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Wed, 21 Jun 2006 00:21:26 +0000 Subject: [PATCH] fix 24bit flac support, revised from Thibaut Mattern Originally committed as revision 5507 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/flac.c | 53 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 39 insertions(+), 14 deletions(-) diff --git a/libavcodec/flac.c b/libavcodec/flac.c index 8710e21d3d..8bf00b2d04 100644 --- a/libavcodec/flac.c +++ b/libavcodec/flac.c @@ -296,7 +296,7 @@ static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order) static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) { - int sum, i, j; + int i, j; int coeff_prec, qlevel; int coeffs[pred_order]; @@ -334,12 +334,24 @@ static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) if (decode_residuals(s, channel, pred_order) < 0) return -1; - for (i = pred_order; i < s->blocksize; i++) - { - sum = 0; - for (j = 0; j < pred_order; j++) - sum += coeffs[j] * s->decoded[channel][i-j-1]; - s->decoded[channel][i] += sum >> qlevel; + if (s->bps > 16) { + int64_t sum; + for (i = pred_order; i < s->blocksize; i++) + { + sum = 0; + for (j = 0; j < pred_order; j++) + sum += (int64_t)coeffs[j] * s->decoded[channel][i-j-1]; + s->decoded[channel][i] += sum >> qlevel; + } + } else { + int sum; + for (i = pred_order; i < s->blocksize; i++) + { + sum = 0; + for (j = 0; j < pred_order; j++) + sum += coeffs[j] * s->decoded[channel][i-j-1]; + s->decoded[channel][i] += sum >> qlevel; + } } return 0; @@ -538,6 +550,17 @@ static int decode_frame(FLACContext *s) return 0; } +static inline int16_t shift_to_16_bits(int32_t data, int bps) +{ + if (bps == 24) { + return (data >> 8); + } else if (bps == 20) { + return (data >> 4); + } else { + return data; + } +} + static int flac_decode_frame(AVCodecContext *avctx, void *data, int *data_size, uint8_t *buf, int buf_size) @@ -680,23 +703,25 @@ static int flac_decode_frame(AVCodecContext *avctx, for (j = 0; j < s->blocksize; j++) { for (i = 0; i < s->channels; i++) - *(samples++) = s->decoded[i][j]; + *(samples++) = shift_to_16_bits(s->decoded[i][j], s->bps); } break; case LEFT_SIDE: assert(s->channels == 2); for (i = 0; i < s->blocksize; i++) { - *(samples++) = s->decoded[0][i]; - *(samples++) = s->decoded[0][i] - s->decoded[1][i]; + *(samples++) = shift_to_16_bits(s->decoded[0][i], s->bps); + *(samples++) = shift_to_16_bits(s->decoded[0][i] + - s->decoded[1][i], s->bps); } break; case RIGHT_SIDE: assert(s->channels == 2); for (i = 0; i < s->blocksize; i++) { - *(samples++) = s->decoded[0][i] + s->decoded[1][i]; - *(samples++) = s->decoded[1][i]; + *(samples++) = shift_to_16_bits(s->decoded[0][i] + + s->decoded[1][i], s->bps); + *(samples++) = shift_to_16_bits(s->decoded[1][i], s->bps); } break; case MID_SIDE: @@ -709,8 +734,8 @@ static int flac_decode_frame(AVCodecContext *avctx, #if 1 //needs to be checked but IMHO it should be binary identical mid -= side>>1; - *(samples++) = mid + side; - *(samples++) = mid; + *(samples++) = shift_to_16_bits(mid + side, s->bps); + *(samples++) = shift_to_16_bits(mid, s->bps); #else mid <<= 1; -- GitLab