Mercurial > libavcodec.hg
view mpc.c @ 10273:38147f0f94cc libavcodec
Specify maximum sample rate of MLP by defining the factor relative to 48000
instead of directly.
This makes clear that the code assumes the maximum sample rate to be
a multiple of 48000 and also removes the division from the MAX_BLOCKSIZE
macros, which causes an issue with the Solaris assembler where "/" is
a comment marker unless the --divide option is used.
author | reimar |
---|---|
date | Sat, 26 Sep 2009 16:04:35 +0000 |
parents | 0dce4fe6e6f3 |
children | 899237b1961f |
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/* * Musepack decoder core * Copyright (c) 2006 Konstantin Shishkov * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file libavcodec/mpc.c Musepack decoder core * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples * divided into 32 subbands. */ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" #include "mpegaudio.h" #include "mpc.h" #include "mpcdata.h" static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); void ff_mpc_init(void) { ff_mpa_synth_init(mpa_window); } /** * Process decoded Musepack data and produce PCM */ static void mpc_synth(MPCContext *c, int16_t *out) { int dither_state = 0; int i, ch; OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr; for(ch = 0; ch < 2; ch++){ samples_ptr = samples + ch; for(i = 0; i < SAMPLES_PER_BAND; i++) { ff_mpa_synth_filter(c->synth_buf[ch], &(c->synth_buf_offset[ch]), mpa_window, &dither_state, samples_ptr, 2, c->sb_samples[ch][i]); samples_ptr += 64; } } for(i = 0; i < MPC_FRAME_SIZE*2; i++) *out++=samples[i]; } void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data) { int i, j, ch; Band *bands = c->bands; int off; float mul; /* dequantize */ memset(c->sb_samples, 0, sizeof(c->sb_samples)); off = 0; for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){ for(ch = 0; ch < 2; ch++){ if(bands[i].res[ch]){ j = 0; mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0]]; for(; j < 12; j++) c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off]; mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1]]; for(; j < 24; j++) c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off]; mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2]]; for(; j < 36; j++) c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off]; } } if(bands[i].msf){ int t1, t2; for(j = 0; j < SAMPLES_PER_BAND; j++){ t1 = c->sb_samples[0][j][i]; t2 = c->sb_samples[1][j][i]; c->sb_samples[0][j][i] = t1 + t2; c->sb_samples[1][j][i] = t1 - t2; } } } mpc_synth(c, data); }