/*
- * AC3 decoder
- * Copyright (c) 2001 Gerard Lantau.
+ * AC-3 Audio Decoder
+ * This code is developed as part of Google Summer of Code 2006 Program.
*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com).
+ * Copyright (c) 2007 Justin Ruggles
*
- * This program is distributed in the hope that it will be useful,
+ * Portions of this code are derived from liba52
+ * http://liba52.sourceforge.net
+ * Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
+ * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 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 General Public License for more details.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
*
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ * You should have received a copy of the GNU 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
*/
-#include <stdlib.h>
+
#include <stdio.h>
+#include <stddef.h>
+#include <math.h>
#include <string.h>
+
#include "avcodec.h"
+#include "ac3_parser.h"
+#include "bitstream.h"
+#include "crc.h"
+#include "dsputil.h"
+#include "random.h"
+
+/**
+ * Table of bin locations for rematrixing bands
+ * reference: Section 7.5.2 Rematrixing : Frequency Band Definitions
+ */
+static const uint8_t rematrix_band_tab[5] = { 13, 25, 37, 61, 253 };
+
+/** table for grouping exponents */
+static uint8_t exp_ungroup_tab[128][3];
-#include <inttypes.h>
-#include "libac3/ac3.h"
-/* currently, I use libac3 which is Copyright (C) Aaron Holtzman and
- released under the GPL license. I may reimplement it someday... */
-typedef struct AC3DecodeState {
- UINT8 inbuf[4096]; /* input buffer */
- UINT8 *inbuf_ptr;
- int frame_size;
- int flags;
- ac3_state_t state;
-} AC3DecodeState;
+/** tables for ungrouping mantissas */
+static int b1_mantissas[32][3];
+static int b2_mantissas[128][3];
+static int b3_mantissas[8];
+static int b4_mantissas[128][2];
+static int b5_mantissas[16];
+
+/**
+ * Quantization table: levels for symmetric. bits for asymmetric.
+ * reference: Table 7.18 Mapping of bap to Quantizer
+ */
+static const uint8_t quantization_tab[16] = {
+ 0, 3, 5, 7, 11, 15,
+ 5, 6, 7, 8, 9, 10, 11, 12, 14, 16
+};
+
+/** dynamic range table. converts codes to scale factors. */
+static float dynamic_range_tab[256];
+
+/** Adjustments in dB gain */
+#define LEVEL_MINUS_3DB 0.7071067811865476
+#define LEVEL_MINUS_4POINT5DB 0.5946035575013605
+#define LEVEL_MINUS_6DB 0.5000000000000000
+#define LEVEL_MINUS_9DB 0.3535533905932738
+#define LEVEL_ZERO 0.0000000000000000
+#define LEVEL_ONE 1.0000000000000000
+
+static const float gain_levels[6] = {
+ LEVEL_ZERO,
+ LEVEL_ONE,
+ LEVEL_MINUS_3DB,
+ LEVEL_MINUS_4POINT5DB,
+ LEVEL_MINUS_6DB,
+ LEVEL_MINUS_9DB
+};
+
+/**
+ * Table for center mix levels
+ * reference: Section 5.4.2.4 cmixlev
+ */
+static const uint8_t center_levels[4] = { 2, 3, 4, 3 };
+
+/**
+ * Table for surround mix levels
+ * reference: Section 5.4.2.5 surmixlev
+ */
+static const uint8_t surround_levels[4] = { 2, 4, 0, 4 };
-static int ac3_decode_init(AVCodecContext *avctx)
+/**
+ * Table for default stereo downmixing coefficients
+ * reference: Section 7.8.2 Downmixing Into Two Channels
+ */
+static const uint8_t ac3_default_coeffs[8][5][2] = {
+ { { 1, 0 }, { 0, 1 }, },
+ { { 2, 2 }, },
+ { { 1, 0 }, { 0, 1 }, },
+ { { 1, 0 }, { 3, 3 }, { 0, 1 }, },
+ { { 1, 0 }, { 0, 1 }, { 4, 4 }, },
+ { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 5, 5 }, },
+ { { 1, 0 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
+ { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
+};
+
+/* override ac3.h to include coupling channel */
+#undef AC3_MAX_CHANNELS
+#define AC3_MAX_CHANNELS 7
+#define CPL_CH 0
+
+#define AC3_OUTPUT_LFEON 8
+
+typedef struct {
+ int channel_mode; ///< channel mode (acmod)
+ int block_switch[AC3_MAX_CHANNELS]; ///< block switch flags
+ int dither_flag[AC3_MAX_CHANNELS]; ///< dither flags
+ int dither_all; ///< true if all channels are dithered
+ int cpl_in_use; ///< coupling in use
+ int channel_in_cpl[AC3_MAX_CHANNELS]; ///< channel in coupling
+ int phase_flags_in_use; ///< phase flags in use
+ int phase_flags[18]; ///< phase flags
+ int cpl_band_struct[18]; ///< coupling band structure
+ int num_rematrixing_bands; ///< number of rematrixing bands
+ int rematrixing_flags[4]; ///< rematrixing flags
+ int exp_strategy[AC3_MAX_CHANNELS]; ///< exponent strategies
+ int snr_offset[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets
+ int fast_gain[AC3_MAX_CHANNELS]; ///< fast gain values (signal-to-mask ratio)
+ int dba_mode[AC3_MAX_CHANNELS]; ///< delta bit allocation mode
+ int dba_nsegs[AC3_MAX_CHANNELS]; ///< number of delta segments
+ uint8_t dba_offsets[AC3_MAX_CHANNELS][8]; ///< delta segment offsets
+ uint8_t dba_lengths[AC3_MAX_CHANNELS][8]; ///< delta segment lengths
+ uint8_t dba_values[AC3_MAX_CHANNELS][8]; ///< delta values for each segment
+
+ int sample_rate; ///< sample frequency, in Hz
+ int bit_rate; ///< stream bit rate, in bits-per-second
+ int frame_size; ///< current frame size, in bytes
+
+ int channels; ///< number of total channels
+ int fbw_channels; ///< number of full-bandwidth channels
+ int lfe_on; ///< lfe channel in use
+ int lfe_ch; ///< index of LFE channel
+ int output_mode; ///< output channel configuration
+ int out_channels; ///< number of output channels
+
+ int center_mix_level; ///< Center mix level index
+ int surround_mix_level; ///< Surround mix level index
+ float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
+ float downmix_coeff_adjust[2]; ///< adjustment needed for each output channel when downmixing
+ float dynamic_range[2]; ///< dynamic range
+ int cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates
+ int num_cpl_bands; ///< number of coupling bands
+ int num_cpl_subbands; ///< number of coupling sub bands
+ int start_freq[AC3_MAX_CHANNELS]; ///< start frequency bin
+ int end_freq[AC3_MAX_CHANNELS]; ///< end frequency bin
+ AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters
+
+ int8_t dexps[AC3_MAX_CHANNELS][256]; ///< decoded exponents
+ uint8_t bap[AC3_MAX_CHANNELS][256]; ///< bit allocation pointers
+ int16_t psd[AC3_MAX_CHANNELS][256]; ///< scaled exponents
+ int16_t band_psd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents
+ int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values
+
+ int fixed_coeffs[AC3_MAX_CHANNELS][256]; ///> fixed-point transform coefficients
+ DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients
+ int downmixed; ///< indicates if coeffs are currently downmixed
+
+ /* For IMDCT. */
+ MDCTContext imdct_512; ///< for 512 sample IMDCT
+ MDCTContext imdct_256; ///< for 256 sample IMDCT
+ DSPContext dsp; ///< for optimization
+ float add_bias; ///< offset for float_to_int16 conversion
+ float mul_bias; ///< scaling for float_to_int16 conversion
+
+ DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][256]); ///< output after imdct transform and windowing
+ DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output
+ DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][256]); ///< delay - added to the next block
+ DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform
+ DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing
+ DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients
+
+ /* Miscellaneous. */
+ GetBitContext gbc; ///< bitstream reader
+ AVRandomState dith_state; ///< for dither generation
+ AVCodecContext *avctx; ///< parent context
+} AC3DecodeContext;
+
+/**
+ * Symmetrical Dequantization
+ * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
+ * Tables 7.19 to 7.23
+ */
+static inline int
+symmetric_dequant(int code, int levels)
{
- AC3DecodeState *s = avctx->priv_data;
+ return ((code - (levels >> 1)) << 24) / levels;
+}
- ac3_init ();
- s->inbuf_ptr = s->inbuf;
- s->frame_size = 0;
- return 0;
+/*
+ * Initialize tables at runtime.
+ */
+static av_cold void ac3_tables_init(void)
+{
+ int i;
+
+ /* generate grouped mantissa tables
+ reference: Section 7.3.5 Ungrouping of Mantissas */
+ for(i=0; i<32; i++) {
+ /* bap=1 mantissas */
+ b1_mantissas[i][0] = symmetric_dequant( i / 9 , 3);
+ b1_mantissas[i][1] = symmetric_dequant((i % 9) / 3, 3);
+ b1_mantissas[i][2] = symmetric_dequant((i % 9) % 3, 3);
+ }
+ for(i=0; i<128; i++) {
+ /* bap=2 mantissas */
+ b2_mantissas[i][0] = symmetric_dequant( i / 25 , 5);
+ b2_mantissas[i][1] = symmetric_dequant((i % 25) / 5, 5);
+ b2_mantissas[i][2] = symmetric_dequant((i % 25) % 5, 5);
+
+ /* bap=4 mantissas */
+ b4_mantissas[i][0] = symmetric_dequant(i / 11, 11);
+ b4_mantissas[i][1] = symmetric_dequant(i % 11, 11);
+ }
+ /* generate ungrouped mantissa tables
+ reference: Tables 7.21 and 7.23 */
+ for(i=0; i<7; i++) {
+ /* bap=3 mantissas */
+ b3_mantissas[i] = symmetric_dequant(i, 7);
+ }
+ for(i=0; i<15; i++) {
+ /* bap=5 mantissas */
+ b5_mantissas[i] = symmetric_dequant(i, 15);
+ }
+
+ /* generate dynamic range table
+ reference: Section 7.7.1 Dynamic Range Control */
+ for(i=0; i<256; i++) {
+ int v = (i >> 5) - ((i >> 7) << 3) - 5;
+ dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
+ }
+
+ /* generate exponent tables
+ reference: Section 7.1.3 Exponent Decoding */
+ for(i=0; i<128; i++) {
+ exp_ungroup_tab[i][0] = i / 25;
+ exp_ungroup_tab[i][1] = (i % 25) / 5;
+ exp_ungroup_tab[i][2] = (i % 25) % 5;
+ }
}
-stream_samples_t samples;
-/**** the following two functions comes from ac3dec */
-static inline int blah (int32_t i)
+/**
+ * AVCodec initialization
+ */
+static av_cold int ac3_decode_init(AVCodecContext *avctx)
{
- if (i > 0x43c07fff)
- return 32767;
- else if (i < 0x43bf8000)
- return -32768;
- else
- return i - 0x43c00000;
+ AC3DecodeContext *s = avctx->priv_data;
+ s->avctx = avctx;
+
+ ac3_common_init();
+ ac3_tables_init();
+ ff_mdct_init(&s->imdct_256, 8, 1);
+ ff_mdct_init(&s->imdct_512, 9, 1);
+ ff_kbd_window_init(s->window, 5.0, 256);
+ dsputil_init(&s->dsp, avctx);
+ av_init_random(0, &s->dith_state);
+
+ /* set bias values for float to int16 conversion */
+ if(s->dsp.float_to_int16 == ff_float_to_int16_c) {
+ s->add_bias = 385.0f;
+ s->mul_bias = 1.0f;
+ } else {
+ s->add_bias = 0.0f;
+ s->mul_bias = 32767.0f;
+ }
+
+ /* allow downmixing to stereo or mono */
+ if (avctx->channels > 0 && avctx->request_channels > 0 &&
+ avctx->request_channels < avctx->channels &&
+ avctx->request_channels <= 2) {
+ avctx->channels = avctx->request_channels;
+ }
+ s->downmixed = 1;
+
+ return 0;
}
-static inline void float_to_int (float * _f, INT16 * s16)
+/**
+ * Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
+ * GetBitContext within AC3DecodeContext must point to
+ * start of the synchronized ac3 bitstream.
+ */
+static int ac3_parse_header(AC3DecodeContext *s)
{
- int i;
- int32_t * f = (int32_t *) _f; // XXX assumes IEEE float format
+ AC3HeaderInfo hdr;
+ GetBitContext *gbc = &s->gbc;
+ int err, i;
+
+ err = ff_ac3_parse_header(gbc->buffer, &hdr);
+ if(err)
+ return err;
+
+ if(hdr.bitstream_id > 10)
+ return AC3_PARSE_ERROR_BSID;
+
+ /* get decoding parameters from header info */
+ s->bit_alloc_params.sr_code = hdr.sr_code;
+ s->channel_mode = hdr.channel_mode;
+ s->lfe_on = hdr.lfe_on;
+ s->bit_alloc_params.sr_shift = hdr.sr_shift;
+ s->sample_rate = hdr.sample_rate;
+ s->bit_rate = hdr.bit_rate;
+ s->channels = hdr.channels;
+ s->fbw_channels = s->channels - s->lfe_on;
+ s->lfe_ch = s->fbw_channels + 1;
+ s->frame_size = hdr.frame_size;
+
+ /* set default output to all source channels */
+ s->out_channels = s->channels;
+ s->output_mode = s->channel_mode;
+ if(s->lfe_on)
+ s->output_mode |= AC3_OUTPUT_LFEON;
+
+ /* set default mix levels */
+ s->center_mix_level = 3; // -4.5dB
+ s->surround_mix_level = 4; // -6.0dB
- for (i = 0; i < 256; i++) {
- s16[2*i] = blah (f[i]);
- s16[2*i+1] = blah (f[i+256]);
+ /* skip over portion of header which has already been read */
+ skip_bits(gbc, 16); // skip the sync_word
+ skip_bits(gbc, 16); // skip crc1
+ skip_bits(gbc, 8); // skip fscod and frmsizecod
+ skip_bits(gbc, 11); // skip bsid, bsmod, and acmod
+ if(s->channel_mode == AC3_CHMODE_STEREO) {
+ skip_bits(gbc, 2); // skip dsurmod
+ } else {
+ if((s->channel_mode & 1) && s->channel_mode != AC3_CHMODE_MONO)
+ s->center_mix_level = center_levels[get_bits(gbc, 2)];
+ if(s->channel_mode & 4)
+ s->surround_mix_level = surround_levels[get_bits(gbc, 2)];
}
+ skip_bits1(gbc); // skip lfeon
+
+ /* read the rest of the bsi. read twice for dual mono mode. */
+ i = !(s->channel_mode);
+ do {
+ skip_bits(gbc, 5); // skip dialog normalization
+ if (get_bits1(gbc))
+ skip_bits(gbc, 8); //skip compression
+ if (get_bits1(gbc))
+ skip_bits(gbc, 8); //skip language code
+ if (get_bits1(gbc))
+ skip_bits(gbc, 7); //skip audio production information
+ } while (i--);
+
+ skip_bits(gbc, 2); //skip copyright bit and original bitstream bit
+
+ /* skip the timecodes (or extra bitstream information for Alternate Syntax)
+ TODO: read & use the xbsi1 downmix levels */
+ if (get_bits1(gbc))
+ skip_bits(gbc, 14); //skip timecode1 / xbsi1
+ if (get_bits1(gbc))
+ skip_bits(gbc, 14); //skip timecode2 / xbsi2
+
+ /* skip additional bitstream info */
+ if (get_bits1(gbc)) {
+ i = get_bits(gbc, 6);
+ do {
+ skip_bits(gbc, 8);
+ } while(i--);
+ }
+
+ return 0;
}
-static inline void float_to_int_mono (float * _f, INT16 * s16)
+/**
+ * Set stereo downmixing coefficients based on frame header info.
+ * reference: Section 7.8.2 Downmixing Into Two Channels
+ */
+static void set_downmix_coeffs(AC3DecodeContext *s)
{
int i;
- int32_t * f = (int32_t *) _f; // XXX assumes IEEE float format
-
- for (i = 0; i < 256; i++) {
- s16[i] = blah (f[i]);
- }
-}
-
-/**** end */
-
-#define HEADER_SIZE 7
-
-static int ac3_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- UINT8 *buf, int buf_size)
-{
- AC3DecodeState *s = avctx->priv_data;
- UINT8 *buf_ptr;
- int flags, i, len;
- int sample_rate, bit_rate;
- short *out_samples = data;
- float level;
-
- *data_size = 0;
- buf_ptr = buf;
- while (buf_size > 0) {
- len = s->inbuf_ptr - s->inbuf;
- if (s->frame_size == 0) {
- /* no header seen : find one. We need at least 7 bytes to parse it */
- len = HEADER_SIZE - len;
- if (len > buf_size)
- len = buf_size;
- memcpy(s->inbuf_ptr, buf_ptr, len);
- buf_ptr += len;
- s->inbuf_ptr += len;
- buf_size -= len;
- if ((s->inbuf_ptr - s->inbuf) == HEADER_SIZE) {
- len = ac3_syncinfo (s->inbuf, &s->flags, &sample_rate, &bit_rate);
- if (len == 0) {
- /* no sync found : move by one byte (inefficient, but simple!) */
- memcpy(s->inbuf, s->inbuf + 1, HEADER_SIZE - 1);
- s->inbuf_ptr--;
- } else {
- s->frame_size = len;
- /* update codec info */
- avctx->sample_rate = sample_rate;
- if ((s->flags & AC3_CHANNEL_MASK) == AC3_MONO)
- avctx->channels = 1;
- else
- avctx->channels = 2;
- avctx->bit_rate = bit_rate;
+ float cmix = gain_levels[s->center_mix_level];
+ float smix = gain_levels[s->surround_mix_level];
+
+ for(i=0; i<s->fbw_channels; i++) {
+ s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
+ s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
+ }
+ if(s->channel_mode > 1 && s->channel_mode & 1) {
+ s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix;
+ }
+ if(s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) {
+ int nf = s->channel_mode - 2;
+ s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
+ }
+ if(s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) {
+ int nf = s->channel_mode - 4;
+ s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix;
+ }
+
+ /* calculate adjustment needed for each channel to avoid clipping */
+ s->downmix_coeff_adjust[0] = s->downmix_coeff_adjust[1] = 0.0f;
+ for(i=0; i<s->fbw_channels; i++) {
+ s->downmix_coeff_adjust[0] += s->downmix_coeffs[i][0];
+ s->downmix_coeff_adjust[1] += s->downmix_coeffs[i][1];
+ }
+ s->downmix_coeff_adjust[0] = 1.0f / s->downmix_coeff_adjust[0];
+ s->downmix_coeff_adjust[1] = 1.0f / s->downmix_coeff_adjust[1];
+}
+
+/**
+ * Decode the grouped exponents according to exponent strategy.
+ * reference: Section 7.1.3 Exponent Decoding
+ */
+static void decode_exponents(GetBitContext *gbc, int exp_strategy, int ngrps,
+ uint8_t absexp, int8_t *dexps)
+{
+ int i, j, grp, group_size;
+ int dexp[256];
+ int expacc, prevexp;
+
+ /* unpack groups */
+ group_size = exp_strategy + (exp_strategy == EXP_D45);
+ for(grp=0,i=0; grp<ngrps; grp++) {
+ expacc = get_bits(gbc, 7);
+ dexp[i++] = exp_ungroup_tab[expacc][0];
+ dexp[i++] = exp_ungroup_tab[expacc][1];
+ dexp[i++] = exp_ungroup_tab[expacc][2];
+ }
+
+ /* convert to absolute exps and expand groups */
+ prevexp = absexp;
+ for(i=0; i<ngrps*3; i++) {
+ prevexp = av_clip(prevexp + dexp[i]-2, 0, 24);
+ for(j=0; j<group_size; j++) {
+ dexps[(i*group_size)+j] = prevexp;
+ }
+ }
+}
+
+/**
+ * Generate transform coefficients for each coupled channel in the coupling
+ * range using the coupling coefficients and coupling coordinates.
+ * reference: Section 7.4.3 Coupling Coordinate Format
+ */
+static void uncouple_channels(AC3DecodeContext *s)
+{
+ int i, j, ch, bnd, subbnd;
+
+ subbnd = -1;
+ i = s->start_freq[CPL_CH];
+ for(bnd=0; bnd<s->num_cpl_bands; bnd++) {
+ do {
+ subbnd++;
+ for(j=0; j<12; j++) {
+ for(ch=1; ch<=s->fbw_channels; ch++) {
+ if(s->channel_in_cpl[ch]) {
+ s->fixed_coeffs[ch][i] = ((int64_t)s->fixed_coeffs[CPL_CH][i] * (int64_t)s->cpl_coords[ch][bnd]) >> 23;
+ if (ch == 2 && s->phase_flags[bnd])
+ s->fixed_coeffs[ch][i] = -s->fixed_coeffs[ch][i];
+ }
}
+ i++;
}
- } else if (len < s->frame_size) {
- len = s->frame_size - len;
- if (len > buf_size)
- len = buf_size;
-
- memcpy(s->inbuf_ptr, buf_ptr, len);
- buf_ptr += len;
- s->inbuf_ptr += len;
- buf_size -= len;
- } else {
- if (avctx->channels == 1)
- flags = AC3_MONO;
+ } while(s->cpl_band_struct[subbnd]);
+ }
+}
+
+/**
+ * Grouped mantissas for 3-level 5-level and 11-level quantization
+ */
+typedef struct {
+ int b1_mant[3];
+ int b2_mant[3];
+ int b4_mant[2];
+ int b1ptr;
+ int b2ptr;
+ int b4ptr;
+} mant_groups;
+
+/**
+ * Get the transform coefficients for a particular channel
+ * reference: Section 7.3 Quantization and Decoding of Mantissas
+ */
+static int get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m)
+{
+ GetBitContext *gbc = &s->gbc;
+ int i, gcode, tbap, start, end;
+ uint8_t *exps;
+ uint8_t *bap;
+ int *coeffs;
+
+ exps = s->dexps[ch_index];
+ bap = s->bap[ch_index];
+ coeffs = s->fixed_coeffs[ch_index];
+ start = s->start_freq[ch_index];
+ end = s->end_freq[ch_index];
+
+ for (i = start; i < end; i++) {
+ tbap = bap[i];
+ switch (tbap) {
+ case 0:
+ coeffs[i] = (av_random(&s->dith_state) & 0x7FFFFF) - 4194304;
+ break;
+
+ case 1:
+ if(m->b1ptr > 2) {
+ gcode = get_bits(gbc, 5);
+ m->b1_mant[0] = b1_mantissas[gcode][0];
+ m->b1_mant[1] = b1_mantissas[gcode][1];
+ m->b1_mant[2] = b1_mantissas[gcode][2];
+ m->b1ptr = 0;
+ }
+ coeffs[i] = m->b1_mant[m->b1ptr++];
+ break;
+
+ case 2:
+ if(m->b2ptr > 2) {
+ gcode = get_bits(gbc, 7);
+ m->b2_mant[0] = b2_mantissas[gcode][0];
+ m->b2_mant[1] = b2_mantissas[gcode][1];
+ m->b2_mant[2] = b2_mantissas[gcode][2];
+ m->b2ptr = 0;
+ }
+ coeffs[i] = m->b2_mant[m->b2ptr++];
+ break;
+
+ case 3:
+ coeffs[i] = b3_mantissas[get_bits(gbc, 3)];
+ break;
+
+ case 4:
+ if(m->b4ptr > 1) {
+ gcode = get_bits(gbc, 7);
+ m->b4_mant[0] = b4_mantissas[gcode][0];
+ m->b4_mant[1] = b4_mantissas[gcode][1];
+ m->b4ptr = 0;
+ }
+ coeffs[i] = m->b4_mant[m->b4ptr++];
+ break;
+
+ case 5:
+ coeffs[i] = b5_mantissas[get_bits(gbc, 4)];
+ break;
+
+ default: {
+ /* asymmetric dequantization */
+ int qlevel = quantization_tab[tbap];
+ coeffs[i] = get_sbits(gbc, qlevel) << (24 - qlevel);
+ break;
+ }
+ }
+ coeffs[i] >>= exps[i];
+ }
+
+ return 0;
+}
+
+/**
+ * Remove random dithering from coefficients with zero-bit mantissas
+ * reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0)
+ */
+static void remove_dithering(AC3DecodeContext *s) {
+ int ch, i;
+ int end=0;
+ int *coeffs;
+ uint8_t *bap;
+
+ for(ch=1; ch<=s->fbw_channels; ch++) {
+ if(!s->dither_flag[ch]) {
+ coeffs = s->fixed_coeffs[ch];
+ bap = s->bap[ch];
+ if(s->channel_in_cpl[ch])
+ end = s->start_freq[CPL_CH];
else
- flags = AC3_STEREO;
-
- flags |= AC3_ADJUST_LEVEL;
- level = 1;
- if (ac3_frame (&s->state, s->inbuf, &flags, &level, 384)) {
- fail:
- s->inbuf_ptr = s->inbuf;
- s->frame_size = 0;
- continue;
+ end = s->end_freq[ch];
+ for(i=0; i<end; i++) {
+ if(!bap[i])
+ coeffs[i] = 0;
+ }
+ if(s->channel_in_cpl[ch]) {
+ bap = s->bap[CPL_CH];
+ for(; i<s->end_freq[CPL_CH]; i++) {
+ if(!bap[i])
+ coeffs[i] = 0;
+ }
+ }
+ }
+ }
+}
+
+/**
+ * Get the transform coefficients.
+ */
+static int get_transform_coeffs(AC3DecodeContext *s)
+{
+ int ch, end;
+ int got_cplchan = 0;
+ mant_groups m;
+
+ m.b1ptr = m.b2ptr = m.b4ptr = 3;
+
+ for (ch = 1; ch <= s->channels; ch++) {
+ /* transform coefficients for full-bandwidth channel */
+ if (get_transform_coeffs_ch(s, ch, &m))
+ return -1;
+ /* tranform coefficients for coupling channel come right after the
+ coefficients for the first coupled channel*/
+ if (s->channel_in_cpl[ch]) {
+ if (!got_cplchan) {
+ if (get_transform_coeffs_ch(s, CPL_CH, &m)) {
+ av_log(s->avctx, AV_LOG_ERROR, "error in decoupling channels\n");
+ return -1;
+ }
+ uncouple_channels(s);
+ got_cplchan = 1;
}
- for (i = 0; i < 6; i++) {
- if (ac3_block (&s->state))
- goto fail;
- if (avctx->channels == 1)
- float_to_int_mono (*samples, out_samples + i * 256);
- else
- float_to_int (*samples, out_samples + i * 512);
+ end = s->end_freq[CPL_CH];
+ } else {
+ end = s->end_freq[ch];
+ }
+ do
+ s->transform_coeffs[ch][end] = 0;
+ while(++end < 256);
+ }
+
+ /* if any channel doesn't use dithering, zero appropriate coefficients */
+ if(!s->dither_all)
+ remove_dithering(s);
+
+ return 0;
+}
+
+/**
+ * Stereo rematrixing.
+ * reference: Section 7.5.4 Rematrixing : Decoding Technique
+ */
+static void do_rematrixing(AC3DecodeContext *s)
+{
+ int bnd, i;
+ int end, bndend;
+ int tmp0, tmp1;
+
+ end = FFMIN(s->end_freq[1], s->end_freq[2]);
+
+ for(bnd=0; bnd<s->num_rematrixing_bands; bnd++) {
+ if(s->rematrixing_flags[bnd]) {
+ bndend = FFMIN(end, rematrix_band_tab[bnd+1]);
+ for(i=rematrix_band_tab[bnd]; i<bndend; i++) {
+ tmp0 = s->fixed_coeffs[1][i];
+ tmp1 = s->fixed_coeffs[2][i];
+ s->fixed_coeffs[1][i] = tmp0 + tmp1;
+ s->fixed_coeffs[2][i] = tmp0 - tmp1;
}
- s->inbuf_ptr = s->inbuf;
- s->frame_size = 0;
- *data_size = 6 * avctx->channels * 256 * sizeof(INT16);
+ }
+ }
+}
+
+/**
+ * Perform the 256-point IMDCT
+ */
+static void do_imdct_256(AC3DecodeContext *s, int chindex)
+{
+ int i, k;
+ DECLARE_ALIGNED_16(float, x[128]);
+ FFTComplex z[2][64];
+ float *o_ptr = s->tmp_output;
+
+ for(i=0; i<2; i++) {
+ /* de-interleave coefficients */
+ for(k=0; k<128; k++) {
+ x[k] = s->transform_coeffs[chindex][2*k+i];
+ }
+
+ /* run standard IMDCT */
+ s->imdct_256.fft.imdct_calc(&s->imdct_256, o_ptr, x, s->tmp_imdct);
+
+ /* reverse the post-rotation & reordering from standard IMDCT */
+ for(k=0; k<32; k++) {
+ z[i][32+k].re = -o_ptr[128+2*k];
+ z[i][32+k].im = -o_ptr[2*k];
+ z[i][31-k].re = o_ptr[2*k+1];
+ z[i][31-k].im = o_ptr[128+2*k+1];
+ }
+ }
+
+ /* apply AC-3 post-rotation & reordering */
+ for(k=0; k<64; k++) {
+ o_ptr[ 2*k ] = -z[0][ k].im;
+ o_ptr[ 2*k+1] = z[0][63-k].re;
+ o_ptr[128+2*k ] = -z[0][ k].re;
+ o_ptr[128+2*k+1] = z[0][63-k].im;
+ o_ptr[256+2*k ] = -z[1][ k].re;
+ o_ptr[256+2*k+1] = z[1][63-k].im;
+ o_ptr[384+2*k ] = z[1][ k].im;
+ o_ptr[384+2*k+1] = -z[1][63-k].re;
+ }
+}
+
+/**
+ * Inverse MDCT Transform.
+ * Convert frequency domain coefficients to time-domain audio samples.
+ * reference: Section 7.9.4 Transformation Equations
+ */
+static inline void do_imdct(AC3DecodeContext *s, int channels)
+{
+ int ch;
+
+ for (ch=1; ch<=channels; ch++) {
+ if (s->block_switch[ch]) {
+ do_imdct_256(s, ch);
+ } else {
+ s->imdct_512.fft.imdct_calc(&s->imdct_512, s->tmp_output,
+ s->transform_coeffs[ch], s->tmp_imdct);
+ }
+ /* For the first half of the block, apply the window, add the delay
+ from the previous block, and send to output */
+ s->dsp.vector_fmul_add_add(s->output[ch-1], s->tmp_output,
+ s->window, s->delay[ch-1], 0, 256, 1);
+ /* For the second half of the block, apply the window and store the
+ samples to delay, to be combined with the next block */
+ s->dsp.vector_fmul_reverse(s->delay[ch-1], s->tmp_output+256,
+ s->window, 256);
+ }
+}
+
+/**
+ * Downmix the output to mono or stereo.
+ */
+static void ac3_downmix(AC3DecodeContext *s,
+ float samples[AC3_MAX_CHANNELS][256], int ch_offset)
+{
+ int i, j;
+ float v0, v1;
+
+ for(i=0; i<256; i++) {
+ v0 = v1 = 0.0f;
+ for(j=0; j<s->fbw_channels; j++) {
+ v0 += samples[j+ch_offset][i] * s->downmix_coeffs[j][0];
+ v1 += samples[j+ch_offset][i] * s->downmix_coeffs[j][1];
+ }
+ v0 *= s->downmix_coeff_adjust[0];
+ v1 *= s->downmix_coeff_adjust[1];
+ if(s->output_mode == AC3_CHMODE_MONO) {
+ samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB;
+ } else if(s->output_mode == AC3_CHMODE_STEREO) {
+ samples[ ch_offset][i] = v0;
+ samples[1+ch_offset][i] = v1;
+ }
+ }
+}
+
+/**
+ * Upmix delay samples from stereo to original channel layout.
+ */
+static void ac3_upmix_delay(AC3DecodeContext *s)
+{
+ int channel_data_size = sizeof(s->delay[0]);
+ switch(s->channel_mode) {
+ case AC3_CHMODE_DUALMONO:
+ case AC3_CHMODE_STEREO:
+ /* upmix mono to stereo */
+ memcpy(s->delay[1], s->delay[0], channel_data_size);
break;
+ case AC3_CHMODE_2F2R:
+ memset(s->delay[3], 0, channel_data_size);
+ case AC3_CHMODE_2F1R:
+ memset(s->delay[2], 0, channel_data_size);
+ break;
+ case AC3_CHMODE_3F2R:
+ memset(s->delay[4], 0, channel_data_size);
+ case AC3_CHMODE_3F1R:
+ memset(s->delay[3], 0, channel_data_size);
+ case AC3_CHMODE_3F:
+ memcpy(s->delay[2], s->delay[1], channel_data_size);
+ memset(s->delay[1], 0, channel_data_size);
+ break;
+ }
+}
+
+/**
+ * Parse an audio block from AC-3 bitstream.
+ */
+static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
+{
+ int fbw_channels = s->fbw_channels;
+ int channel_mode = s->channel_mode;
+ int i, bnd, seg, ch;
+ int different_transforms;
+ int downmix_output;
+ GetBitContext *gbc = &s->gbc;
+ uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
+
+ memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
+
+ /* block switch flags */
+ different_transforms = 0;
+ for (ch = 1; ch <= fbw_channels; ch++) {
+ s->block_switch[ch] = get_bits1(gbc);
+ if(ch > 1 && s->block_switch[ch] != s->block_switch[1])
+ different_transforms = 1;
+ }
+
+ /* dithering flags */
+ s->dither_all = 1;
+ for (ch = 1; ch <= fbw_channels; ch++) {
+ s->dither_flag[ch] = get_bits1(gbc);
+ if(!s->dither_flag[ch])
+ s->dither_all = 0;
+ }
+
+ /* dynamic range */
+ i = !(s->channel_mode);
+ do {
+ if(get_bits1(gbc)) {
+ s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)]-1.0) *
+ s->avctx->drc_scale)+1.0;
+ } else if(blk == 0) {
+ s->dynamic_range[i] = 1.0f;
+ }
+ } while(i--);
+
+ /* coupling strategy */
+ if (get_bits1(gbc)) {
+ memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
+ s->cpl_in_use = get_bits1(gbc);
+ if (s->cpl_in_use) {
+ /* coupling in use */
+ int cpl_begin_freq, cpl_end_freq;
+
+ /* determine which channels are coupled */
+ for (ch = 1; ch <= fbw_channels; ch++)
+ s->channel_in_cpl[ch] = get_bits1(gbc);
+
+ /* phase flags in use */
+ if (channel_mode == AC3_CHMODE_STEREO)
+ s->phase_flags_in_use = get_bits1(gbc);
+
+ /* coupling frequency range and band structure */
+ cpl_begin_freq = get_bits(gbc, 4);
+ cpl_end_freq = get_bits(gbc, 4);
+ if (3 + cpl_end_freq - cpl_begin_freq < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "3+cplendf = %d < cplbegf = %d\n", 3+cpl_end_freq, cpl_begin_freq);
+ return -1;
+ }
+ s->num_cpl_bands = s->num_cpl_subbands = 3 + cpl_end_freq - cpl_begin_freq;
+ s->start_freq[CPL_CH] = cpl_begin_freq * 12 + 37;
+ s->end_freq[CPL_CH] = cpl_end_freq * 12 + 73;
+ for (bnd = 0; bnd < s->num_cpl_subbands - 1; bnd++) {
+ if (get_bits1(gbc)) {
+ s->cpl_band_struct[bnd] = 1;
+ s->num_cpl_bands--;
+ }
+ }
+ s->cpl_band_struct[s->num_cpl_subbands-1] = 0;
+ } else {
+ /* coupling not in use */
+ for (ch = 1; ch <= fbw_channels; ch++)
+ s->channel_in_cpl[ch] = 0;
+ }
+ }
+
+ /* coupling coordinates */
+ if (s->cpl_in_use) {
+ int cpl_coords_exist = 0;
+
+ for (ch = 1; ch <= fbw_channels; ch++) {
+ if (s->channel_in_cpl[ch]) {
+ if (get_bits1(gbc)) {
+ int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
+ cpl_coords_exist = 1;
+ master_cpl_coord = 3 * get_bits(gbc, 2);
+ for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
+ cpl_coord_exp = get_bits(gbc, 4);
+ cpl_coord_mant = get_bits(gbc, 4);
+ if (cpl_coord_exp == 15)
+ s->cpl_coords[ch][bnd] = cpl_coord_mant << 22;
+ else
+ s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16) << 21;
+ s->cpl_coords[ch][bnd] >>= (cpl_coord_exp + master_cpl_coord);
+ }
+ }
+ }
+ }
+ /* phase flags */
+ if (channel_mode == AC3_CHMODE_STEREO && cpl_coords_exist) {
+ for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
+ s->phase_flags[bnd] = s->phase_flags_in_use? get_bits1(gbc) : 0;
+ }
+ }
+ }
+
+ /* stereo rematrixing strategy and band structure */
+ if (channel_mode == AC3_CHMODE_STEREO) {
+ if (get_bits1(gbc)) {
+ s->num_rematrixing_bands = 4;
+ if(s->cpl_in_use && s->start_freq[CPL_CH] <= 61)
+ s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37);
+ for(bnd=0; bnd<s->num_rematrixing_bands; bnd++)
+ s->rematrixing_flags[bnd] = get_bits1(gbc);
+ }
+ }
+
+ /* exponent strategies for each channel */
+ s->exp_strategy[CPL_CH] = EXP_REUSE;
+ s->exp_strategy[s->lfe_ch] = EXP_REUSE;
+ for (ch = !s->cpl_in_use; ch <= s->channels; ch++) {
+ if(ch == s->lfe_ch)
+ s->exp_strategy[ch] = get_bits(gbc, 1);
+ else
+ s->exp_strategy[ch] = get_bits(gbc, 2);
+ if(s->exp_strategy[ch] != EXP_REUSE)
+ bit_alloc_stages[ch] = 3;
+ }
+
+ /* channel bandwidth */
+ for (ch = 1; ch <= fbw_channels; ch++) {
+ s->start_freq[ch] = 0;
+ if (s->exp_strategy[ch] != EXP_REUSE) {
+ int prev = s->end_freq[ch];
+ if (s->channel_in_cpl[ch])
+ s->end_freq[ch] = s->start_freq[CPL_CH];
+ else {
+ int bandwidth_code = get_bits(gbc, 6);
+ if (bandwidth_code > 60) {
+ av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60", bandwidth_code);
+ return -1;
+ }
+ s->end_freq[ch] = bandwidth_code * 3 + 73;
+ }
+ if(blk > 0 && s->end_freq[ch] != prev)
+ memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
}
}
- return buf_ptr - buf;
+ s->start_freq[s->lfe_ch] = 0;
+ s->end_freq[s->lfe_ch] = 7;
+
+ /* decode exponents for each channel */
+ for (ch = !s->cpl_in_use; ch <= s->channels; ch++) {
+ if (s->exp_strategy[ch] != EXP_REUSE) {
+ int group_size, num_groups;
+ group_size = 3 << (s->exp_strategy[ch] - 1);
+ if(ch == CPL_CH)
+ num_groups = (s->end_freq[ch] - s->start_freq[ch]) / group_size;
+ else if(ch == s->lfe_ch)
+ num_groups = 2;
+ else
+ num_groups = (s->end_freq[ch] + group_size - 4) / group_size;
+ s->dexps[ch][0] = get_bits(gbc, 4) << !ch;
+ decode_exponents(gbc, s->exp_strategy[ch], num_groups, s->dexps[ch][0],
+ &s->dexps[ch][s->start_freq[ch]+!!ch]);
+ if(ch != CPL_CH && ch != s->lfe_ch)
+ skip_bits(gbc, 2); /* skip gainrng */
+ }
+ }
+
+ /* bit allocation information */
+ if (get_bits1(gbc)) {
+ s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
+ s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
+ s->bit_alloc_params.slow_gain = ff_ac3_slow_gain_tab[get_bits(gbc, 2)];
+ s->bit_alloc_params.db_per_bit = ff_ac3_db_per_bit_tab[get_bits(gbc, 2)];
+ s->bit_alloc_params.floor = ff_ac3_floor_tab[get_bits(gbc, 3)];
+ for(ch=!s->cpl_in_use; ch<=s->channels; ch++) {
+ bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
+ }
+ }
+
+ /* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */
+ if (get_bits1(gbc)) {
+ int csnr;
+ csnr = (get_bits(gbc, 6) - 15) << 4;
+ for (ch = !s->cpl_in_use; ch <= s->channels; ch++) { /* snr offset and fast gain */
+ s->snr_offset[ch] = (csnr + get_bits(gbc, 4)) << 2;
+ s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)];
+ }
+ memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
+ }
+
+ /* coupling leak information */
+ if (s->cpl_in_use && get_bits1(gbc)) {
+ s->bit_alloc_params.cpl_fast_leak = get_bits(gbc, 3);
+ s->bit_alloc_params.cpl_slow_leak = get_bits(gbc, 3);
+ bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2);
+ }
+
+ /* delta bit allocation information */
+ if (get_bits1(gbc)) {
+ /* delta bit allocation exists (strategy) */
+ for (ch = !s->cpl_in_use; ch <= fbw_channels; ch++) {
+ s->dba_mode[ch] = get_bits(gbc, 2);
+ if (s->dba_mode[ch] == DBA_RESERVED) {
+ av_log(s->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n");
+ return -1;
+ }
+ bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
+ }
+ /* channel delta offset, len and bit allocation */
+ for (ch = !s->cpl_in_use; ch <= fbw_channels; ch++) {
+ if (s->dba_mode[ch] == DBA_NEW) {
+ s->dba_nsegs[ch] = get_bits(gbc, 3);
+ for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) {
+ s->dba_offsets[ch][seg] = get_bits(gbc, 5);
+ s->dba_lengths[ch][seg] = get_bits(gbc, 4);
+ s->dba_values[ch][seg] = get_bits(gbc, 3);
+ }
+ }
+ }
+ } else if(blk == 0) {
+ for(ch=0; ch<=s->channels; ch++) {
+ s->dba_mode[ch] = DBA_NONE;
+ }
+ }
+
+ /* Bit allocation */
+ for(ch=!s->cpl_in_use; ch<=s->channels; ch++) {
+ if(bit_alloc_stages[ch] > 2) {
+ /* Exponent mapping into PSD and PSD integration */
+ ff_ac3_bit_alloc_calc_psd(s->dexps[ch],
+ s->start_freq[ch], s->end_freq[ch],
+ s->psd[ch], s->band_psd[ch]);
+ }
+ if(bit_alloc_stages[ch] > 1) {
+ /* Compute excitation function, Compute masking curve, and
+ Apply delta bit allocation */
+ ff_ac3_bit_alloc_calc_mask(&s->bit_alloc_params, s->band_psd[ch],
+ s->start_freq[ch], s->end_freq[ch],
+ s->fast_gain[ch], (ch == s->lfe_ch),
+ s->dba_mode[ch], s->dba_nsegs[ch],
+ s->dba_offsets[ch], s->dba_lengths[ch],
+ s->dba_values[ch], s->mask[ch]);
+ }
+ if(bit_alloc_stages[ch] > 0) {
+ /* Compute bit allocation */
+ ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
+ s->start_freq[ch], s->end_freq[ch],
+ s->snr_offset[ch],
+ s->bit_alloc_params.floor,
+ s->bap[ch]);
+ }
+ }
+
+ /* unused dummy data */
+ if (get_bits1(gbc)) {
+ int skipl = get_bits(gbc, 9);
+ while(skipl--)
+ skip_bits(gbc, 8);
+ }
+
+ /* unpack the transform coefficients
+ this also uncouples channels if coupling is in use. */
+ if (get_transform_coeffs(s)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Error in routine get_transform_coeffs\n");
+ return -1;
+ }
+
+ /* recover coefficients if rematrixing is in use */
+ if(s->channel_mode == AC3_CHMODE_STEREO)
+ do_rematrixing(s);
+
+ /* apply scaling to coefficients (headroom, dynrng) */
+ for(ch=1; ch<=s->channels; ch++) {
+ float gain = s->mul_bias / 4194304.0f;
+ if(s->channel_mode == AC3_CHMODE_DUALMONO) {
+ gain *= s->dynamic_range[ch-1];
+ } else {
+ gain *= s->dynamic_range[0];
+ }
+ for(i=0; i<256; i++) {
+ s->transform_coeffs[ch][i] = s->fixed_coeffs[ch][i] * gain;
+ }
+ }
+
+ /* downmix and MDCT. order depends on whether block switching is used for
+ any channel in this block. this is because coefficients for the long
+ and short transforms cannot be mixed. */
+ downmix_output = s->channels != s->out_channels &&
+ !((s->output_mode & AC3_OUTPUT_LFEON) &&
+ s->fbw_channels == s->out_channels);
+ if(different_transforms) {
+ /* the delay samples have already been downmixed, so we upmix the delay
+ samples in order to reconstruct all channels before downmixing. */
+ if(s->downmixed) {
+ s->downmixed = 0;
+ ac3_upmix_delay(s);
+ }
+
+ do_imdct(s, s->channels);
+
+ if(downmix_output) {
+ ac3_downmix(s, s->output, 0);
+ }
+ } else {
+ if(downmix_output) {
+ ac3_downmix(s, s->transform_coeffs, 1);
+ }
+
+ if(!s->downmixed) {
+ s->downmixed = 1;
+ ac3_downmix(s, s->delay, 0);
+ }
+
+ do_imdct(s, s->out_channels);
+ }
+
+ /* convert float to 16-bit integer */
+ for(ch=0; ch<s->out_channels; ch++) {
+ for(i=0; i<256; i++) {
+ s->output[ch][i] += s->add_bias;
+ }
+ s->dsp.float_to_int16(s->int_output[ch], s->output[ch], 256);
+ }
+
+ return 0;
+}
+
+/**
+ * Decode a single AC-3 frame.
+ */
+static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
+ const uint8_t *buf, int buf_size)
+{
+ AC3DecodeContext *s = avctx->priv_data;
+ int16_t *out_samples = (int16_t *)data;
+ int i, blk, ch, err;
+
+ /* initialize the GetBitContext with the start of valid AC-3 Frame */
+ init_get_bits(&s->gbc, buf, buf_size * 8);
+
+ /* parse the syncinfo */
+ err = ac3_parse_header(s);
+ if(err) {
+ switch(err) {
+ case AC3_PARSE_ERROR_SYNC:
+ av_log(avctx, AV_LOG_ERROR, "frame sync error\n");
+ break;
+ case AC3_PARSE_ERROR_BSID:
+ av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n");
+ break;
+ case AC3_PARSE_ERROR_SAMPLE_RATE:
+ av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
+ break;
+ case AC3_PARSE_ERROR_FRAME_SIZE:
+ av_log(avctx, AV_LOG_ERROR, "invalid frame size\n");
+ break;
+ default:
+ av_log(avctx, AV_LOG_ERROR, "invalid header\n");
+ break;
+ }
+ return -1;
+ }
+
+ /* check that reported frame size fits in input buffer */
+ if(s->frame_size > buf_size) {
+ av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
+ return -1;
+ }
+
+ /* check for crc mismatch */
+ if(avctx->error_resilience >= FF_ER_CAREFUL) {
+ if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) {
+ av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
+ return -1;
+ }
+ /* TODO: error concealment */
+ }
+
+ avctx->sample_rate = s->sample_rate;
+ avctx->bit_rate = s->bit_rate;
+
+ /* channel config */
+ s->out_channels = s->channels;
+ if (avctx->request_channels > 0 && avctx->request_channels <= 2 &&
+ avctx->request_channels < s->channels) {
+ s->out_channels = avctx->request_channels;
+ s->output_mode = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
+ }
+ avctx->channels = s->out_channels;
+
+ /* set downmixing coefficients if needed */
+ if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
+ s->fbw_channels == s->out_channels)) {
+ set_downmix_coeffs(s);
+ }
+
+ /* parse the audio blocks */
+ for (blk = 0; blk < NB_BLOCKS; blk++) {
+ if (ac3_parse_audio_block(s, blk)) {
+ av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n");
+ *data_size = 0;
+ return s->frame_size;
+ }
+ for (i = 0; i < 256; i++)
+ for (ch = 0; ch < s->out_channels; ch++)
+ *(out_samples++) = s->int_output[ch][i];
+ }
+ *data_size = NB_BLOCKS * 256 * avctx->channels * sizeof (int16_t);
+ return s->frame_size;
}
-static int ac3_decode_end(AVCodecContext *s)
+/**
+ * Uninitialize the AC-3 decoder.
+ */
+static av_cold int ac3_decode_end(AVCodecContext *avctx)
{
+ AC3DecodeContext *s = avctx->priv_data;
+ ff_mdct_end(&s->imdct_512);
+ ff_mdct_end(&s->imdct_256);
+
return 0;
}
AVCodec ac3_decoder = {
- "ac3",
- CODEC_TYPE_AUDIO,
- CODEC_ID_AC3,
- sizeof(AC3DecodeState),
- ac3_decode_init,
- NULL,
- ac3_decode_end,
- ac3_decode_frame,
+ .name = "ac3",
+ .type = CODEC_TYPE_AUDIO,
+ .id = CODEC_ID_AC3,
+ .priv_data_size = sizeof (AC3DecodeContext),
+ .init = ac3_decode_init,
+ .close = ac3_decode_end,
+ .decode = ac3_decode_frame,
};
projects / ffmpeg / blobdiff