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-rw-r--r--media/ffvpx/libavcodec/mpegaudiodec_template.c1899
1 files changed, 1899 insertions, 0 deletions
diff --git a/media/ffvpx/libavcodec/mpegaudiodec_template.c b/media/ffvpx/libavcodec/mpegaudiodec_template.c
new file mode 100644
index 0000000000..3e4ee79be6
--- /dev/null
+++ b/media/ffvpx/libavcodec/mpegaudiodec_template.c
@@ -0,0 +1,1899 @@
+/*
+ * MPEG Audio decoder
+ * Copyright (c) 2001, 2002 Fabrice Bellard
+ *
+ * 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
+ * MPEG Audio decoder
+ */
+
+#include "config_components.h"
+
+#include "libavutil/attributes.h"
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/crc.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/libm.h"
+#include "libavutil/mem_internal.h"
+#include "libavutil/thread.h"
+
+#include "avcodec.h"
+#include "decode.h"
+#include "get_bits.h"
+#include "mathops.h"
+#include "mpegaudiodsp.h"
+
+/*
+ * TODO:
+ * - test lsf / mpeg25 extensively.
+ */
+
+#include "mpegaudio.h"
+#include "mpegaudiodecheader.h"
+
+#define BACKSTEP_SIZE 512
+#define EXTRABYTES 24
+#define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
+
+/* layer 3 "granule" */
+typedef struct GranuleDef {
+ uint8_t scfsi;
+ int part2_3_length;
+ int big_values;
+ int global_gain;
+ int scalefac_compress;
+ uint8_t block_type;
+ uint8_t switch_point;
+ int table_select[3];
+ int subblock_gain[3];
+ uint8_t scalefac_scale;
+ uint8_t count1table_select;
+ int region_size[3]; /* number of huffman codes in each region */
+ int preflag;
+ int short_start, long_end; /* long/short band indexes */
+ uint8_t scale_factors[40];
+ DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
+} GranuleDef;
+
+typedef struct MPADecodeContext {
+ MPA_DECODE_HEADER
+ uint8_t last_buf[LAST_BUF_SIZE];
+ int last_buf_size;
+ int extrasize;
+ /* next header (used in free format parsing) */
+ uint32_t free_format_next_header;
+ GetBitContext gb;
+ GetBitContext in_gb;
+ DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
+ int synth_buf_offset[MPA_MAX_CHANNELS];
+ DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
+ INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
+ GranuleDef granules[2][2]; /* Used in Layer 3 */
+ int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
+ int dither_state;
+ int err_recognition;
+ AVCodecContext* avctx;
+ MPADSPContext mpadsp;
+ void (*butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len);
+ AVFrame *frame;
+ uint32_t crc;
+} MPADecodeContext;
+
+#define HEADER_SIZE 4
+
+#include "mpegaudiodata.h"
+
+#include "mpegaudio_tablegen.h"
+/* intensity stereo coef table */
+static INTFLOAT is_table_lsf[2][2][16];
+
+/* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
+static int32_t scale_factor_mult[15][3];
+/* mult table for layer 2 group quantization */
+
+#define SCALE_GEN(v) \
+{ FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
+
+static const int32_t scale_factor_mult2[3][3] = {
+ SCALE_GEN(4.0 / 3.0), /* 3 steps */
+ SCALE_GEN(4.0 / 5.0), /* 5 steps */
+ SCALE_GEN(4.0 / 9.0), /* 9 steps */
+};
+
+/**
+ * Convert region offsets to region sizes and truncate
+ * size to big_values.
+ */
+static void region_offset2size(GranuleDef *g)
+{
+ int i, k, j = 0;
+ g->region_size[2] = 576 / 2;
+ for (i = 0; i < 3; i++) {
+ k = FFMIN(g->region_size[i], g->big_values);
+ g->region_size[i] = k - j;
+ j = k;
+ }
+}
+
+static void init_short_region(MPADecodeContext *s, GranuleDef *g)
+{
+ if (g->block_type == 2) {
+ if (s->sample_rate_index != 8)
+ g->region_size[0] = (36 / 2);
+ else
+ g->region_size[0] = (72 / 2);
+ } else {
+ if (s->sample_rate_index <= 2)
+ g->region_size[0] = (36 / 2);
+ else if (s->sample_rate_index != 8)
+ g->region_size[0] = (54 / 2);
+ else
+ g->region_size[0] = (108 / 2);
+ }
+ g->region_size[1] = (576 / 2);
+}
+
+static void init_long_region(MPADecodeContext *s, GranuleDef *g,
+ int ra1, int ra2)
+{
+ int l;
+ g->region_size[0] = ff_band_index_long[s->sample_rate_index][ra1 + 1];
+ /* should not overflow */
+ l = FFMIN(ra1 + ra2 + 2, 22);
+ g->region_size[1] = ff_band_index_long[s->sample_rate_index][ l];
+}
+
+static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
+{
+ if (g->block_type == 2) {
+ if (g->switch_point) {
+ if(s->sample_rate_index == 8)
+ avpriv_request_sample(s->avctx, "switch point in 8khz");
+ /* if switched mode, we handle the 36 first samples as
+ long blocks. For 8000Hz, we handle the 72 first
+ exponents as long blocks */
+ if (s->sample_rate_index <= 2)
+ g->long_end = 8;
+ else
+ g->long_end = 6;
+
+ g->short_start = 3;
+ } else {
+ g->long_end = 0;
+ g->short_start = 0;
+ }
+ } else {
+ g->short_start = 13;
+ g->long_end = 22;
+ }
+}
+
+/* layer 1 unscaling */
+/* n = number of bits of the mantissa minus 1 */
+static inline int l1_unscale(int n, int mant, int scale_factor)
+{
+ int shift, mod;
+ int64_t val;
+
+ shift = ff_scale_factor_modshift[scale_factor];
+ mod = shift & 3;
+ shift >>= 2;
+ val = MUL64((int)(mant + (-1U << n) + 1), scale_factor_mult[n-1][mod]);
+ shift += n;
+ /* NOTE: at this point, 1 <= shift >= 21 + 15 */
+ return (int)((val + (1LL << (shift - 1))) >> shift);
+}
+
+static inline int l2_unscale_group(int steps, int mant, int scale_factor)
+{
+ int shift, mod, val;
+
+ shift = ff_scale_factor_modshift[scale_factor];
+ mod = shift & 3;
+ shift >>= 2;
+
+ val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
+ /* NOTE: at this point, 0 <= shift <= 21 */
+ if (shift > 0)
+ val = (val + (1 << (shift - 1))) >> shift;
+ return val;
+}
+
+/* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
+static inline int l3_unscale(int value, int exponent)
+{
+ unsigned int m;
+ int e;
+
+ e = ff_table_4_3_exp [4 * value + (exponent & 3)];
+ m = ff_table_4_3_value[4 * value + (exponent & 3)];
+ e -= exponent >> 2;
+#ifdef DEBUG
+ if(e < 1)
+ av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
+#endif
+ if (e > (SUINT)31)
+ return 0;
+ m = (m + ((1U << e) >> 1)) >> e;
+
+ return m;
+}
+
+static av_cold void decode_init_static(void)
+{
+ int i, j;
+
+ /* scale factor multiply for layer 1 */
+ for (i = 0; i < 15; i++) {
+ int n, norm;
+ n = i + 2;
+ norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
+ scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
+ scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
+ scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
+ ff_dlog(NULL, "%d: norm=%x s=%"PRIx32" %"PRIx32" %"PRIx32"\n", i,
+ (unsigned)norm,
+ scale_factor_mult[i][0],
+ scale_factor_mult[i][1],
+ scale_factor_mult[i][2]);
+ }
+
+ /* compute n ^ (4/3) and store it in mantissa/exp format */
+
+ mpegaudio_tableinit();
+
+ for (i = 0; i < 16; i++) {
+ double f;
+ int e, k;
+
+ for (j = 0; j < 2; j++) {
+ e = -(j + 1) * ((i + 1) >> 1);
+ f = exp2(e / 4.0);
+ k = i & 1;
+ is_table_lsf[j][k ^ 1][i] = FIXR(f);
+ is_table_lsf[j][k ][i] = FIXR(1.0);
+ ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
+ i, j, (float) is_table_lsf[j][0][i],
+ (float) is_table_lsf[j][1][i]);
+ }
+ }
+ RENAME(ff_mpa_synth_init)();
+ ff_mpegaudiodec_common_init_static();
+}
+
+static av_cold int decode_init(AVCodecContext * avctx)
+{
+ static AVOnce init_static_once = AV_ONCE_INIT;
+ MPADecodeContext *s = avctx->priv_data;
+
+ s->avctx = avctx;
+
+#if USE_FLOATS
+ {
+ AVFloatDSPContext *fdsp;
+ fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+ if (!fdsp)
+ return AVERROR(ENOMEM);
+ s->butterflies_float = fdsp->butterflies_float;
+ av_free(fdsp);
+ }
+#endif
+
+ ff_mpadsp_init(&s->mpadsp);
+
+ if (avctx->request_sample_fmt == OUT_FMT &&
+ avctx->codec_id != AV_CODEC_ID_MP3ON4)
+ avctx->sample_fmt = OUT_FMT;
+ else
+ avctx->sample_fmt = OUT_FMT_P;
+ s->err_recognition = avctx->err_recognition;
+
+ if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
+ s->adu_mode = 1;
+
+ ff_thread_once(&init_static_once, decode_init_static);
+
+ return 0;
+}
+
+#define C3 FIXHR(0.86602540378443864676/2)
+#define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
+#define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
+#define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
+
+/* 12 points IMDCT. We compute it "by hand" by factorizing obvious
+ cases. */
+static void imdct12(INTFLOAT *out, SUINTFLOAT *in)
+{
+ SUINTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
+
+ in0 = in[0*3];
+ in1 = in[1*3] + in[0*3];
+ in2 = in[2*3] + in[1*3];
+ in3 = in[3*3] + in[2*3];
+ in4 = in[4*3] + in[3*3];
+ in5 = in[5*3] + in[4*3];
+ in5 += in3;
+ in3 += in1;
+
+ in2 = MULH3(in2, C3, 2);
+ in3 = MULH3(in3, C3, 4);
+
+ t1 = in0 - in4;
+ t2 = MULH3(in1 - in5, C4, 2);
+
+ out[ 7] =
+ out[10] = t1 + t2;
+ out[ 1] =
+ out[ 4] = t1 - t2;
+
+ in0 += SHR(in4, 1);
+ in4 = in0 + in2;
+ in5 += 2*in1;
+ in1 = MULH3(in5 + in3, C5, 1);
+ out[ 8] =
+ out[ 9] = in4 + in1;
+ out[ 2] =
+ out[ 3] = in4 - in1;
+
+ in0 -= in2;
+ in5 = MULH3(in5 - in3, C6, 2);
+ out[ 0] =
+ out[ 5] = in0 - in5;
+ out[ 6] =
+ out[11] = in0 + in5;
+}
+
+static int handle_crc(MPADecodeContext *s, int sec_len)
+{
+ if (s->error_protection && (s->err_recognition & AV_EF_CRCCHECK)) {
+ const uint8_t *buf = s->gb.buffer - HEADER_SIZE;
+ int sec_byte_len = sec_len >> 3;
+ int sec_rem_bits = sec_len & 7;
+ const AVCRC *crc_tab = av_crc_get_table(AV_CRC_16_ANSI);
+ uint8_t tmp_buf[4];
+ uint32_t crc_val = av_crc(crc_tab, UINT16_MAX, &buf[2], 2);
+ crc_val = av_crc(crc_tab, crc_val, &buf[6], sec_byte_len);
+
+ AV_WB32(tmp_buf,
+ ((buf[6 + sec_byte_len] & (0xFF00U >> sec_rem_bits)) << 24) +
+ ((s->crc << 16) >> sec_rem_bits));
+
+ crc_val = av_crc(crc_tab, crc_val, tmp_buf, 3);
+
+ if (crc_val) {
+ av_log(s->avctx, AV_LOG_ERROR, "CRC mismatch %X!\n", crc_val);
+ if (s->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ return 0;
+}
+
+/* return the number of decoded frames */
+static int mp_decode_layer1(MPADecodeContext *s)
+{
+ int bound, i, v, n, ch, j, mant;
+ uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
+ uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
+ int ret;
+
+ ret = handle_crc(s, (s->nb_channels == 1) ? 8*16 : 8*32);
+ if (ret < 0)
+ return ret;
+
+ if (s->mode == MPA_JSTEREO)
+ bound = (s->mode_ext + 1) * 4;
+ else
+ bound = SBLIMIT;
+
+ /* allocation bits */
+ for (i = 0; i < bound; i++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ allocation[ch][i] = get_bits(&s->gb, 4);
+ }
+ }
+ for (i = bound; i < SBLIMIT; i++)
+ allocation[0][i] = get_bits(&s->gb, 4);
+
+ /* scale factors */
+ for (i = 0; i < bound; i++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ if (allocation[ch][i])
+ scale_factors[ch][i] = get_bits(&s->gb, 6);
+ }
+ }
+ for (i = bound; i < SBLIMIT; i++) {
+ if (allocation[0][i]) {
+ scale_factors[0][i] = get_bits(&s->gb, 6);
+ scale_factors[1][i] = get_bits(&s->gb, 6);
+ }
+ }
+
+ /* compute samples */
+ for (j = 0; j < 12; j++) {
+ for (i = 0; i < bound; i++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ n = allocation[ch][i];
+ if (n) {
+ mant = get_bits(&s->gb, n + 1);
+ v = l1_unscale(n, mant, scale_factors[ch][i]);
+ } else {
+ v = 0;
+ }
+ s->sb_samples[ch][j][i] = v;
+ }
+ }
+ for (i = bound; i < SBLIMIT; i++) {
+ n = allocation[0][i];
+ if (n) {
+ mant = get_bits(&s->gb, n + 1);
+ v = l1_unscale(n, mant, scale_factors[0][i]);
+ s->sb_samples[0][j][i] = v;
+ v = l1_unscale(n, mant, scale_factors[1][i]);
+ s->sb_samples[1][j][i] = v;
+ } else {
+ s->sb_samples[0][j][i] = 0;
+ s->sb_samples[1][j][i] = 0;
+ }
+ }
+ }
+ return 12;
+}
+
+static int mp_decode_layer2(MPADecodeContext *s)
+{
+ int sblimit; /* number of used subbands */
+ const unsigned char *alloc_table;
+ int table, bit_alloc_bits, i, j, ch, bound, v;
+ unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
+ unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
+ unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
+ int scale, qindex, bits, steps, k, l, m, b;
+ int ret;
+
+ /* select decoding table */
+ table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
+ s->sample_rate, s->lsf);
+ sblimit = ff_mpa_sblimit_table[table];
+ alloc_table = ff_mpa_alloc_tables[table];
+
+ if (s->mode == MPA_JSTEREO)
+ bound = (s->mode_ext + 1) * 4;
+ else
+ bound = sblimit;
+
+ ff_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
+
+ /* sanity check */
+ if (bound > sblimit)
+ bound = sblimit;
+
+ /* parse bit allocation */
+ j = 0;
+ for (i = 0; i < bound; i++) {
+ bit_alloc_bits = alloc_table[j];
+ for (ch = 0; ch < s->nb_channels; ch++)
+ bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
+ j += 1 << bit_alloc_bits;
+ }
+ for (i = bound; i < sblimit; i++) {
+ bit_alloc_bits = alloc_table[j];
+ v = get_bits(&s->gb, bit_alloc_bits);
+ bit_alloc[0][i] = v;
+ bit_alloc[1][i] = v;
+ j += 1 << bit_alloc_bits;
+ }
+
+ /* scale codes */
+ for (i = 0; i < sblimit; i++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ if (bit_alloc[ch][i])
+ scale_code[ch][i] = get_bits(&s->gb, 2);
+ }
+ }
+
+ ret = handle_crc(s, get_bits_count(&s->gb) - 16);
+ if (ret < 0)
+ return ret;
+
+ /* scale factors */
+ for (i = 0; i < sblimit; i++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ if (bit_alloc[ch][i]) {
+ sf = scale_factors[ch][i];
+ switch (scale_code[ch][i]) {
+ default:
+ case 0:
+ sf[0] = get_bits(&s->gb, 6);
+ sf[1] = get_bits(&s->gb, 6);
+ sf[2] = get_bits(&s->gb, 6);
+ break;
+ case 2:
+ sf[0] = get_bits(&s->gb, 6);
+ sf[1] = sf[0];
+ sf[2] = sf[0];
+ break;
+ case 1:
+ sf[0] = get_bits(&s->gb, 6);
+ sf[2] = get_bits(&s->gb, 6);
+ sf[1] = sf[0];
+ break;
+ case 3:
+ sf[0] = get_bits(&s->gb, 6);
+ sf[2] = get_bits(&s->gb, 6);
+ sf[1] = sf[2];
+ break;
+ }
+ }
+ }
+ }
+
+ /* samples */
+ for (k = 0; k < 3; k++) {
+ for (l = 0; l < 12; l += 3) {
+ j = 0;
+ for (i = 0; i < bound; i++) {
+ bit_alloc_bits = alloc_table[j];
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ b = bit_alloc[ch][i];
+ if (b) {
+ scale = scale_factors[ch][i][k];
+ qindex = alloc_table[j+b];
+ bits = ff_mpa_quant_bits[qindex];
+ if (bits < 0) {
+ int v2;
+ /* 3 values at the same time */
+ v = get_bits(&s->gb, -bits);
+ v2 = ff_division_tabs[qindex][v];
+ steps = ff_mpa_quant_steps[qindex];
+
+ s->sb_samples[ch][k * 12 + l + 0][i] =
+ l2_unscale_group(steps, v2 & 15, scale);
+ s->sb_samples[ch][k * 12 + l + 1][i] =
+ l2_unscale_group(steps, (v2 >> 4) & 15, scale);
+ s->sb_samples[ch][k * 12 + l + 2][i] =
+ l2_unscale_group(steps, v2 >> 8 , scale);
+ } else {
+ for (m = 0; m < 3; m++) {
+ v = get_bits(&s->gb, bits);
+ v = l1_unscale(bits - 1, v, scale);
+ s->sb_samples[ch][k * 12 + l + m][i] = v;
+ }
+ }
+ } else {
+ s->sb_samples[ch][k * 12 + l + 0][i] = 0;
+ s->sb_samples[ch][k * 12 + l + 1][i] = 0;
+ s->sb_samples[ch][k * 12 + l + 2][i] = 0;
+ }
+ }
+ /* next subband in alloc table */
+ j += 1 << bit_alloc_bits;
+ }
+ /* XXX: find a way to avoid this duplication of code */
+ for (i = bound; i < sblimit; i++) {
+ bit_alloc_bits = alloc_table[j];
+ b = bit_alloc[0][i];
+ if (b) {
+ int mant, scale0, scale1;
+ scale0 = scale_factors[0][i][k];
+ scale1 = scale_factors[1][i][k];
+ qindex = alloc_table[j + b];
+ bits = ff_mpa_quant_bits[qindex];
+ if (bits < 0) {
+ /* 3 values at the same time */
+ v = get_bits(&s->gb, -bits);
+ steps = ff_mpa_quant_steps[qindex];
+ mant = v % steps;
+ v = v / steps;
+ s->sb_samples[0][k * 12 + l + 0][i] =
+ l2_unscale_group(steps, mant, scale0);
+ s->sb_samples[1][k * 12 + l + 0][i] =
+ l2_unscale_group(steps, mant, scale1);
+ mant = v % steps;
+ v = v / steps;
+ s->sb_samples[0][k * 12 + l + 1][i] =
+ l2_unscale_group(steps, mant, scale0);
+ s->sb_samples[1][k * 12 + l + 1][i] =
+ l2_unscale_group(steps, mant, scale1);
+ s->sb_samples[0][k * 12 + l + 2][i] =
+ l2_unscale_group(steps, v, scale0);
+ s->sb_samples[1][k * 12 + l + 2][i] =
+ l2_unscale_group(steps, v, scale1);
+ } else {
+ for (m = 0; m < 3; m++) {
+ mant = get_bits(&s->gb, bits);
+ s->sb_samples[0][k * 12 + l + m][i] =
+ l1_unscale(bits - 1, mant, scale0);
+ s->sb_samples[1][k * 12 + l + m][i] =
+ l1_unscale(bits - 1, mant, scale1);
+ }
+ }
+ } else {
+ s->sb_samples[0][k * 12 + l + 0][i] = 0;
+ s->sb_samples[0][k * 12 + l + 1][i] = 0;
+ s->sb_samples[0][k * 12 + l + 2][i] = 0;
+ s->sb_samples[1][k * 12 + l + 0][i] = 0;
+ s->sb_samples[1][k * 12 + l + 1][i] = 0;
+ s->sb_samples[1][k * 12 + l + 2][i] = 0;
+ }
+ /* next subband in alloc table */
+ j += 1 << bit_alloc_bits;
+ }
+ /* fill remaining samples to zero */
+ for (i = sblimit; i < SBLIMIT; i++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ s->sb_samples[ch][k * 12 + l + 0][i] = 0;
+ s->sb_samples[ch][k * 12 + l + 1][i] = 0;
+ s->sb_samples[ch][k * 12 + l + 2][i] = 0;
+ }
+ }
+ }
+ }
+ return 3 * 12;
+}
+
+#define SPLIT(dst,sf,n) \
+ if (n == 3) { \
+ int m = (sf * 171) >> 9; \
+ dst = sf - 3 * m; \
+ sf = m; \
+ } else if (n == 4) { \
+ dst = sf & 3; \
+ sf >>= 2; \
+ } else if (n == 5) { \
+ int m = (sf * 205) >> 10; \
+ dst = sf - 5 * m; \
+ sf = m; \
+ } else if (n == 6) { \
+ int m = (sf * 171) >> 10; \
+ dst = sf - 6 * m; \
+ sf = m; \
+ } else { \
+ dst = 0; \
+ }
+
+static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
+ int n3)
+{
+ SPLIT(slen[3], sf, n3)
+ SPLIT(slen[2], sf, n2)
+ SPLIT(slen[1], sf, n1)
+ slen[0] = sf;
+}
+
+static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
+ int16_t *exponents)
+{
+ const uint8_t *bstab, *pretab;
+ int len, i, j, k, l, v0, shift, gain, gains[3];
+ int16_t *exp_ptr;
+
+ exp_ptr = exponents;
+ gain = g->global_gain - 210;
+ shift = g->scalefac_scale + 1;
+
+ bstab = ff_band_size_long[s->sample_rate_index];
+ pretab = ff_mpa_pretab[g->preflag];
+ for (i = 0; i < g->long_end; i++) {
+ v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
+ len = bstab[i];
+ for (j = len; j > 0; j--)
+ *exp_ptr++ = v0;
+ }
+
+ if (g->short_start < 13) {
+ bstab = ff_band_size_short[s->sample_rate_index];
+ gains[0] = gain - (g->subblock_gain[0] << 3);
+ gains[1] = gain - (g->subblock_gain[1] << 3);
+ gains[2] = gain - (g->subblock_gain[2] << 3);
+ k = g->long_end;
+ for (i = g->short_start; i < 13; i++) {
+ len = bstab[i];
+ for (l = 0; l < 3; l++) {
+ v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
+ for (j = len; j > 0; j--)
+ *exp_ptr++ = v0;
+ }
+ }
+ }
+}
+
+static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
+ int *end_pos2)
+{
+ if (s->in_gb.buffer && *pos >= s->gb.size_in_bits - s->extrasize * 8) {
+ s->gb = s->in_gb;
+ s->in_gb.buffer = NULL;
+ s->extrasize = 0;
+ av_assert2((get_bits_count(&s->gb) & 7) == 0);
+ skip_bits_long(&s->gb, *pos - *end_pos);
+ *end_pos2 =
+ *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
+ *pos = get_bits_count(&s->gb);
+ }
+}
+
+/* Following is an optimized code for
+ INTFLOAT v = *src
+ if(get_bits1(&s->gb))
+ v = -v;
+ *dst = v;
+*/
+#if USE_FLOATS
+#define READ_FLIP_SIGN(dst,src) \
+ v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
+ AV_WN32A(dst, v);
+#else
+#define READ_FLIP_SIGN(dst,src) \
+ v = -get_bits1(&s->gb); \
+ *(dst) = (*(src) ^ v) - v;
+#endif
+
+static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
+ int16_t *exponents, int end_pos2)
+{
+ int s_index;
+ int i;
+ int last_pos, bits_left;
+ VLC *vlc;
+ int end_pos = FFMIN(end_pos2, s->gb.size_in_bits - s->extrasize * 8);
+
+ /* low frequencies (called big values) */
+ s_index = 0;
+ for (i = 0; i < 3; i++) {
+ int j, k, l, linbits;
+ j = g->region_size[i];
+ if (j == 0)
+ continue;
+ /* select vlc table */
+ k = g->table_select[i];
+ l = ff_mpa_huff_data[k][0];
+ linbits = ff_mpa_huff_data[k][1];
+ vlc = &ff_huff_vlc[l];
+
+ if (!l) {
+ memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
+ s_index += 2 * j;
+ continue;
+ }
+
+ /* read huffcode and compute each couple */
+ for (; j > 0; j--) {
+ int exponent, x, y;
+ int v;
+ int pos = get_bits_count(&s->gb);
+
+ if (pos >= end_pos){
+ switch_buffer(s, &pos, &end_pos, &end_pos2);
+ if (pos >= end_pos)
+ break;
+ }
+ y = get_vlc2(&s->gb, vlc->table, 7, 3);
+
+ if (!y) {
+ g->sb_hybrid[s_index ] =
+ g->sb_hybrid[s_index + 1] = 0;
+ s_index += 2;
+ continue;
+ }
+
+ exponent= exponents[s_index];
+
+ ff_dlog(s->avctx, "region=%d n=%d y=%d exp=%d\n",
+ i, g->region_size[i] - j, y, exponent);
+ if (y & 16) {
+ x = y >> 5;
+ y = y & 0x0f;
+ if (x < 15) {
+ READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
+ } else {
+ x += get_bitsz(&s->gb, linbits);
+ v = l3_unscale(x, exponent);
+ if (get_bits1(&s->gb))
+ v = -v;
+ g->sb_hybrid[s_index] = v;
+ }
+ if (y < 15) {
+ READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
+ } else {
+ y += get_bitsz(&s->gb, linbits);
+ v = l3_unscale(y, exponent);
+ if (get_bits1(&s->gb))
+ v = -v;
+ g->sb_hybrid[s_index + 1] = v;
+ }
+ } else {
+ x = y >> 5;
+ y = y & 0x0f;
+ x += y;
+ if (x < 15) {
+ READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
+ } else {
+ x += get_bitsz(&s->gb, linbits);
+ v = l3_unscale(x, exponent);
+ if (get_bits1(&s->gb))
+ v = -v;
+ g->sb_hybrid[s_index+!!y] = v;
+ }
+ g->sb_hybrid[s_index + !y] = 0;
+ }
+ s_index += 2;
+ }
+ }
+
+ /* high frequencies */
+ vlc = &ff_huff_quad_vlc[g->count1table_select];
+ last_pos = 0;
+ while (s_index <= 572) {
+ int pos, code;
+ pos = get_bits_count(&s->gb);
+ if (pos >= end_pos) {
+ if (pos > end_pos2 && last_pos) {
+ /* some encoders generate an incorrect size for this
+ part. We must go back into the data */
+ s_index -= 4;
+ skip_bits_long(&s->gb, last_pos - pos);
+ av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
+ if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
+ s_index=0;
+ break;
+ }
+ switch_buffer(s, &pos, &end_pos, &end_pos2);
+ if (pos >= end_pos)
+ break;
+ }
+ last_pos = pos;
+
+ code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
+ ff_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
+ g->sb_hybrid[s_index + 0] =
+ g->sb_hybrid[s_index + 1] =
+ g->sb_hybrid[s_index + 2] =
+ g->sb_hybrid[s_index + 3] = 0;
+ while (code) {
+ static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
+ int v;
+ int pos = s_index + idxtab[code];
+ code ^= 8 >> idxtab[code];
+ READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
+ }
+ s_index += 4;
+ }
+ /* skip extension bits */
+ bits_left = end_pos2 - get_bits_count(&s->gb);
+ if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
+ av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
+ s_index=0;
+ } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
+ av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
+ s_index = 0;
+ }
+ memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
+ skip_bits_long(&s->gb, bits_left);
+
+ i = get_bits_count(&s->gb);
+ switch_buffer(s, &i, &end_pos, &end_pos2);
+
+ return 0;
+}
+
+/* Reorder short blocks from bitstream order to interleaved order. It
+ would be faster to do it in parsing, but the code would be far more
+ complicated */
+static void reorder_block(MPADecodeContext *s, GranuleDef *g)
+{
+ int i, j, len;
+ INTFLOAT *ptr, *dst, *ptr1;
+ INTFLOAT tmp[576];
+
+ if (g->block_type != 2)
+ return;
+
+ if (g->switch_point) {
+ if (s->sample_rate_index != 8)
+ ptr = g->sb_hybrid + 36;
+ else
+ ptr = g->sb_hybrid + 72;
+ } else {
+ ptr = g->sb_hybrid;
+ }
+
+ for (i = g->short_start; i < 13; i++) {
+ len = ff_band_size_short[s->sample_rate_index][i];
+ ptr1 = ptr;
+ dst = tmp;
+ for (j = len; j > 0; j--) {
+ *dst++ = ptr[0*len];
+ *dst++ = ptr[1*len];
+ *dst++ = ptr[2*len];
+ ptr++;
+ }
+ ptr += 2 * len;
+ memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
+ }
+}
+
+#define ISQRT2 FIXR(0.70710678118654752440)
+
+static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
+{
+ int i, j, k, l;
+ int sf_max, sf, len, non_zero_found;
+ INTFLOAT *tab0, *tab1, v1, v2;
+ const INTFLOAT (*is_tab)[16];
+ SUINTFLOAT tmp0, tmp1;
+ int non_zero_found_short[3];
+
+ /* intensity stereo */
+ if (s->mode_ext & MODE_EXT_I_STEREO) {
+ if (!s->lsf) {
+ is_tab = is_table;
+ sf_max = 7;
+ } else {
+ is_tab = is_table_lsf[g1->scalefac_compress & 1];
+ sf_max = 16;
+ }
+
+ tab0 = g0->sb_hybrid + 576;
+ tab1 = g1->sb_hybrid + 576;
+
+ non_zero_found_short[0] = 0;
+ non_zero_found_short[1] = 0;
+ non_zero_found_short[2] = 0;
+ k = (13 - g1->short_start) * 3 + g1->long_end - 3;
+ for (i = 12; i >= g1->short_start; i--) {
+ /* for last band, use previous scale factor */
+ if (i != 11)
+ k -= 3;
+ len = ff_band_size_short[s->sample_rate_index][i];
+ for (l = 2; l >= 0; l--) {
+ tab0 -= len;
+ tab1 -= len;
+ if (!non_zero_found_short[l]) {
+ /* test if non zero band. if so, stop doing i-stereo */
+ for (j = 0; j < len; j++) {
+ if (tab1[j] != 0) {
+ non_zero_found_short[l] = 1;
+ goto found1;
+ }
+ }
+ sf = g1->scale_factors[k + l];
+ if (sf >= sf_max)
+ goto found1;
+
+ v1 = is_tab[0][sf];
+ v2 = is_tab[1][sf];
+ for (j = 0; j < len; j++) {
+ tmp0 = tab0[j];
+ tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
+ tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
+ }
+ } else {
+found1:
+ if (s->mode_ext & MODE_EXT_MS_STEREO) {
+ /* lower part of the spectrum : do ms stereo
+ if enabled */
+ for (j = 0; j < len; j++) {
+ tmp0 = tab0[j];
+ tmp1 = tab1[j];
+ tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
+ tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
+ }
+ }
+ }
+ }
+ }
+
+ non_zero_found = non_zero_found_short[0] |
+ non_zero_found_short[1] |
+ non_zero_found_short[2];
+
+ for (i = g1->long_end - 1;i >= 0;i--) {
+ len = ff_band_size_long[s->sample_rate_index][i];
+ tab0 -= len;
+ tab1 -= len;
+ /* test if non zero band. if so, stop doing i-stereo */
+ if (!non_zero_found) {
+ for (j = 0; j < len; j++) {
+ if (tab1[j] != 0) {
+ non_zero_found = 1;
+ goto found2;
+ }
+ }
+ /* for last band, use previous scale factor */
+ k = (i == 21) ? 20 : i;
+ sf = g1->scale_factors[k];
+ if (sf >= sf_max)
+ goto found2;
+ v1 = is_tab[0][sf];
+ v2 = is_tab[1][sf];
+ for (j = 0; j < len; j++) {
+ tmp0 = tab0[j];
+ tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
+ tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
+ }
+ } else {
+found2:
+ if (s->mode_ext & MODE_EXT_MS_STEREO) {
+ /* lower part of the spectrum : do ms stereo
+ if enabled */
+ for (j = 0; j < len; j++) {
+ tmp0 = tab0[j];
+ tmp1 = tab1[j];
+ tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
+ tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
+ }
+ }
+ }
+ }
+ } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
+ /* ms stereo ONLY */
+ /* NOTE: the 1/sqrt(2) normalization factor is included in the
+ global gain */
+#if USE_FLOATS
+ s->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
+#else
+ tab0 = g0->sb_hybrid;
+ tab1 = g1->sb_hybrid;
+ for (i = 0; i < 576; i++) {
+ tmp0 = tab0[i];
+ tmp1 = tab1[i];
+ tab0[i] = tmp0 + tmp1;
+ tab1[i] = tmp0 - tmp1;
+ }
+#endif
+ }
+}
+
+#if USE_FLOATS
+#if HAVE_MIPSFPU
+# include "mips/compute_antialias_float.h"
+#endif /* HAVE_MIPSFPU */
+#else
+#if HAVE_MIPSDSP
+# include "mips/compute_antialias_fixed.h"
+#endif /* HAVE_MIPSDSP */
+#endif /* USE_FLOATS */
+
+#ifndef compute_antialias
+#if USE_FLOATS
+#define AA(j) do { \
+ float tmp0 = ptr[-1-j]; \
+ float tmp1 = ptr[ j]; \
+ ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
+ ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
+ } while (0)
+#else
+#define AA(j) do { \
+ SUINT tmp0 = ptr[-1-j]; \
+ SUINT tmp1 = ptr[ j]; \
+ SUINT tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
+ ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
+ ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
+ } while (0)
+#endif
+
+static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
+{
+ INTFLOAT *ptr;
+ int n, i;
+
+ /* we antialias only "long" bands */
+ if (g->block_type == 2) {
+ if (!g->switch_point)
+ return;
+ /* XXX: check this for 8000Hz case */
+ n = 1;
+ } else {
+ n = SBLIMIT - 1;
+ }
+
+ ptr = g->sb_hybrid + 18;
+ for (i = n; i > 0; i--) {
+ AA(0);
+ AA(1);
+ AA(2);
+ AA(3);
+ AA(4);
+ AA(5);
+ AA(6);
+ AA(7);
+
+ ptr += 18;
+ }
+}
+#endif /* compute_antialias */
+
+static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
+ INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
+{
+ INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
+ INTFLOAT out2[12];
+ int i, j, mdct_long_end, sblimit;
+
+ /* find last non zero block */
+ ptr = g->sb_hybrid + 576;
+ ptr1 = g->sb_hybrid + 2 * 18;
+ while (ptr >= ptr1) {
+ int32_t *p;
+ ptr -= 6;
+ p = (int32_t*)ptr;
+ if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
+ break;
+ }
+ sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
+
+ if (g->block_type == 2) {
+ /* XXX: check for 8000 Hz */
+ if (g->switch_point)
+ mdct_long_end = 2;
+ else
+ mdct_long_end = 0;
+ } else {
+ mdct_long_end = sblimit;
+ }
+
+ s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
+ mdct_long_end, g->switch_point,
+ g->block_type);
+
+ buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
+ ptr = g->sb_hybrid + 18 * mdct_long_end;
+
+ for (j = mdct_long_end; j < sblimit; j++) {
+ /* select frequency inversion */
+ win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
+ out_ptr = sb_samples + j;
+
+ for (i = 0; i < 6; i++) {
+ *out_ptr = buf[4*i];
+ out_ptr += SBLIMIT;
+ }
+ imdct12(out2, ptr + 0);
+ for (i = 0; i < 6; i++) {
+ *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
+ buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
+ out_ptr += SBLIMIT;
+ }
+ imdct12(out2, ptr + 1);
+ for (i = 0; i < 6; i++) {
+ *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
+ buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
+ out_ptr += SBLIMIT;
+ }
+ imdct12(out2, ptr + 2);
+ for (i = 0; i < 6; i++) {
+ buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
+ buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
+ buf[4*(i + 6*2)] = 0;
+ }
+ ptr += 18;
+ buf += (j&3) != 3 ? 1 : (4*18-3);
+ }
+ /* zero bands */
+ for (j = sblimit; j < SBLIMIT; j++) {
+ /* overlap */
+ out_ptr = sb_samples + j;
+ for (i = 0; i < 18; i++) {
+ *out_ptr = buf[4*i];
+ buf[4*i] = 0;
+ out_ptr += SBLIMIT;
+ }
+ buf += (j&3) != 3 ? 1 : (4*18-3);
+ }
+}
+
+/* main layer3 decoding function */
+static int mp_decode_layer3(MPADecodeContext *s)
+{
+ int nb_granules, main_data_begin;
+ int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
+ GranuleDef *g;
+ int16_t exponents[576]; //FIXME try INTFLOAT
+ int ret;
+
+ /* read side info */
+ if (s->lsf) {
+ ret = handle_crc(s, ((s->nb_channels == 1) ? 8*9 : 8*17));
+ main_data_begin = get_bits(&s->gb, 8);
+ skip_bits(&s->gb, s->nb_channels);
+ nb_granules = 1;
+ } else {
+ ret = handle_crc(s, ((s->nb_channels == 1) ? 8*17 : 8*32));
+ main_data_begin = get_bits(&s->gb, 9);
+ if (s->nb_channels == 2)
+ skip_bits(&s->gb, 3);
+ else
+ skip_bits(&s->gb, 5);
+ nb_granules = 2;
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
+ s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
+ }
+ }
+ if (ret < 0)
+ return ret;
+
+ for (gr = 0; gr < nb_granules; gr++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ ff_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
+ g = &s->granules[ch][gr];
+ g->part2_3_length = get_bits(&s->gb, 12);
+ g->big_values = get_bits(&s->gb, 9);
+ if (g->big_values > 288) {
+ av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ g->global_gain = get_bits(&s->gb, 8);
+ /* if MS stereo only is selected, we precompute the
+ 1/sqrt(2) renormalization factor */
+ if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
+ MODE_EXT_MS_STEREO)
+ g->global_gain -= 2;
+ if (s->lsf)
+ g->scalefac_compress = get_bits(&s->gb, 9);
+ else
+ g->scalefac_compress = get_bits(&s->gb, 4);
+ blocksplit_flag = get_bits1(&s->gb);
+ if (blocksplit_flag) {
+ g->block_type = get_bits(&s->gb, 2);
+ if (g->block_type == 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
+ return AVERROR_INVALIDDATA;
+ }
+ g->switch_point = get_bits1(&s->gb);
+ for (i = 0; i < 2; i++)
+ g->table_select[i] = get_bits(&s->gb, 5);
+ for (i = 0; i < 3; i++)
+ g->subblock_gain[i] = get_bits(&s->gb, 3);
+ init_short_region(s, g);
+ } else {
+ int region_address1, region_address2;
+ g->block_type = 0;
+ g->switch_point = 0;
+ for (i = 0; i < 3; i++)
+ g->table_select[i] = get_bits(&s->gb, 5);
+ /* compute huffman coded region sizes */
+ region_address1 = get_bits(&s->gb, 4);
+ region_address2 = get_bits(&s->gb, 3);
+ ff_dlog(s->avctx, "region1=%d region2=%d\n",
+ region_address1, region_address2);
+ init_long_region(s, g, region_address1, region_address2);
+ }
+ region_offset2size(g);
+ compute_band_indexes(s, g);
+
+ g->preflag = 0;
+ if (!s->lsf)
+ g->preflag = get_bits1(&s->gb);
+ g->scalefac_scale = get_bits1(&s->gb);
+ g->count1table_select = get_bits1(&s->gb);
+ ff_dlog(s->avctx, "block_type=%d switch_point=%d\n",
+ g->block_type, g->switch_point);
+ }
+ }
+
+ if (!s->adu_mode) {
+ int skip;
+ const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb) >> 3);
+ s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0,
+ FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
+ av_assert1((get_bits_count(&s->gb) & 7) == 0);
+ /* now we get bits from the main_data_begin offset */
+ ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
+ main_data_begin, s->last_buf_size);
+
+ memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize);
+ s->in_gb = s->gb;
+ init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8);
+ s->last_buf_size <<= 3;
+ for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ g = &s->granules[ch][gr];
+ s->last_buf_size += g->part2_3_length;
+ memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
+ compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
+ }
+ }
+ skip = s->last_buf_size - 8 * main_data_begin;
+ if (skip >= s->gb.size_in_bits - s->extrasize * 8 && s->in_gb.buffer) {
+ skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8);
+ s->gb = s->in_gb;
+ s->in_gb.buffer = NULL;
+ s->extrasize = 0;
+ } else {
+ skip_bits_long(&s->gb, skip);
+ }
+ } else {
+ gr = 0;
+ s->extrasize = 0;
+ }
+
+ for (; gr < nb_granules; gr++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ g = &s->granules[ch][gr];
+ bits_pos = get_bits_count(&s->gb);
+
+ if (!s->lsf) {
+ uint8_t *sc;
+ int slen, slen1, slen2;
+
+ /* MPEG-1 scale factors */
+ slen1 = ff_slen_table[0][g->scalefac_compress];
+ slen2 = ff_slen_table[1][g->scalefac_compress];
+ ff_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
+ if (g->block_type == 2) {
+ n = g->switch_point ? 17 : 18;
+ j = 0;
+ if (slen1) {
+ for (i = 0; i < n; i++)
+ g->scale_factors[j++] = get_bits(&s->gb, slen1);
+ } else {
+ for (i = 0; i < n; i++)
+ g->scale_factors[j++] = 0;
+ }
+ if (slen2) {
+ for (i = 0; i < 18; i++)
+ g->scale_factors[j++] = get_bits(&s->gb, slen2);
+ for (i = 0; i < 3; i++)
+ g->scale_factors[j++] = 0;
+ } else {
+ for (i = 0; i < 21; i++)
+ g->scale_factors[j++] = 0;
+ }
+ } else {
+ sc = s->granules[ch][0].scale_factors;
+ j = 0;
+ for (k = 0; k < 4; k++) {
+ n = k == 0 ? 6 : 5;
+ if ((g->scfsi & (0x8 >> k)) == 0) {
+ slen = (k < 2) ? slen1 : slen2;
+ if (slen) {
+ for (i = 0; i < n; i++)
+ g->scale_factors[j++] = get_bits(&s->gb, slen);
+ } else {
+ for (i = 0; i < n; i++)
+ g->scale_factors[j++] = 0;
+ }
+ } else {
+ /* simply copy from last granule */
+ for (i = 0; i < n; i++) {
+ g->scale_factors[j] = sc[j];
+ j++;
+ }
+ }
+ }
+ g->scale_factors[j++] = 0;
+ }
+ } else {
+ int tindex, tindex2, slen[4], sl, sf;
+
+ /* LSF scale factors */
+ if (g->block_type == 2)
+ tindex = g->switch_point ? 2 : 1;
+ else
+ tindex = 0;
+
+ sf = g->scalefac_compress;
+ if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
+ /* intensity stereo case */
+ sf >>= 1;
+ if (sf < 180) {
+ lsf_sf_expand(slen, sf, 6, 6, 0);
+ tindex2 = 3;
+ } else if (sf < 244) {
+ lsf_sf_expand(slen, sf - 180, 4, 4, 0);
+ tindex2 = 4;
+ } else {
+ lsf_sf_expand(slen, sf - 244, 3, 0, 0);
+ tindex2 = 5;
+ }
+ } else {
+ /* normal case */
+ if (sf < 400) {
+ lsf_sf_expand(slen, sf, 5, 4, 4);
+ tindex2 = 0;
+ } else if (sf < 500) {
+ lsf_sf_expand(slen, sf - 400, 5, 4, 0);
+ tindex2 = 1;
+ } else {
+ lsf_sf_expand(slen, sf - 500, 3, 0, 0);
+ tindex2 = 2;
+ g->preflag = 1;
+ }
+ }
+
+ j = 0;
+ for (k = 0; k < 4; k++) {
+ n = ff_lsf_nsf_table[tindex2][tindex][k];
+ sl = slen[k];
+ if (sl) {
+ for (i = 0; i < n; i++)
+ g->scale_factors[j++] = get_bits(&s->gb, sl);
+ } else {
+ for (i = 0; i < n; i++)
+ g->scale_factors[j++] = 0;
+ }
+ }
+ /* XXX: should compute exact size */
+ for (; j < 40; j++)
+ g->scale_factors[j] = 0;
+ }
+
+ exponents_from_scale_factors(s, g, exponents);
+
+ /* read Huffman coded residue */
+ huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
+ } /* ch */
+
+ if (s->mode == MPA_JSTEREO)
+ compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
+
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ g = &s->granules[ch][gr];
+
+ reorder_block(s, g);
+ compute_antialias(s, g);
+ compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
+ }
+ } /* gr */
+ if (get_bits_count(&s->gb) < 0)
+ skip_bits_long(&s->gb, -get_bits_count(&s->gb));
+ return nb_granules * 18;
+}
+
+static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
+ const uint8_t *buf, int buf_size)
+{
+ int i, nb_frames, ch, ret;
+ OUT_INT *samples_ptr;
+
+ init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
+ if (s->error_protection)
+ s->crc = get_bits(&s->gb, 16);
+
+ switch(s->layer) {
+ case 1:
+ s->avctx->frame_size = 384;
+ nb_frames = mp_decode_layer1(s);
+ break;
+ case 2:
+ s->avctx->frame_size = 1152;
+ nb_frames = mp_decode_layer2(s);
+ break;
+ case 3:
+ s->avctx->frame_size = s->lsf ? 576 : 1152;
+ default:
+ nb_frames = mp_decode_layer3(s);
+
+ s->last_buf_size=0;
+ if (s->in_gb.buffer) {
+ align_get_bits(&s->gb);
+ i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
+ if (i >= 0 && i <= BACKSTEP_SIZE) {
+ memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb) >> 3), i);
+ s->last_buf_size=i;
+ } else
+ av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
+ s->gb = s->in_gb;
+ s->in_gb.buffer = NULL;
+ s->extrasize = 0;
+ }
+
+ align_get_bits(&s->gb);
+ av_assert1((get_bits_count(&s->gb) & 7) == 0);
+ i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
+ if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
+ if (i < 0)
+ av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
+ i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
+ }
+ av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
+ memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
+ s->last_buf_size += i;
+ }
+
+ if(nb_frames < 0)
+ return nb_frames;
+
+ /* get output buffer */
+ if (!samples) {
+ av_assert0(s->frame);
+ s->frame->nb_samples = s->avctx->frame_size;
+ if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
+ return ret;
+ samples = (OUT_INT **)s->frame->extended_data;
+ }
+
+ /* apply the synthesis filter */
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ int sample_stride;
+ if (s->avctx->sample_fmt == OUT_FMT_P) {
+ samples_ptr = samples[ch];
+ sample_stride = 1;
+ } else {
+ samples_ptr = samples[0] + ch;
+ sample_stride = s->nb_channels;
+ }
+ for (i = 0; i < nb_frames; i++) {
+ RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
+ &(s->synth_buf_offset[ch]),
+ RENAME(ff_mpa_synth_window),
+ &s->dither_state, samples_ptr,
+ sample_stride, s->sb_samples[ch][i]);
+ samples_ptr += 32 * sample_stride;
+ }
+ }
+
+ return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
+}
+
+static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ MPADecodeContext *s = avctx->priv_data;
+ uint32_t header;
+ int ret;
+
+ int skipped = 0;
+ while(buf_size && !*buf){
+ buf++;
+ buf_size--;
+ skipped++;
+ }
+
+ if (buf_size < HEADER_SIZE)
+ return AVERROR_INVALIDDATA;
+
+ header = AV_RB32(buf);
+ if (header >> 8 == AV_RB32("TAG") >> 8) {
+ av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
+ return buf_size + skipped;
+ }
+ ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Header missing\n");
+ return AVERROR_INVALIDDATA;
+ } else if (ret == 1) {
+ /* free format: prepare to compute frame size */
+ s->frame_size = -1;
+ return AVERROR_INVALIDDATA;
+ }
+ /* update codec info */
+ av_channel_layout_uninit(&avctx->ch_layout);
+ avctx->ch_layout = s->nb_channels == 1 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO :
+ (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
+ if (!avctx->bit_rate)
+ avctx->bit_rate = s->bit_rate;
+
+ if (s->frame_size <= 0) {
+ av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
+ return AVERROR_INVALIDDATA;
+ } else if (s->frame_size < buf_size) {
+ av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
+ buf_size= s->frame_size;
+ }
+
+ s->frame = frame;
+
+ ret = mp_decode_frame(s, NULL, buf, buf_size);
+ if (ret >= 0) {
+ s->frame->nb_samples = avctx->frame_size;
+ *got_frame_ptr = 1;
+ avctx->sample_rate = s->sample_rate;
+ //FIXME maybe move the other codec info stuff from above here too
+ } else {
+ av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
+ /* Only return an error if the bad frame makes up the whole packet or
+ * the error is related to buffer management.
+ * If there is more data in the packet, just consume the bad frame
+ * instead of returning an error, which would discard the whole
+ * packet. */
+ *got_frame_ptr = 0;
+ if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
+ return ret;
+ }
+ s->frame_size = 0;
+ return buf_size + skipped;
+}
+
+static void mp_flush(MPADecodeContext *ctx)
+{
+ memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
+ memset(ctx->mdct_buf, 0, sizeof(ctx->mdct_buf));
+ ctx->last_buf_size = 0;
+ ctx->dither_state = 0;
+}
+
+static void flush(AVCodecContext *avctx)
+{
+ mp_flush(avctx->priv_data);
+}
+
+#if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
+static int decode_frame_adu(AVCodecContext *avctx, AVFrame *frame,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ MPADecodeContext *s = avctx->priv_data;
+ uint32_t header;
+ int len, ret;
+
+ len = buf_size;
+
+ // Discard too short frames
+ if (buf_size < HEADER_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+
+ if (len > MPA_MAX_CODED_FRAME_SIZE)
+ len = MPA_MAX_CODED_FRAME_SIZE;
+
+ // Get header and restore sync word
+ header = AV_RB32(buf) | 0xffe00000;
+
+ ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
+ return ret;
+ }
+ /* update codec info */
+ avctx->sample_rate = s->sample_rate;
+ av_channel_layout_uninit(&avctx->ch_layout);
+ avctx->ch_layout = s->nb_channels == 1 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO :
+ (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
+ if (!avctx->bit_rate)
+ avctx->bit_rate = s->bit_rate;
+
+ s->frame_size = len;
+
+ s->frame = frame;
+
+ ret = mp_decode_frame(s, NULL, buf, buf_size);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
+ return ret;
+ }
+
+ *got_frame_ptr = 1;
+
+ return buf_size;
+}
+#endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
+
+#if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
+
+/**
+ * Context for MP3On4 decoder
+ */
+typedef struct MP3On4DecodeContext {
+ int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
+ int syncword; ///< syncword patch
+ const uint8_t *coff; ///< channel offsets in output buffer
+ MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
+} MP3On4DecodeContext;
+
+#include "mpeg4audio.h"
+
+/* Next 3 arrays are indexed by channel config number (passed via codecdata) */
+
+/* number of mp3 decoder instances */
+static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
+
+/* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
+static const uint8_t chan_offset[8][5] = {
+ { 0 },
+ { 0 }, // C
+ { 0 }, // FLR
+ { 2, 0 }, // C FLR
+ { 2, 0, 3 }, // C FLR BS
+ { 2, 0, 3 }, // C FLR BLRS
+ { 2, 0, 4, 3 }, // C FLR BLRS LFE
+ { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
+};
+
+/* mp3on4 channel layouts */
+static const int16_t chan_layout[8] = {
+ 0,
+ AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ AV_CH_LAYOUT_SURROUND,
+ AV_CH_LAYOUT_4POINT0,
+ AV_CH_LAYOUT_5POINT0,
+ AV_CH_LAYOUT_5POINT1,
+ AV_CH_LAYOUT_7POINT1
+};
+
+static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
+{
+ MP3On4DecodeContext *s = avctx->priv_data;
+ int i;
+
+ for (i = 0; i < s->frames; i++)
+ av_freep(&s->mp3decctx[i]);
+
+ return 0;
+}
+
+
+static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
+{
+ MP3On4DecodeContext *s = avctx->priv_data;
+ MPEG4AudioConfig cfg;
+ int i, ret;
+
+ if ((avctx->extradata_size < 2) || !avctx->extradata) {
+ av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ avpriv_mpeg4audio_get_config2(&cfg, avctx->extradata,
+ avctx->extradata_size, 1, avctx);
+ if (!cfg.chan_config || cfg.chan_config > 7) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
+ return AVERROR_INVALIDDATA;
+ }
+ s->frames = mp3Frames[cfg.chan_config];
+ s->coff = chan_offset[cfg.chan_config];
+ av_channel_layout_uninit(&avctx->ch_layout);
+ av_channel_layout_from_mask(&avctx->ch_layout, chan_layout[cfg.chan_config]);
+
+ if (cfg.sample_rate < 16000)
+ s->syncword = 0xffe00000;
+ else
+ s->syncword = 0xfff00000;
+
+ /* Init the first mp3 decoder in standard way, so that all tables get builded
+ * We replace avctx->priv_data with the context of the first decoder so that
+ * decode_init() does not have to be changed.
+ * Other decoders will be initialized here copying data from the first context
+ */
+ // Allocate zeroed memory for the first decoder context
+ s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
+ if (!s->mp3decctx[0])
+ return AVERROR(ENOMEM);
+ // Put decoder context in place to make init_decode() happy
+ avctx->priv_data = s->mp3decctx[0];
+ ret = decode_init(avctx);
+ // Restore mp3on4 context pointer
+ avctx->priv_data = s;
+ if (ret < 0)
+ return ret;
+ s->mp3decctx[0]->adu_mode = 1; // Set adu mode
+
+ /* Create a separate codec/context for each frame (first is already ok).
+ * Each frame is 1 or 2 channels - up to 5 frames allowed
+ */
+ for (i = 1; i < s->frames; i++) {
+ s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
+ if (!s->mp3decctx[i])
+ return AVERROR(ENOMEM);
+ s->mp3decctx[i]->adu_mode = 1;
+ s->mp3decctx[i]->avctx = avctx;
+ s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
+ s->mp3decctx[i]->butterflies_float = s->mp3decctx[0]->butterflies_float;
+ }
+
+ return 0;
+}
+
+
+static void flush_mp3on4(AVCodecContext *avctx)
+{
+ int i;
+ MP3On4DecodeContext *s = avctx->priv_data;
+
+ for (i = 0; i < s->frames; i++)
+ mp_flush(s->mp3decctx[i]);
+}
+
+
+static int decode_frame_mp3on4(AVCodecContext *avctx, AVFrame *frame,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ MP3On4DecodeContext *s = avctx->priv_data;
+ MPADecodeContext *m;
+ int fsize, len = buf_size, out_size = 0;
+ uint32_t header;
+ OUT_INT **out_samples;
+ OUT_INT *outptr[2];
+ int fr, ch, ret;
+
+ /* get output buffer */
+ frame->nb_samples = MPA_FRAME_SIZE;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+ return ret;
+ out_samples = (OUT_INT **)frame->extended_data;
+
+ // Discard too short frames
+ if (buf_size < HEADER_SIZE)
+ return AVERROR_INVALIDDATA;
+
+ avctx->bit_rate = 0;
+
+ ch = 0;
+ for (fr = 0; fr < s->frames; fr++) {
+ fsize = AV_RB16(buf) >> 4;
+ fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
+ m = s->mp3decctx[fr];
+ av_assert1(m);
+
+ if (fsize < HEADER_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
+ return AVERROR_INVALIDDATA;
+ }
+ header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
+
+ ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (ch + m->nb_channels > avctx->ch_layout.nb_channels ||
+ s->coff[fr] + m->nb_channels > avctx->ch_layout.nb_channels) {
+ av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
+ "channel count\n");
+ return AVERROR_INVALIDDATA;
+ }
+ ch += m->nb_channels;
+
+ outptr[0] = out_samples[s->coff[fr]];
+ if (m->nb_channels > 1)
+ outptr[1] = out_samples[s->coff[fr] + 1];
+
+ if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "failed to decode channel %d\n", ch);
+ memset(outptr[0], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
+ if (m->nb_channels > 1)
+ memset(outptr[1], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
+ ret = m->nb_channels * MPA_FRAME_SIZE*sizeof(OUT_INT);
+ }
+
+ out_size += ret;
+ buf += fsize;
+ len -= fsize;
+
+ avctx->bit_rate += m->bit_rate;
+ }
+ if (ch != avctx->ch_layout.nb_channels) {
+ av_log(avctx, AV_LOG_ERROR, "failed to decode all channels\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* update codec info */
+ avctx->sample_rate = s->mp3decctx[0]->sample_rate;
+
+ frame->nb_samples = out_size / (avctx->ch_layout.nb_channels * sizeof(OUT_INT));
+ *got_frame_ptr = 1;
+
+ return buf_size;
+}
+#endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-27 14:26:02 +0000