/* * Atrac 3 compatible decoder * Copyright (c) 2006-2008 Maxim Poliakovski * Copyright (c) 2006-2008 Benjamin Larsson * * 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 * Atrac 3 compatible decoder. * This decoder handles Sony's ATRAC3 data. * * Container formats used to store atrac 3 data: * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3). * * To use this decoder, a calling application must supply the extradata * bytes provided in the containers above. */ #include <math.h> #include <stddef.h> #include <stdio.h> #include "libavutil/float_dsp.h" #include "libavutil/libm.h" #include "avcodec.h" #include "bytestream.h" #include "fft.h" #include "fmtconvert.h" #include "get_bits.h" #include "atrac.h" #include "atrac3data.h" #define JOINT_STEREO 0x12 #define STEREO 0x2 #define SAMPLES_PER_FRAME 1024 #define MDCT_SIZE 512 typedef struct GainInfo { int num_gain_data; int lev_code[8]; int loc_code[8]; } GainInfo; typedef struct GainBlock { GainInfo g_block[4]; } GainBlock; typedef struct TonalComponent { int pos; int num_coefs; float coef[8]; } TonalComponent; typedef struct ChannelUnit { int bands_coded; int num_components; float prev_frame[SAMPLES_PER_FRAME]; int gc_blk_switch; TonalComponent components[64]; GainBlock gain_block[2]; DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME]; DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME]; float delay_buf1[46]; ///<qmf delay buffers float delay_buf2[46]; float delay_buf3[46]; } ChannelUnit; typedef struct ATRAC3Context { AVFrame frame; GetBitContext gb; //@{ /** stream data */ int coding_mode; ChannelUnit *units; //@} //@{ /** joint-stereo related variables */ int matrix_coeff_index_prev[4]; int matrix_coeff_index_now[4]; int matrix_coeff_index_next[4]; int weighting_delay[6]; //@} //@{ /** data buffers */ uint8_t *decoded_bytes_buffer; float temp_buf[1070]; //@} //@{ /** extradata */ int scrambled_stream; //@} FFTContext mdct_ctx; FmtConvertContext fmt_conv; AVFloatDSPContext fdsp; } ATRAC3Context; static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE]; static VLC_TYPE atrac3_vlc_table[4096][2]; static VLC spectral_coeff_tab[7]; static float gain_tab1[16]; static float gain_tab2[31]; /** * Regular 512 points IMDCT without overlapping, with the exception of the * swapping of odd bands caused by the reverse spectra of the QMF. * * @param odd_band 1 if the band is an odd band */ static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band) { int i; if (odd_band) { /** * Reverse the odd bands before IMDCT, this is an effect of the QMF * transform or it gives better compression to do it this way. * FIXME: It should be possible to handle this in imdct_calc * for that to happen a modification of the prerotation step of * all SIMD code and C code is needed. * Or fix the functions before so they generate a pre reversed spectrum. */ for (i = 0; i < 128; i++) FFSWAP(float, input[i], input[255 - i]); } q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input); /* Perform windowing on the output. */ q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE); } /* * indata descrambling, only used for data coming from the rm container */ static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes) { int i, off; uint32_t c; const uint32_t *buf; uint32_t *output = (uint32_t *)out; off = (intptr_t)input & 3; buf = (const uint32_t *)(input - off); c = av_be2ne32((0x537F6103 >> (off * 8)) | (0x537F6103 << (32 - (off * 8)))); bytes += 3 + off; for (i = 0; i < bytes / 4; i++) output[i] = c ^ buf[i]; if (off) av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off); return off; } static av_cold void init_atrac3_window(void) { int i, j; /* generate the mdct window, for details see * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ for (i = 0, j = 255; i < 128; i++, j--) { float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; float w = 0.5 * (wi * wi + wj * wj); mdct_window[i] = mdct_window[511 - i] = wi / w; mdct_window[j] = mdct_window[511 - j] = wj / w; } } static av_cold int atrac3_decode_close(AVCodecContext *avctx) { ATRAC3Context *q = avctx->priv_data; av_free(q->units); av_free(q->decoded_bytes_buffer); ff_mdct_end(&q->mdct_ctx); return 0; } /** * Mantissa decoding * * @param selector which table the output values are coded with * @param coding_flag constant length coding or variable length coding * @param mantissas mantissa output table * @param num_codes number of values to get */ static void read_quant_spectral_coeffs(GetBitContext *gb, int selector, int coding_flag, int *mantissas, int num_codes) { int i, code, huff_symb; if (selector == 1) num_codes /= 2; if (coding_flag != 0) { /* constant length coding (CLC) */ int num_bits = clc_length_tab[selector]; if (selector > 1) { for (i = 0; i < num_codes; i++) { if (num_bits) code = get_sbits(gb, num_bits); else code = 0; mantissas[i] = code; } } else { for (i = 0; i < num_codes; i++) { if (num_bits) code = get_bits(gb, num_bits); // num_bits is always 4 in this case else code = 0; mantissas[i * 2 ] = mantissa_clc_tab[code >> 2]; mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3]; } } } else { /* variable length coding (VLC) */ if (selector != 1) { for (i = 0; i < num_codes; i++) { huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3); huff_symb += 1; code = huff_symb >> 1; if (huff_symb & 1) code = -code; mantissas[i] = code; } } else { for (i = 0; i < num_codes; i++) { huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table, spectral_coeff_tab[selector - 1].bits, 3); mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ]; mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1]; } } } } /** * Restore the quantized band spectrum coefficients * * @return subband count, fix for broken specification/files */ static int decode_spectrum(GetBitContext *gb, float *output) { int num_subbands, coding_mode, i, j, first, last, subband_size; int subband_vlc_index[32], sf_index[32]; int mantissas[128]; float scale_factor; num_subbands = get_bits(gb, 5); // number of coded subbands coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC /* get the VLC selector table for the subbands, 0 means not coded */ for (i = 0; i <= num_subbands; i++) subband_vlc_index[i] = get_bits(gb, 3); /* read the scale factor indexes from the stream */ for (i = 0; i <= num_subbands; i++) { if (subband_vlc_index[i] != 0) sf_index[i] = get_bits(gb, 6); } for (i = 0; i <= num_subbands; i++) { first = subband_tab[i ]; last = subband_tab[i + 1]; subband_size = last - first; if (subband_vlc_index[i] != 0) { /* decode spectral coefficients for this subband */ /* TODO: This can be done faster is several blocks share the * same VLC selector (subband_vlc_index) */ read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode, mantissas, subband_size); /* decode the scale factor for this subband */ scale_factor = ff_atrac_sf_table[sf_index[i]] * inv_max_quant[subband_vlc_index[i]]; /* inverse quantize the coefficients */ for (j = 0; first < last; first++, j++) output[first] = mantissas[j] * scale_factor; } else { /* this subband was not coded, so zero the entire subband */ memset(output + first, 0, subband_size * sizeof(*output)); } } /* clear the subbands that were not coded */ first = subband_tab[i]; memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output)); return num_subbands; } /** * Restore the quantized tonal components * * @param components tonal components * @param num_bands number of coded bands */ static int decode_tonal_components(GetBitContext *gb, TonalComponent *components, int num_bands) { int i, b, c, m; int nb_components, coding_mode_selector, coding_mode; int band_flags[4], mantissa[8]; int component_count = 0; nb_components = get_bits(gb, 5); /* no tonal components */ if (nb_components == 0) return 0; coding_mode_selector = get_bits(gb, 2); if (coding_mode_selector == 2) return AVERROR_INVALIDDATA; coding_mode = coding_mode_selector & 1; for (i = 0; i < nb_components; i++) { int coded_values_per_component, quant_step_index; for (b = 0; b <= num_bands; b++) band_flags[b] = get_bits1(gb); coded_values_per_component = get_bits(gb, 3); quant_step_index = get_bits(gb, 3); if (quant_step_index <= 1) return AVERROR_INVALIDDATA; if (coding_mode_selector == 3) coding_mode = get_bits1(gb); for (b = 0; b < (num_bands + 1) * 4; b++) { int coded_components; if (band_flags[b >> 2] == 0) continue; coded_components = get_bits(gb, 3); for (c = 0; c < coded_components; c++) { TonalComponent *cmp = &components[component_count]; int sf_index, coded_values, max_coded_values; float scale_factor; sf_index = get_bits(gb, 6); if (component_count >= 64) return AVERROR_INVALIDDATA; cmp->pos = b * 64 + get_bits(gb, 6); max_coded_values = SAMPLES_PER_FRAME - cmp->pos; coded_values = coded_values_per_component + 1; coded_values = FFMIN(max_coded_values, coded_values); scale_factor = ff_atrac_sf_table[sf_index] * inv_max_quant[quant_step_index]; read_quant_spectral_coeffs(gb, quant_step_index, coding_mode, mantissa, coded_values); cmp->num_coefs = coded_values; /* inverse quant */ for (m = 0; m < coded_values; m++) cmp->coef[m] = mantissa[m] * scale_factor; component_count++; } } } return component_count; } /** * Decode gain parameters for the coded bands * * @param block the gainblock for the current band * @param num_bands amount of coded bands */ static int decode_gain_control(GetBitContext *gb, GainBlock *block, int num_bands) { int i, cf, num_data; int *level, *loc; GainInfo *gain = block->g_block; for (i = 0; i <= num_bands; i++) { num_data = get_bits(gb, 3); gain[i].num_gain_data = num_data; level = gain[i].lev_code; loc = gain[i].loc_code; for (cf = 0; cf < gain[i].num_gain_data; cf++) { level[cf] = get_bits(gb, 4); loc [cf] = get_bits(gb, 5); if (cf && loc[cf] <= loc[cf - 1]) return AVERROR_INVALIDDATA; } } /* Clear the unused blocks. */ for (; i < 4 ; i++) gain[i].num_gain_data = 0; return 0; } /** * Apply gain parameters and perform the MDCT overlapping part * * @param input input buffer * @param prev previous buffer to perform overlap against * @param output output buffer * @param gain1 current band gain info * @param gain2 next band gain info */ static void gain_compensate_and_overlap(float *input, float *prev, float *output, GainInfo *gain1, GainInfo *gain2) { float g1, g2, gain_inc; int i, j, num_data, start_loc, end_loc; if (gain2->num_gain_data == 0) g1 = 1.0; else g1 = gain_tab1[gain2->lev_code[0]]; if (gain1->num_gain_data == 0) { for (i = 0; i < 256; i++) output[i] = input[i] * g1 + prev[i]; } else { num_data = gain1->num_gain_data; gain1->loc_code[num_data] = 32; gain1->lev_code[num_data] = 4; for (i = 0, j = 0; i < num_data; i++) { start_loc = gain1->loc_code[i] * 8; end_loc = start_loc + 8; g2 = gain_tab1[gain1->lev_code[i]]; gain_inc = gain_tab2[gain1->lev_code[i + 1] - gain1->lev_code[i ] + 15]; /* interpolate */ for (; j < start_loc; j++) output[j] = (input[j] * g1 + prev[j]) * g2; /* interpolation is done over eight samples */ for (; j < end_loc; j++) { output[j] = (input[j] * g1 + prev[j]) * g2; g2 *= gain_inc; } } for (; j < 256; j++) output[j] = input[j] * g1 + prev[j]; } /* Delay for the overlapping part. */ memcpy(prev, &input[256], 256 * sizeof(*prev)); } /** * Combine the tonal band spectrum and regular band spectrum * * @param spectrum output spectrum buffer * @param num_components number of tonal components * @param components tonal components for this band * @return position of the last tonal coefficient */ static int add_tonal_components(float *spectrum, int num_components, TonalComponent *components) { int i, j, last_pos = -1; float *input, *output; for (i = 0; i < num_components; i++) { last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos); input = components[i].coef; output = &spectrum[components[i].pos]; for (j = 0; j < components[i].num_coefs; j++) output[i] += input[i]; } return last_pos; } #define INTERPOLATE(old, new, nsample) \ ((old) + (nsample) * 0.125 * ((new) - (old))) static void reverse_matrixing(float *su1, float *su2, int *prev_code, int *curr_code) { int i, nsample, band; float mc1_l, mc1_r, mc2_l, mc2_r; for (i = 0, band = 0; band < 4 * 256; band += 256, i++) { int s1 = prev_code[i]; int s2 = curr_code[i]; nsample = band; if (s1 != s2) { /* Selector value changed, interpolation needed. */ mc1_l = matrix_coeffs[s1 * 2 ]; mc1_r = matrix_coeffs[s1 * 2 + 1]; mc2_l = matrix_coeffs[s2 * 2 ]; mc2_r = matrix_coeffs[s2 * 2 + 1]; /* Interpolation is done over the first eight samples. */ for (; nsample < band + 8; nsample++) { float c1 = su1[nsample]; float c2 = su2[nsample]; c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) + c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band); su1[nsample] = c2; su2[nsample] = c1 * 2.0 - c2; } } /* Apply the matrix without interpolation. */ switch (s2) { case 0: /* M/S decoding */ for (; nsample < band + 256; nsample++) { float c1 = su1[nsample]; float c2 = su2[nsample]; su1[nsample] = c2 * 2.0; su2[nsample] = (c1 - c2) * 2.0; } break; case 1: for (; nsample < band + 256; nsample++) { float c1 = su1[nsample]; float c2 = su2[nsample]; su1[nsample] = (c1 + c2) * 2.0; su2[nsample] = c2 * -2.0; } break; case 2: case 3: for (; nsample < band + 256; nsample++) { float c1 = su1[nsample]; float c2 = su2[nsample]; su1[nsample] = c1 + c2; su2[nsample] = c1 - c2; } break; default: av_assert1(0); } } } static void get_channel_weights(int index, int flag, float ch[2]) { if (index == 7) { ch[0] = 1.0; ch[1] = 1.0; } else { ch[0] = (index & 7) / 7.0; ch[1] = sqrt(2 - ch[0] * ch[0]); if (flag) FFSWAP(float, ch[0], ch[1]); } } static void channel_weighting(float *su1, float *su2, int *p3) { int band, nsample; /* w[x][y] y=0 is left y=1 is right */ float w[2][2]; if (p3[1] != 7 || p3[3] != 7) { get_channel_weights(p3[1], p3[0], w[0]); get_channel_weights(p3[3], p3[2], w[1]); for (band = 256; band < 4 * 256; band += 256) { for (nsample = band; nsample < band + 8; nsample++) { su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band); su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band); } for(; nsample < band + 256; nsample++) { su1[nsample] *= w[1][0]; su2[nsample] *= w[1][1]; } } } } /** * Decode a Sound Unit * * @param snd the channel unit to be used * @param output the decoded samples before IQMF in float representation * @param channel_num channel number * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono) */ static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb, ChannelUnit *snd, float *output, int channel_num, int coding_mode) { int band, ret, num_subbands, last_tonal, num_bands; GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch]; GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch]; if (coding_mode == JOINT_STEREO && channel_num == 1) { if (get_bits(gb, 2) != 3) { av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n"); return AVERROR_INVALIDDATA; } } else { if (get_bits(gb, 6) != 0x28) { av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n"); return AVERROR_INVALIDDATA; } } /* number of coded QMF bands */ snd->bands_coded = get_bits(gb, 2); ret = decode_gain_control(gb, gain2, snd->bands_coded); if (ret) return ret; snd->num_components = decode_tonal_components(gb, snd->components, snd->bands_coded); if (snd->num_components == -1) return -1; num_subbands = decode_spectrum(gb, snd->spectrum); /* Merge the decoded spectrum and tonal components. */ last_tonal = add_tonal_components(snd->spectrum, snd->num_components, snd->components); /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */ num_bands = (subband_tab[num_subbands] - 1) >> 8; if (last_tonal >= 0) num_bands = FFMAX((last_tonal + 256) >> 8, num_bands); /* Reconstruct time domain samples. */ for (band = 0; band < 4; band++) { /* Perform the IMDCT step without overlapping. */ if (band <= num_bands) imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1); else memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf)); /* gain compensation and overlapping */ gain_compensate_and_overlap(snd->imdct_buf, &snd->prev_frame[band * 256], &output[band * 256], &gain1->g_block[band], &gain2->g_block[band]); } /* Swap the gain control buffers for the next frame. */ snd->gc_blk_switch ^= 1; return 0; } static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf, float **out_samples) { ATRAC3Context *q = avctx->priv_data; int ret, i; uint8_t *ptr1; if (q->coding_mode == JOINT_STEREO) { /* channel coupling mode */ /* decode Sound Unit 1 */ init_get_bits(&q->gb, databuf, avctx->block_align * 8); ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0, JOINT_STEREO); if (ret != 0) return ret; /* Framedata of the su2 in the joint-stereo mode is encoded in * reverse byte order so we need to swap it first. */ if (databuf == q->decoded_bytes_buffer) { uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1; ptr1 = q->decoded_bytes_buffer; for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--) FFSWAP(uint8_t, *ptr1, *ptr2); } else { const uint8_t *ptr2 = databuf + avctx->block_align - 1; for (i = 0; i < avctx->block_align; i++) q->decoded_bytes_buffer[i] = *ptr2--; } /* Skip the sync codes (0xF8). */ ptr1 = q->decoded_bytes_buffer; for (i = 4; *ptr1 == 0xF8; i++, ptr1++) { if (i >= avctx->block_align) return AVERROR_INVALIDDATA; } /* set the bitstream reader at the start of the second Sound Unit*/ init_get_bits(&q->gb, ptr1, avctx->block_align * 8); /* Fill the Weighting coeffs delay buffer */ memmove(q->weighting_delay, &q->weighting_delay[2], 4 * sizeof(*q->weighting_delay)); q->weighting_delay[4] = get_bits1(&q->gb); q->weighting_delay[5] = get_bits(&q->gb, 3); for (i = 0; i < 4; i++) { q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i]; q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i]; q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2); } /* Decode Sound Unit 2. */ ret = decode_channel_sound_unit(q, &q->gb, &q->units[1], out_samples[1], 1, JOINT_STEREO); if (ret != 0) return ret; /* Reconstruct the channel coefficients. */ reverse_matrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now); channel_weighting(out_samples[0], out_samples[1], q->weighting_delay); } else { /* normal stereo mode or mono */ /* Decode the channel sound units. */ for (i = 0; i < avctx->channels; i++) { /* Set the bitstream reader at the start of a channel sound unit. */ init_get_bits(&q->gb, databuf + i * avctx->block_align / avctx->channels, avctx->block_align * 8 / avctx->channels); ret = decode_channel_sound_unit(q, &q->gb, &q->units[i], out_samples[i], i, q->coding_mode); if (ret != 0) return ret; } } /* Apply the iQMF synthesis filter. */ for (i = 0; i < avctx->channels; i++) { float *p1 = out_samples[i]; float *p2 = p1 + 256; float *p3 = p2 + 256; float *p4 = p3 + 256; ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf); ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf); ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf); } return 0; } static int atrac3_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; ATRAC3Context *q = avctx->priv_data; int ret; const uint8_t *databuf; if (buf_size < avctx->block_align) { av_log(avctx, AV_LOG_ERROR, "Frame too small (%d bytes). Truncated file?\n", buf_size); return AVERROR_INVALIDDATA; } /* get output buffer */ q->frame.nb_samples = SAMPLES_PER_FRAME; if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } /* Check if we need to descramble and what buffer to pass on. */ if (q->scrambled_stream) { decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align); databuf = q->decoded_bytes_buffer; } else { databuf = buf; } ret = decode_frame(avctx, databuf, (float **)q->frame.extended_data); if (ret) { av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n"); return ret; } *got_frame_ptr = 1; *(AVFrame *)data = q->frame; return avctx->block_align; } static void atrac3_init_static_data(void) { int i; init_atrac3_window(); ff_atrac_generate_tables(); /* Initialize the VLC tables. */ for (i = 0; i < 7; i++) { spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]]; spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i ]; init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i], huff_bits[i], 1, 1, huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC); } /* Generate gain tables */ for (i = 0; i < 16; i++) gain_tab1[i] = exp2f (4 - i); for (i = -15; i < 16; i++) gain_tab2[i + 15] = exp2f (i * -0.125); } static av_cold int atrac3_decode_init(AVCodecContext *avctx) { static int static_init_done; int i, ret; int version, delay, samples_per_frame, frame_factor; const uint8_t *edata_ptr = avctx->extradata; ATRAC3Context *q = avctx->priv_data; if (avctx->channels <= 0 || avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n"); return AVERROR(EINVAL); } if (!static_init_done) atrac3_init_static_data(); static_init_done = 1; /* Take care of the codec-specific extradata. */ if (avctx->extradata_size == 14) { /* Parse the extradata, WAV format */ av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n", bytestream_get_le16(&edata_ptr)); // Unknown value always 1 edata_ptr += 4; // samples per channel q->coding_mode = bytestream_get_le16(&edata_ptr); av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n", bytestream_get_le16(&edata_ptr)); //Dupe of coding mode frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n", bytestream_get_le16(&edata_ptr)); // Unknown always 0 /* setup */ samples_per_frame = SAMPLES_PER_FRAME * avctx->channels; version = 4; delay = 0x88E; q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO; q->scrambled_stream = 0; if (avctx->block_align != 96 * avctx->channels * frame_factor && avctx->block_align != 152 * avctx->channels * frame_factor && avctx->block_align != 192 * avctx->channels * frame_factor) { av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor " "configuration %d/%d/%d\n", avctx->block_align, avctx->channels, frame_factor); return AVERROR_INVALIDDATA; } } else if (avctx->extradata_size == 10) { /* Parse the extradata, RM format. */ version = bytestream_get_be32(&edata_ptr); samples_per_frame = bytestream_get_be16(&edata_ptr); delay = bytestream_get_be16(&edata_ptr); q->coding_mode = bytestream_get_be16(&edata_ptr); q->scrambled_stream = 1; } else { av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n", avctx->extradata_size); return AVERROR(EINVAL); } if (q->coding_mode == JOINT_STEREO && avctx->channels < 2) { av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n"); return AVERROR_INVALIDDATA; } /* Check the extradata */ if (version != 4) { av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version); return AVERROR_INVALIDDATA; } if (samples_per_frame != SAMPLES_PER_FRAME && samples_per_frame != SAMPLES_PER_FRAME * 2) { av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n", samples_per_frame); return AVERROR_INVALIDDATA; } if (delay != 0x88E) { av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n", delay); return AVERROR_INVALIDDATA; } if (q->coding_mode == STEREO) av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n"); else if (q->coding_mode == JOINT_STEREO) av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n"); else { av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n", q->coding_mode); return AVERROR_INVALIDDATA; } if (avctx->block_align >= UINT_MAX / 2) return AVERROR(EINVAL); q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) + FF_INPUT_BUFFER_PADDING_SIZE); if (q->decoded_bytes_buffer == NULL) return AVERROR(ENOMEM); avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; /* initialize the MDCT transform */ if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) { av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); av_freep(&q->decoded_bytes_buffer); return ret; } /* init the joint-stereo decoding data */ q->weighting_delay[0] = 0; q->weighting_delay[1] = 7; q->weighting_delay[2] = 0; q->weighting_delay[3] = 7; q->weighting_delay[4] = 0; q->weighting_delay[5] = 7; for (i = 0; i < 4; i++) { q->matrix_coeff_index_prev[i] = 3; q->matrix_coeff_index_now[i] = 3; q->matrix_coeff_index_next[i] = 3; } avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); ff_fmt_convert_init(&q->fmt_conv, avctx); q->units = av_mallocz(sizeof(*q->units) * avctx->channels); if (!q->units) { atrac3_decode_close(avctx); return AVERROR(ENOMEM); } avcodec_get_frame_defaults(&q->frame); avctx->coded_frame = &q->frame; return 0; } AVCodec ff_atrac3_decoder = { .name = "atrac3", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_ATRAC3, .priv_data_size = sizeof(ATRAC3Context), .init = atrac3_decode_init, .close = atrac3_decode_close, .decode = atrac3_decode_frame, .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, };