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Diffstat (limited to 'ffmpeg/libavcodec/atrac3.c')
| -rw-r--r-- | ffmpeg/libavcodec/atrac3.c | 1015 |
1 files changed, 1015 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/atrac3.c b/ffmpeg/libavcodec/atrac3.c new file mode 100644 index 0000000..a9e98f8 --- /dev/null +++ b/ffmpeg/libavcodec/atrac3.c @@ -0,0 +1,1015 @@ +/* + * 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 "internal.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 { + 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); + if (off) + c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8)))); + else + c = av_be2ne32(0x537F6103U); + bytes += 3 + off; + for (i = 0; i < bytes / 4; i++) + output[i] = c ^ buf[i]; + + if (off) + avpriv_request_sample(NULL, "Offset of %d", 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[j] += input[j]; + } + + 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_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1); + + /* 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) +{ + AVFrame *frame = data; + 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 */ + frame->nb_samples = SAMPLES_PER_FRAME; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + 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 **)frame->extended_data); + if (ret) { + av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n"); + return ret; + } + + *got_frame_ptr = 1; + + 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 == 12 || 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); + } + + 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 }, +}; |