diff options
Diffstat (limited to 'ffmpeg1/libavcodec/atrac3.c')
| -rw-r--r-- | ffmpeg1/libavcodec/atrac3.c | 1015 |
1 files changed, 0 insertions, 1015 deletions
diff --git a/ffmpeg1/libavcodec/atrac3.c b/ffmpeg1/libavcodec/atrac3.c deleted file mode 100644 index a9e98f8..0000000 --- a/ffmpeg1/libavcodec/atrac3.c +++ /dev/null @@ -1,1015 +0,0 @@ -/* - * 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 }, -}; |