2 * ATRAC3 compatible decoder
3 * Copyright (c) 2006-2008 Maxim Poliakovski
4 * Copyright (c) 2006-2008 Benjamin Larsson
6 * This file is part of FFmpeg.
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 * ATRAC3 compatible decoder.
26 * This decoder handles Sony's ATRAC3 data.
28 * Container formats used to store ATRAC3 data:
29 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
31 * To use this decoder, a calling application must supply the extradata
32 * bytes provided in the containers above.
39 #include "libavutil/attributes.h"
40 #include "libavutil/float_dsp.h"
41 #include "libavutil/libm.h"
43 #include "bytestream.h"
45 #include "fmtconvert.h"
50 #include "atrac3data.h"
52 #define JOINT_STEREO 0x12
55 #define SAMPLES_PER_FRAME 1024
58 typedef struct GainBlock {
59 AtracGainInfo g_block[4];
62 typedef struct TonalComponent {
68 typedef struct ChannelUnit {
71 float prev_frame[SAMPLES_PER_FRAME];
73 TonalComponent components[64];
74 GainBlock gain_block[2];
76 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
77 DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
79 float delay_buf1[46]; ///<qmf delay buffers
84 typedef struct ATRAC3Context {
93 /** joint-stereo related variables */
94 int matrix_coeff_index_prev[4];
95 int matrix_coeff_index_now[4];
96 int matrix_coeff_index_next[4];
97 int weighting_delay[6];
101 uint8_t *decoded_bytes_buffer;
102 float temp_buf[1070];
106 int scrambled_stream;
109 AtracGCContext gainc_ctx;
111 FmtConvertContext fmt_conv;
112 AVFloatDSPContext fdsp;
115 static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
116 static VLC_TYPE atrac3_vlc_table[4096][2];
117 static VLC spectral_coeff_tab[7];
120 * Regular 512 points IMDCT without overlapping, with the exception of the
121 * swapping of odd bands caused by the reverse spectra of the QMF.
123 * @param odd_band 1 if the band is an odd band
125 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
131 * Reverse the odd bands before IMDCT, this is an effect of the QMF
132 * transform or it gives better compression to do it this way.
133 * FIXME: It should be possible to handle this in imdct_calc
134 * for that to happen a modification of the prerotation step of
135 * all SIMD code and C code is needed.
136 * Or fix the functions before so they generate a pre reversed spectrum.
138 for (i = 0; i < 128; i++)
139 FFSWAP(float, input[i], input[255 - i]);
142 q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
144 /* Perform windowing on the output. */
145 q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);
149 * indata descrambling, only used for data coming from the rm container
151 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
156 uint32_t *output = (uint32_t *)out;
158 off = (intptr_t)input & 3;
159 buf = (const uint32_t *)(input - off);
161 c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
163 c = av_be2ne32(0x537F6103U);
165 for (i = 0; i < bytes / 4; i++)
166 output[i] = c ^ buf[i];
169 avpriv_request_sample(NULL, "Offset of %d", off);
174 static av_cold void init_imdct_window(void)
178 /* generate the mdct window, for details see
179 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
180 for (i = 0, j = 255; i < 128; i++, j--) {
181 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
182 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
183 float w = 0.5 * (wi * wi + wj * wj);
184 mdct_window[i] = mdct_window[511 - i] = wi / w;
185 mdct_window[j] = mdct_window[511 - j] = wj / w;
189 static av_cold int atrac3_decode_close(AVCodecContext *avctx)
191 ATRAC3Context *q = avctx->priv_data;
194 av_freep(&q->decoded_bytes_buffer);
196 ff_mdct_end(&q->mdct_ctx);
204 * @param selector which table the output values are coded with
205 * @param coding_flag constant length coding or variable length coding
206 * @param mantissas mantissa output table
207 * @param num_codes number of values to get
209 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
210 int coding_flag, int *mantissas,
213 int i, code, huff_symb;
218 if (coding_flag != 0) {
219 /* constant length coding (CLC) */
220 int num_bits = clc_length_tab[selector];
223 for (i = 0; i < num_codes; i++) {
225 code = get_sbits(gb, num_bits);
231 for (i = 0; i < num_codes; i++) {
233 code = get_bits(gb, num_bits); // num_bits is always 4 in this case
236 mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
237 mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
241 /* variable length coding (VLC) */
243 for (i = 0; i < num_codes; i++) {
244 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
245 spectral_coeff_tab[selector-1].bits, 3);
247 code = huff_symb >> 1;
253 for (i = 0; i < num_codes; i++) {
254 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
255 spectral_coeff_tab[selector - 1].bits, 3);
256 mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
257 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
264 * Restore the quantized band spectrum coefficients
266 * @return subband count, fix for broken specification/files
268 static int decode_spectrum(GetBitContext *gb, float *output)
270 int num_subbands, coding_mode, i, j, first, last, subband_size;
271 int subband_vlc_index[32], sf_index[32];
275 num_subbands = get_bits(gb, 5); // number of coded subbands
276 coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
278 /* get the VLC selector table for the subbands, 0 means not coded */
279 for (i = 0; i <= num_subbands; i++)
280 subband_vlc_index[i] = get_bits(gb, 3);
282 /* read the scale factor indexes from the stream */
283 for (i = 0; i <= num_subbands; i++) {
284 if (subband_vlc_index[i] != 0)
285 sf_index[i] = get_bits(gb, 6);
288 for (i = 0; i <= num_subbands; i++) {
289 first = subband_tab[i ];
290 last = subband_tab[i + 1];
292 subband_size = last - first;
294 if (subband_vlc_index[i] != 0) {
295 /* decode spectral coefficients for this subband */
296 /* TODO: This can be done faster is several blocks share the
297 * same VLC selector (subband_vlc_index) */
298 read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
299 mantissas, subband_size);
301 /* decode the scale factor for this subband */
302 scale_factor = ff_atrac_sf_table[sf_index[i]] *
303 inv_max_quant[subband_vlc_index[i]];
305 /* inverse quantize the coefficients */
306 for (j = 0; first < last; first++, j++)
307 output[first] = mantissas[j] * scale_factor;
309 /* this subband was not coded, so zero the entire subband */
310 memset(output + first, 0, subband_size * sizeof(*output));
314 /* clear the subbands that were not coded */
315 first = subband_tab[i];
316 memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
321 * Restore the quantized tonal components
323 * @param components tonal components
324 * @param num_bands number of coded bands
326 static int decode_tonal_components(GetBitContext *gb,
327 TonalComponent *components, int num_bands)
330 int nb_components, coding_mode_selector, coding_mode;
331 int band_flags[4], mantissa[8];
332 int component_count = 0;
334 nb_components = get_bits(gb, 5);
336 /* no tonal components */
337 if (nb_components == 0)
340 coding_mode_selector = get_bits(gb, 2);
341 if (coding_mode_selector == 2)
342 return AVERROR_INVALIDDATA;
344 coding_mode = coding_mode_selector & 1;
346 for (i = 0; i < nb_components; i++) {
347 int coded_values_per_component, quant_step_index;
349 for (b = 0; b <= num_bands; b++)
350 band_flags[b] = get_bits1(gb);
352 coded_values_per_component = get_bits(gb, 3);
354 quant_step_index = get_bits(gb, 3);
355 if (quant_step_index <= 1)
356 return AVERROR_INVALIDDATA;
358 if (coding_mode_selector == 3)
359 coding_mode = get_bits1(gb);
361 for (b = 0; b < (num_bands + 1) * 4; b++) {
362 int coded_components;
364 if (band_flags[b >> 2] == 0)
367 coded_components = get_bits(gb, 3);
369 for (c = 0; c < coded_components; c++) {
370 TonalComponent *cmp = &components[component_count];
371 int sf_index, coded_values, max_coded_values;
374 sf_index = get_bits(gb, 6);
375 if (component_count >= 64)
376 return AVERROR_INVALIDDATA;
378 cmp->pos = b * 64 + get_bits(gb, 6);
380 max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
381 coded_values = coded_values_per_component + 1;
382 coded_values = FFMIN(max_coded_values, coded_values);
384 scale_factor = ff_atrac_sf_table[sf_index] *
385 inv_max_quant[quant_step_index];
387 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
388 mantissa, coded_values);
390 cmp->num_coefs = coded_values;
393 for (m = 0; m < coded_values; m++)
394 cmp->coef[m] = mantissa[m] * scale_factor;
401 return component_count;
405 * Decode gain parameters for the coded bands
407 * @param block the gainblock for the current band
408 * @param num_bands amount of coded bands
410 static int decode_gain_control(GetBitContext *gb, GainBlock *block,
416 AtracGainInfo *gain = block->g_block;
418 for (b = 0; b <= num_bands; b++) {
419 gain[b].num_points = get_bits(gb, 3);
420 level = gain[b].lev_code;
421 loc = gain[b].loc_code;
423 for (j = 0; j < gain[b].num_points; j++) {
424 level[j] = get_bits(gb, 4);
425 loc[j] = get_bits(gb, 5);
426 if (j && loc[j] <= loc[j - 1])
427 return AVERROR_INVALIDDATA;
431 /* Clear the unused blocks. */
433 gain[b].num_points = 0;
439 * Combine the tonal band spectrum and regular band spectrum
441 * @param spectrum output spectrum buffer
442 * @param num_components number of tonal components
443 * @param components tonal components for this band
444 * @return position of the last tonal coefficient
446 static int add_tonal_components(float *spectrum, int num_components,
447 TonalComponent *components)
449 int i, j, last_pos = -1;
450 float *input, *output;
452 for (i = 0; i < num_components; i++) {
453 last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
454 input = components[i].coef;
455 output = &spectrum[components[i].pos];
457 for (j = 0; j < components[i].num_coefs; j++)
458 output[j] += input[j];
464 #define INTERPOLATE(old, new, nsample) \
465 ((old) + (nsample) * 0.125 * ((new) - (old)))
467 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
470 int i, nsample, band;
471 float mc1_l, mc1_r, mc2_l, mc2_r;
473 for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
474 int s1 = prev_code[i];
475 int s2 = curr_code[i];
479 /* Selector value changed, interpolation needed. */
480 mc1_l = matrix_coeffs[s1 * 2 ];
481 mc1_r = matrix_coeffs[s1 * 2 + 1];
482 mc2_l = matrix_coeffs[s2 * 2 ];
483 mc2_r = matrix_coeffs[s2 * 2 + 1];
485 /* Interpolation is done over the first eight samples. */
486 for (; nsample < band + 8; nsample++) {
487 float c1 = su1[nsample];
488 float c2 = su2[nsample];
489 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
490 c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
492 su2[nsample] = c1 * 2.0 - c2;
496 /* Apply the matrix without interpolation. */
498 case 0: /* M/S decoding */
499 for (; nsample < band + 256; nsample++) {
500 float c1 = su1[nsample];
501 float c2 = su2[nsample];
502 su1[nsample] = c2 * 2.0;
503 su2[nsample] = (c1 - c2) * 2.0;
507 for (; nsample < band + 256; nsample++) {
508 float c1 = su1[nsample];
509 float c2 = su2[nsample];
510 su1[nsample] = (c1 + c2) * 2.0;
511 su2[nsample] = c2 * -2.0;
516 for (; nsample < band + 256; nsample++) {
517 float c1 = su1[nsample];
518 float c2 = su2[nsample];
519 su1[nsample] = c1 + c2;
520 su2[nsample] = c1 - c2;
529 static void get_channel_weights(int index, int flag, float ch[2])
535 ch[0] = (index & 7) / 7.0;
536 ch[1] = sqrt(2 - ch[0] * ch[0]);
538 FFSWAP(float, ch[0], ch[1]);
542 static void channel_weighting(float *su1, float *su2, int *p3)
545 /* w[x][y] y=0 is left y=1 is right */
548 if (p3[1] != 7 || p3[3] != 7) {
549 get_channel_weights(p3[1], p3[0], w[0]);
550 get_channel_weights(p3[3], p3[2], w[1]);
552 for (band = 256; band < 4 * 256; band += 256) {
553 for (nsample = band; nsample < band + 8; nsample++) {
554 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
555 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
557 for(; nsample < band + 256; nsample++) {
558 su1[nsample] *= w[1][0];
559 su2[nsample] *= w[1][1];
566 * Decode a Sound Unit
568 * @param snd the channel unit to be used
569 * @param output the decoded samples before IQMF in float representation
570 * @param channel_num channel number
571 * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
573 static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
574 ChannelUnit *snd, float *output,
575 int channel_num, int coding_mode)
577 int band, ret, num_subbands, last_tonal, num_bands;
578 GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
579 GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
581 if (coding_mode == JOINT_STEREO && channel_num == 1) {
582 if (get_bits(gb, 2) != 3) {
583 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
584 return AVERROR_INVALIDDATA;
587 if (get_bits(gb, 6) != 0x28) {
588 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
589 return AVERROR_INVALIDDATA;
593 /* number of coded QMF bands */
594 snd->bands_coded = get_bits(gb, 2);
596 ret = decode_gain_control(gb, gain2, snd->bands_coded);
600 snd->num_components = decode_tonal_components(gb, snd->components,
602 if (snd->num_components < 0)
603 return snd->num_components;
605 num_subbands = decode_spectrum(gb, snd->spectrum);
607 /* Merge the decoded spectrum and tonal components. */
608 last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
612 /* calculate number of used MLT/QMF bands according to the amount of coded
614 num_bands = (subband_tab[num_subbands] - 1) >> 8;
616 num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
619 /* Reconstruct time domain samples. */
620 for (band = 0; band < 4; band++) {
621 /* Perform the IMDCT step without overlapping. */
622 if (band <= num_bands)
623 imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
625 memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
627 /* gain compensation and overlapping */
628 ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
629 &snd->prev_frame[band * 256],
630 &gain1->g_block[band], &gain2->g_block[band],
631 256, &output[band * 256]);
634 /* Swap the gain control buffers for the next frame. */
635 snd->gc_blk_switch ^= 1;
640 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
643 ATRAC3Context *q = avctx->priv_data;
647 if (q->coding_mode == JOINT_STEREO) {
648 /* channel coupling mode */
649 /* decode Sound Unit 1 */
650 init_get_bits(&q->gb, databuf, avctx->block_align * 8);
652 ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
657 /* Framedata of the su2 in the joint-stereo mode is encoded in
658 * reverse byte order so we need to swap it first. */
659 if (databuf == q->decoded_bytes_buffer) {
660 uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
661 ptr1 = q->decoded_bytes_buffer;
662 for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
663 FFSWAP(uint8_t, *ptr1, *ptr2);
665 const uint8_t *ptr2 = databuf + avctx->block_align - 1;
666 for (i = 0; i < avctx->block_align; i++)
667 q->decoded_bytes_buffer[i] = *ptr2--;
670 /* Skip the sync codes (0xF8). */
671 ptr1 = q->decoded_bytes_buffer;
672 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
673 if (i >= avctx->block_align)
674 return AVERROR_INVALIDDATA;
678 /* set the bitstream reader at the start of the second Sound Unit*/
679 init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);
681 /* Fill the Weighting coeffs delay buffer */
682 memmove(q->weighting_delay, &q->weighting_delay[2],
683 4 * sizeof(*q->weighting_delay));
684 q->weighting_delay[4] = get_bits1(&q->gb);
685 q->weighting_delay[5] = get_bits(&q->gb, 3);
687 for (i = 0; i < 4; i++) {
688 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
689 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
690 q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
693 /* Decode Sound Unit 2. */
694 ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
695 out_samples[1], 1, JOINT_STEREO);
699 /* Reconstruct the channel coefficients. */
700 reverse_matrixing(out_samples[0], out_samples[1],
701 q->matrix_coeff_index_prev,
702 q->matrix_coeff_index_now);
704 channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
706 /* normal stereo mode or mono */
707 /* Decode the channel sound units. */
708 for (i = 0; i < avctx->channels; i++) {
709 /* Set the bitstream reader at the start of a channel sound unit. */
710 init_get_bits(&q->gb,
711 databuf + i * avctx->block_align / avctx->channels,
712 avctx->block_align * 8 / avctx->channels);
714 ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
715 out_samples[i], i, q->coding_mode);
721 /* Apply the iQMF synthesis filter. */
722 for (i = 0; i < avctx->channels; i++) {
723 float *p1 = out_samples[i];
724 float *p2 = p1 + 256;
725 float *p3 = p2 + 256;
726 float *p4 = p3 + 256;
727 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
728 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
729 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
735 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
736 int *got_frame_ptr, AVPacket *avpkt)
738 AVFrame *frame = data;
739 const uint8_t *buf = avpkt->data;
740 int buf_size = avpkt->size;
741 ATRAC3Context *q = avctx->priv_data;
743 const uint8_t *databuf;
745 if (buf_size < avctx->block_align) {
746 av_log(avctx, AV_LOG_ERROR,
747 "Frame too small (%d bytes). Truncated file?\n", buf_size);
748 return AVERROR_INVALIDDATA;
751 /* get output buffer */
752 frame->nb_samples = SAMPLES_PER_FRAME;
753 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
756 /* Check if we need to descramble and what buffer to pass on. */
757 if (q->scrambled_stream) {
758 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
759 databuf = q->decoded_bytes_buffer;
764 ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
766 av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
772 return avctx->block_align;
775 static av_cold void atrac3_init_static_data(void)
780 ff_atrac_generate_tables();
782 /* Initialize the VLC tables. */
783 for (i = 0; i < 7; i++) {
784 spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
785 spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
787 init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
789 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
793 static av_cold int atrac3_decode_init(AVCodecContext *avctx)
795 static int static_init_done;
797 int version, delay, samples_per_frame, frame_factor;
798 const uint8_t *edata_ptr = avctx->extradata;
799 ATRAC3Context *q = avctx->priv_data;
801 if (avctx->channels <= 0 || avctx->channels > 2) {
802 av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
803 return AVERROR(EINVAL);
806 if (!static_init_done)
807 atrac3_init_static_data();
808 static_init_done = 1;
810 /* Take care of the codec-specific extradata. */
811 if (avctx->extradata_size == 14) {
812 /* Parse the extradata, WAV format */
813 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
814 bytestream_get_le16(&edata_ptr)); // Unknown value always 1
815 edata_ptr += 4; // samples per channel
816 q->coding_mode = bytestream_get_le16(&edata_ptr);
817 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
818 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
819 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
820 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
821 bytestream_get_le16(&edata_ptr)); // Unknown always 0
824 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
827 q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO;
828 q->scrambled_stream = 0;
830 if (avctx->block_align != 96 * avctx->channels * frame_factor &&
831 avctx->block_align != 152 * avctx->channels * frame_factor &&
832 avctx->block_align != 192 * avctx->channels * frame_factor) {
833 av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
834 "configuration %d/%d/%d\n", avctx->block_align,
835 avctx->channels, frame_factor);
836 return AVERROR_INVALIDDATA;
838 } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
839 /* Parse the extradata, RM format. */
840 version = bytestream_get_be32(&edata_ptr);
841 samples_per_frame = bytestream_get_be16(&edata_ptr);
842 delay = bytestream_get_be16(&edata_ptr);
843 q->coding_mode = bytestream_get_be16(&edata_ptr);
844 q->scrambled_stream = 1;
847 av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
848 avctx->extradata_size);
849 return AVERROR(EINVAL);
852 /* Check the extradata */
855 av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
856 return AVERROR_INVALIDDATA;
859 if (samples_per_frame != SAMPLES_PER_FRAME &&
860 samples_per_frame != SAMPLES_PER_FRAME * 2) {
861 av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
863 return AVERROR_INVALIDDATA;
866 if (delay != 0x88E) {
867 av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
869 return AVERROR_INVALIDDATA;
872 if (q->coding_mode == STEREO)
873 av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
874 else if (q->coding_mode == JOINT_STEREO) {
875 if (avctx->channels != 2) {
876 av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
877 return AVERROR_INVALIDDATA;
879 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
881 av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
883 return AVERROR_INVALIDDATA;
886 if (avctx->block_align >= UINT_MAX / 2)
887 return AVERROR(EINVAL);
889 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
890 FF_INPUT_BUFFER_PADDING_SIZE);
891 if (!q->decoded_bytes_buffer)
892 return AVERROR(ENOMEM);
894 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
896 /* initialize the MDCT transform */
897 if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
898 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
899 av_freep(&q->decoded_bytes_buffer);
903 /* init the joint-stereo decoding data */
904 q->weighting_delay[0] = 0;
905 q->weighting_delay[1] = 7;
906 q->weighting_delay[2] = 0;
907 q->weighting_delay[3] = 7;
908 q->weighting_delay[4] = 0;
909 q->weighting_delay[5] = 7;
911 for (i = 0; i < 4; i++) {
912 q->matrix_coeff_index_prev[i] = 3;
913 q->matrix_coeff_index_now[i] = 3;
914 q->matrix_coeff_index_next[i] = 3;
917 ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
918 avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
919 ff_fmt_convert_init(&q->fmt_conv, avctx);
921 q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
923 atrac3_decode_close(avctx);
924 return AVERROR(ENOMEM);
930 AVCodec ff_atrac3_decoder = {
932 .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
933 .type = AVMEDIA_TYPE_AUDIO,
934 .id = AV_CODEC_ID_ATRAC3,
935 .priv_data_size = sizeof(ATRAC3Context),
936 .init = atrac3_decode_init,
937 .close = atrac3_decode_close,
938 .decode = atrac3_decode_frame,
939 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
940 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
941 AV_SAMPLE_FMT_NONE },