X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=2a45495c7cdc7200a75b52eabe09ff2b27d5dde0;hb=87cf70eb237e7586cc7399627dafa1b980ec0b7d;hp=a3373a16d963b98550f666ec1597bfcbe2f0892f;hpb=5ef251e50437ce84a00735c5cac8dd836fb032e9;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index a3373a16d96..2a45495c7cd 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -5,27 +5,28 @@ * Copyright (c) 2005 Alex Beregszaszi * Copyright (c) 2005 Roberto Togni * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file libavcodec/qdm2.c + * @file * QDM2 decoder * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni + * * The decoder is not perfect yet, there are still some distortions * especially on files encoded with 16 or 8 subbands. */ @@ -36,20 +37,19 @@ #define ALT_BITSTREAM_READER_LE #include "avcodec.h" -#include "bitstream.h" +#include "get_bits.h" #include "dsputil.h" +#include "rdft.h" +#include "mpegaudiodsp.h" #include "mpegaudio.h" #include "qdm2data.h" +#include "qdm2_tablegen.h" #undef NDEBUG #include -#define SOFTCLIP_THRESHOLD 27600 -#define HARDCLIP_THRESHOLD 35716 - - #define QDM2_LIST_ADD(list, size, packet) \ do { \ if (size > 0) { \ @@ -122,7 +122,7 @@ typedef struct { } FFTCoefficient; typedef struct { - DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]); + DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; } QDM2FFT; /** @@ -172,9 +172,11 @@ typedef struct { float output_buffer[1024]; /// Synthesis filter - DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]); + MPADSPContext mpadsp; + DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; int synth_buf_offset[MPA_MAX_CHANNELS]; - DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]); + DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; + DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; /// Mixed temporary data used in decoding float tone_level[MPA_MAX_CHANNELS][30][64]; @@ -213,148 +215,124 @@ static VLC vlc_tab_type30; static VLC vlc_tab_type34; static VLC vlc_tab_fft_tone_offset[5]; -static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; -static float noise_table[4096]; -static uint8_t random_dequant_index[256][5]; -static uint8_t random_dequant_type24[128][3]; -static float noise_samples[128]; - -static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); - - -static av_cold void softclip_table_init(void) { - int i; - double dfl = SOFTCLIP_THRESHOLD - 32767; - float delta = 1.0 / -dfl; - for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++) - softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF); -} - - -// random generated table -static av_cold void rnd_table_init(void) { - int i,j; - uint32_t ldw,hdw; - uint64_t tmp64_1; - uint64_t random_seed = 0; - float delta = 1.0 / 16384.0; - for(i = 0; i < 4096 ;i++) { - random_seed = random_seed * 214013 + 2531011; - noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3; - } - - for (i = 0; i < 256 ;i++) { - random_seed = 81; - ldw = i; - for (j = 0; j < 5 ;j++) { - random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); - ldw = (uint32_t)ldw % (uint32_t)random_seed; - tmp64_1 = (random_seed * 0x55555556); - hdw = (uint32_t)(tmp64_1 >> 32); - random_seed = (uint64_t)(hdw + (ldw >> 31)); - } - } - for (i = 0; i < 128 ;i++) { - random_seed = 25; - ldw = i; - for (j = 0; j < 3 ;j++) { - random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); - ldw = (uint32_t)ldw % (uint32_t)random_seed; - tmp64_1 = (random_seed * 0x66666667); - hdw = (uint32_t)(tmp64_1 >> 33); - random_seed = hdw + (ldw >> 31); - } - } -} - - -static av_cold void init_noise_samples(void) { - int i; - int random_seed = 0; - float delta = 1.0 / 16384.0; - for (i = 0; i < 128;i++) { - random_seed = random_seed * 214013 + 2531011; - noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0); - } -} - +static const uint16_t qdm2_vlc_offs[] = { + 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, +}; static av_cold void qdm2_init_vlc(void) { - init_vlc (&vlc_tab_level, 8, 24, - vlc_tab_level_huffbits, 1, 1, - vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_diff, 8, 37, - vlc_tab_diff_huffbits, 1, 1, - vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_run, 5, 6, - vlc_tab_run_huffbits, 1, 1, - vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_level_exp_alt_vlc, 8, 28, - fft_level_exp_alt_huffbits, 1, 1, - fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_level_exp_vlc, 8, 20, - fft_level_exp_huffbits, 1, 1, - fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_stereo_exp_vlc, 6, 7, - fft_stereo_exp_huffbits, 1, 1, - fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_stereo_phase_vlc, 6, 9, - fft_stereo_phase_huffbits, 1, 1, - fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, - vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, - vlc_tab_tone_level_idx_mid_huffbits, 1, 1, - vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, - vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_type30, 6, 9, - vlc_tab_type30_huffbits, 1, 1, - vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_type34, 5, 10, - vlc_tab_type34_huffbits, 1, 1, - vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, - vlc_tab_fft_tone_offset_0_huffbits, 1, 1, - vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, - vlc_tab_fft_tone_offset_1_huffbits, 1, 1, - vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, - vlc_tab_fft_tone_offset_2_huffbits, 1, 1, - vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, - vlc_tab_fft_tone_offset_3_huffbits, 1, 1, - vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, - vlc_tab_fft_tone_offset_4_huffbits, 1, 1, - vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); + static int vlcs_initialized = 0; + static VLC_TYPE qdm2_table[3838][2]; + + if (!vlcs_initialized) { + + vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; + vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; + init_vlc (&vlc_tab_level, 8, 24, + vlc_tab_level_huffbits, 1, 1, + vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; + vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; + init_vlc (&vlc_tab_diff, 8, 37, + vlc_tab_diff_huffbits, 1, 1, + vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; + vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; + init_vlc (&vlc_tab_run, 5, 6, + vlc_tab_run_huffbits, 1, 1, + vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; + fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; + init_vlc (&fft_level_exp_alt_vlc, 8, 28, + fft_level_exp_alt_huffbits, 1, 1, + fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + + fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; + fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; + init_vlc (&fft_level_exp_vlc, 8, 20, + fft_level_exp_huffbits, 1, 1, + fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; + fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; + init_vlc (&fft_stereo_exp_vlc, 6, 7, + fft_stereo_exp_huffbits, 1, 1, + fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; + fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; + init_vlc (&fft_stereo_phase_vlc, 6, 9, + fft_stereo_phase_huffbits, 1, 1, + fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; + vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; + init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, + vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, + vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; + vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; + init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, + vlc_tab_tone_level_idx_mid_huffbits, 1, 1, + vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; + vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; + init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, + vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, + vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; + vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; + init_vlc (&vlc_tab_type30, 6, 9, + vlc_tab_type30_huffbits, 1, 1, + vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; + vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; + init_vlc (&vlc_tab_type34, 5, 10, + vlc_tab_type34_huffbits, 1, 1, + vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; + vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; + init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, + vlc_tab_fft_tone_offset_0_huffbits, 1, 1, + vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; + vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; + init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, + vlc_tab_fft_tone_offset_1_huffbits, 1, 1, + vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; + vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; + init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, + vlc_tab_fft_tone_offset_2_huffbits, 1, 1, + vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; + vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; + init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, + vlc_tab_fft_tone_offset_3_huffbits, 1, 1, + vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; + vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; + init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, + vlc_tab_fft_tone_offset_4_huffbits, 1, 1, + vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlcs_initialized=1; + } } - -/* for floating point to fixed point conversion */ -static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); - - static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) { int value; @@ -406,7 +384,7 @@ static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value /** - * Fills a QDM2SubPacket structure with packet type, size, and data pointer. + * Fill a QDM2SubPacket structure with packet type, size, and data pointer. * * @param gb bitreader context * @param sub_packet packet under analysis @@ -457,7 +435,7 @@ static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int /** - * Replaces 8 elements with their average value. + * Replace 8 elements with their average value. * Called by qdm2_decode_superblock before starting subblock decoding. * * @param q context @@ -503,8 +481,8 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) for (ch = 0; ch < q->nb_channels; ch++) for (j = 0; j < 64; j++) { - q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); - q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); + q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; + q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; } } @@ -707,8 +685,7 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra for (sb = 0; sb < 30; sb++) for (j = 0; j < 64; j++) acc += tone_level_idx_temp[ch][sb][j]; - if (acc) - tmp = c * 256 / (acc & 0xffff); + multres = 0x66666667 * (acc * 10); esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); for (ch = 0; ch < nb_channels; ch++) @@ -945,11 +922,11 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l for (chs = 0; chs < q->nb_channels; chs++) for (k = 0; k < run; k++) if ((j + k) < 128) - q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); + q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs]; } else { for (k = 0; k < run; k++) if ((j + k) < 128) - q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); + q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; } j += run; @@ -964,7 +941,6 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l * This is similar to process_subpacket_9, but for a single channel and for element [0] * same VLC tables as process_subpacket_9 are used. * - * @param q context * @param quantized_coeffs pointer to quantized_coeffs[ch][0] * @param gb bitreader context * @param length packet length in bits @@ -1232,7 +1208,8 @@ static void qdm2_decode_super_block (QDM2Context *q) init_get_bits(&gb, header.data, header.size*8); if (header.type == 2 || header.type == 4 || header.type == 5) { - int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); + int csum = 257 * get_bits(&gb, 8); + csum += 2 * get_bits(&gb, 8); csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); @@ -1610,7 +1587,7 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) int i; q->fft.complex[channel][0].re *= 2.0f; q->fft.complex[channel][0].im = 0.0f; - ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); + q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); /* add samples to output buffer */ for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; @@ -1623,7 +1600,6 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) */ static void qdm2_synthesis_filter (QDM2Context *q, int index) { - OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; int i, k, ch, sb_used, sub_sampling, dither_state = 0; /* copy sb_samples */ @@ -1635,11 +1611,12 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) q->sb_samples[ch][(8 * index) + i][k] = 0; for (ch = 0; ch < q->nb_channels; ch++) { - OUT_INT *samples_ptr = samples + ch; + float *samples_ptr = q->samples + ch; for (i = 0; i < 8; i++) { - ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), - mpa_window, &dither_state, + ff_mpa_synth_filter_float(&q->mpadsp, + q->synth_buf[ch], &(q->synth_buf_offset[ch]), + ff_mpa_synth_window_float, &dither_state, samples_ptr, q->nb_channels, q->sb_samples[ch][(8 * index) + i]); samples_ptr += 32 * q->nb_channels; @@ -1651,7 +1628,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) for (ch = 0; ch < q->channels; ch++) for (i = 0; i < q->frame_size; i++) - q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); + q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; } @@ -1668,7 +1645,7 @@ static av_cold void qdm2_init(QDM2Context *q) { initialized = 1; qdm2_init_vlc(); - ff_mpa_synth_init(mpa_window); + ff_mpa_synth_init_float(ff_mpa_synth_window_float); softclip_table_init(); rnd_table_init(); init_noise_samples(); @@ -1835,7 +1812,6 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) extradata += 4; s->checksum_size = AV_RB32(extradata); - extradata += 4; s->fft_order = av_log2(s->fft_size) + 1; s->fft_frame_size = 2 * s->fft_size; // complex has two floats @@ -1885,11 +1861,12 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) return -1; } - ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT); + ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); + ff_mpadsp_init(&s->mpadsp); qdm2_init(s); - avctx->sample_fmt = SAMPLE_FMT_S16; + avctx->sample_fmt = AV_SAMPLE_FMT_S16; // dump_context(s); return 0; @@ -1906,7 +1883,7 @@ static av_cold int qdm2_decode_close(AVCodecContext *avctx) } -static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) +static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) { int ch, i; const int frame_size = (q->frame_size * q->channels); @@ -1942,7 +1919,7 @@ static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { SAMPLES_NEEDED_2("has errors, and C list is not empty") - return; + return -1; } } @@ -1963,39 +1940,44 @@ static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) out[i] = value; } + + return 0; } static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - const uint8_t *buf, int buf_size) + AVPacket *avpkt) { + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; QDM2Context *s = avctx->priv_data; + int16_t *out = data; + int i; if(!buf) return 0; if(buf_size < s->checksum_size) return -1; - *data_size = s->channels * s->frame_size * sizeof(int16_t); - av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", buf_size, buf, s->checksum_size, data, *data_size); - qdm2_decode(s, buf, data); - - // reading only when next superblock found - if (s->sub_packet == 0) { - return s->checksum_size; + for (i = 0; i < 16; i++) { + if (qdm2_decode(s, buf, out) < 0) + return -1; + out += s->channels * s->frame_size; } - return 0; + *data_size = (uint8_t*)out - (uint8_t*)data; + + return s->checksum_size; } -AVCodec qdm2_decoder = +AVCodec ff_qdm2_decoder = { .name = "qdm2", - .type = CODEC_TYPE_AUDIO, + .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_QDM2, .priv_data_size = sizeof(QDM2Context), .init = qdm2_decode_init,