X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=198f11f2715c42f357e72766194992b5012f1ace;hb=d1be646e906487b395190af1d6dd8d33c22bf25f;hp=f385077c3a4db90369321c15493eda324952c7c1;hpb=db795a1c57e13345a53fcb7533549ab6c539efde;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index f385077c3a4..198f11f2715 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -5,28 +5,29 @@ * Copyright (c) 2005 Alex Beregszaszi * Copyright (c) 2005 Roberto Togni * - * This library is free software; you can redistribute it and/or + * This file is part of Libav. + * + * 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 of the License, or (at your option) any later version. + * version 2.1 of the License, or (at your option) any later version. * - * This library 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 this library; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file qdm2.c + * @file * QDM2 decoder * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni - * The decoder is not perfect yet, there are still some distorions expecially - * on files encoded with 16 or 8 subbands + * The decoder is not perfect yet, there are still some distortions + * especially on files encoded with 16 or 8 subbands. */ #include @@ -35,25 +36,18 @@ #define ALT_BITSTREAM_READER_LE #include "avcodec.h" -#include "bitstream.h" +#include "get_bits.h" #include "dsputil.h" - -#ifdef CONFIG_MPEGAUDIO_HP -#define USE_HIGHPRECISION -#endif - +#include "fft.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) { \ @@ -94,18 +88,22 @@ typedef struct { } QDM2SubPacket; /** - * A node in subpacket list + * A node in the subpacket list */ -typedef struct _QDM2SubPNode { +typedef struct QDM2SubPNode { QDM2SubPacket *packet; ///< packet - struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node + struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node } QDM2SubPNode; +typedef struct { + float re; + float im; +} QDM2Complex; + typedef struct { float level; - float *samples_im; - float *samples_re; - float *table; + QDM2Complex *complex; + const float *table; int phase; int phase_shift; int duration; @@ -122,14 +120,7 @@ typedef struct { } FFTCoefficient; typedef struct { - float re; - float im; -} QDM2Complex; - -typedef struct { - QDM2Complex complex[256 + 1] __attribute__((aligned(16))); - float samples_im[MPA_MAX_CHANNELS][256]; - float samples_re[MPA_MAX_CHANNELS][256]; + DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; } QDM2FFT; /** @@ -170,19 +161,18 @@ typedef struct { int fft_coefs_min_index[5]; int fft_coefs_max_index[5]; int fft_level_exp[6]; - FFTContext fft_ctx; - FFTComplex exptab[128]; + RDFTContext rdft_ctx; QDM2FFT fft; /// I/O data - uint8_t *compressed_data; + const uint8_t *compressed_data; int compressed_size; float output_buffer[1024]; /// Synthesis filter - MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16))); + DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; int synth_buf_offset[MPA_MAX_CHANNELS]; - int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16))); + DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; /// Mixed temporary data used in decoding float tone_level[MPA_MAX_CHANNELS][30][64]; @@ -196,12 +186,12 @@ typedef struct { int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; // Flags - int has_errors; ///< packet have errors + int has_errors; ///< packet has errors int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type int do_synth_filter; ///< used to perform or skip synthesis filter int sub_packet; - int noise_idx; ///< Index for dithering noise table + int noise_idx; ///< index for dithering noise table } QDM2Context; @@ -221,146 +211,127 @@ 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 MPA_INT mpa_window[512] __attribute__((aligned(16))); - - -static void softclip_table_init() { - 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 void rnd_table_init() { - 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 void init_noise_samples() { - 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 void qdm2_init_vlc() +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 float f2i_scale = (float) (1 << (FRAC_BITS - 15)); +static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) @@ -401,9 +372,9 @@ static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) * @param length data length * @param value checksum value * - * @return 0 if checksum is ok + * @return 0 if checksum is OK */ -static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) { +static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { int i; for (i=0; i < length; i++) @@ -414,7 +385,7 @@ static uint16_t qdm2_packet_checksum (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 @@ -441,15 +412,15 @@ static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data } - av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n", + av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); } /** - * Return node pointer to first packet of requested type in list + * Return node pointer to first packet of requested type in list. * - * @param list list of subpacket to be scanned + * @param list list of subpackets to be scanned * @param type type of searched subpacket * @return node pointer for subpacket if found, else NULL */ @@ -465,8 +436,8 @@ static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int /** - * Replaces 8 elements with their average value - * Called by qdm2_decode_superblock before starting subblocks decoding + * Replace 8 elements with their average value. + * Called by qdm2_decode_superblock before starting subblock decoding. * * @param q context */ @@ -494,8 +465,8 @@ static void average_quantized_coeffs (QDM2Context *q) /** - * Build subband samples with noise weighted by q->tone_level - * Called by synthfilt_build_sb_samples + * Build subband samples with noise weighted by q->tone_level. + * Called by synthfilt_build_sb_samples. * * @param q context * @param sb subband index @@ -518,14 +489,14 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) /** - * Called while processing data from subpackets 11 and 12 - * Used after making changes to coding_method array + * Called while processing data from subpackets 11 and 12. + * Used after making changes to coding_method array. * * @param sb subband index * @param channels number of channels * @param coding_method q->coding_method[0][0][0] */ - void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) +static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) { int j,k; int ch; @@ -538,7 +509,7 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) run = 1; case_val = 8; } else { - switch (switchtable[coding_method[ch][sb][j]]) { + switch (switchtable[coding_method[ch][sb][j]-8]) { case 0: run = 10; case_val = 10; break; case 1: run = 1; case_val = 16; break; case 2: run = 5; case_val = 24; break; @@ -657,7 +628,7 @@ static void fill_tone_level_array (QDM2Context *q, int flag) * c is built with data from subpacket 11 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples * - * @param tone_level_idx + * @param tone_level_idx * @param tone_level_idx_temp * @param coding_method q->coding_method[0][0][0] * @param nb_channels number of channels @@ -683,7 +654,7 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra SAMPLES_NEEDED for (ch = 0; ch < nb_channels; ch++) for (sb = 0; sb < 30; sb++) { - for (j = 1; j < 64; j++) { + for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer add1 = tone_level_idx[ch][sb][j] - 10; if (add1 < 0) add1 = 0; @@ -715,8 +686,7 @@ 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++) @@ -790,7 +760,7 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra * * @param q context * @param gb bitreader context - * @param length packet length in bit + * @param length packet length in bits * @param sb_min lower subband processed (sb_min included) * @param sb_max higher subband processed (sb_max excluded) */ @@ -916,7 +886,7 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; else samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); - + run = 1; break; @@ -968,14 +938,13 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l /** - * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]) + * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). * 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 + * 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 bit + * @param length packet length in bits */ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) { @@ -995,10 +964,10 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext if (BITS_LEFT(length,gb) < 16) break; diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); - + for (k = 1; k <= run; k++) quantized_coeffs[i + k] = (level + ((k * diff) / run)); - + level += diff; i += run; } @@ -1012,7 +981,7 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext * * @param q context * @param gb bitreader context - * @param length packet length in bit + * @param length packet length in bits */ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) { @@ -1114,7 +1083,7 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) * * @param q context * @param node pointer to node with packet - * @param length packet length in bit + * @param length packet length in bits */ static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) { @@ -1160,7 +1129,7 @@ static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length * * @param q context * @param node pointer to node with packet - * @param length packet length in bit + * @param length packet length in bits */ static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) { @@ -1205,7 +1174,7 @@ static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) /* - * Decode superblock, fill packet lists + * Decode superblock, fill packet lists. * * @param q context */ @@ -1240,7 +1209,8 @@ 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); @@ -1274,7 +1244,7 @@ static void qdm2_decode_super_block (QDM2Context *q) break; } - /* decode sub packet */ + /* decode subpacket */ packet = &q->sub_packets[i]; qdm2_decode_sub_packet_header(&gb, packet); next_index = packet->size + get_bits_count(&gb) / 8; @@ -1291,10 +1261,10 @@ static void qdm2_decode_super_block (QDM2Context *q) packet_bytes -= sub_packet_size; - /* add sub packet to 'all sub packets' list */ + /* add subpacket to 'all subpackets' list */ q->sub_packet_list_A[i].packet = packet; - /* add sub packet to related list */ + /* add subpacket to related list */ if (packet->type == 8) { SAMPLES_NEEDED_2("packet type 8"); return; @@ -1430,17 +1400,17 @@ static void qdm2_decode_fft_packets (QDM2Context *q) if (q->sub_packet_list_B[0].packet == NULL) return; - /* reset minimum indices for FFT coefficients */ + /* reset minimum indexes for FFT coefficients */ q->fft_coefs_index = 0; for (i=0; i < 5; i++) q->fft_coefs_min_index[i] = -1; - /* process sub packets ordered by type, largest type first */ + /* process subpackets ordered by type, largest type first */ for (i = 0, max = 256; i < q->sub_packets_B; i++) { - QDM2SubPacket *packet; + QDM2SubPacket *packet= NULL; - /* find sub packet with largest type less than max */ - for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) { + /* find subpacket with largest type less than max */ + for (j = 0, min = 0; j < q->sub_packets_B; j++) { value = q->sub_packet_list_B[j].packet->type; if (value > min && value < max) { min = value; @@ -1451,6 +1421,9 @@ static void qdm2_decode_fft_packets (QDM2Context *q) max = min; /* check for errors (?) */ + if (!packet) + return; + if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) return; @@ -1470,17 +1443,17 @@ static void qdm2_decode_fft_packets (QDM2Context *q) if (duration >= 0 && duration < 4) qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); } else if (type == 31) { - for (i=0; i < 4; i++) - qdm2_fft_decode_tones(q, i, &gb, unknown_flag); + for (j=0; j < 4; j++) + qdm2_fft_decode_tones(q, j, &gb, unknown_flag); } else if (type == 46) { - for (i=0; i < 6; i++) - q->fft_level_exp[i] = get_bits(&gb, 6); - for (i=0; i < 4; i++) - qdm2_fft_decode_tones(q, i, &gb, unknown_flag); + for (j=0; j < 6; j++) + q->fft_level_exp[j] = get_bits(&gb, 6); + for (j=0; j < 4; j++) + qdm2_fft_decode_tones(q, j, &gb, unknown_flag); } } // Loop on B packets - /* calculate maximum indices for FFT coefficients */ + /* calculate maximum indexes for FFT coefficients */ for (i = 0, j = -1; i < 5; i++) if (q->fft_coefs_min_index[i] >= 0) { if (j >= 0) @@ -1508,10 +1481,10 @@ static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) /* generate FFT coefficients for tone */ if (tone->duration >= 3 || tone->cutoff >= 3) { - tone->samples_im[0] += c.im; - tone->samples_re[0] += c.re; - tone->samples_im[1] -= c.im; - tone->samples_re[1] -= c.re; + tone->complex[0].im += c.im; + tone->complex[0].re += c.re; + tone->complex[1].im -= c.im; + tone->complex[1].re -= c.re; } else { f[1] = -tone->table[4]; f[0] = tone->table[3] - tone->table[0]; @@ -1520,12 +1493,12 @@ static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) f[4] = tone->table[0] - tone->table[1]; f[5] = tone->table[2]; for (i = 0; i < 2; i++) { - tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i]; - tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); + tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; + tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); } for (i = 0; i < 4; i++) { - tone->samples_re[i] += c.re * f[i+2]; - tone->samples_im[i] += c.im * f[i+2]; + tone->complex[i].re += c.re * f[i+2]; + tone->complex[i].im += c.im * f[i+2]; } } @@ -1543,8 +1516,7 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) const double iscale = 0.25 * M_PI; for (ch = 0; ch < q->channels; ch++) { - memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float)); - memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float)); + memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); } @@ -1562,10 +1534,10 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) c.re = level * cos(q->fft_coefs[i].phase * iscale); c.im = level * sin(q->fft_coefs[i].phase * iscale); - q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re; - q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im; - q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re; - q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im; + q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; + q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; + q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; + q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; } /* generate existing FFT tones */ @@ -1595,9 +1567,8 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) tone.cutoff = (offset >= 60) ? 3 : 2; tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63]; - tone.samples_im = &q->fft.samples_im[ch][offset]; - tone.samples_re = &q->fft.samples_re[ch][offset]; - tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; + tone.complex = &q->fft.complex[ch][offset]; + tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); tone.duration = i; @@ -1613,37 +1584,14 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) { - const int n = 1 << (q->fft_order - 1); - const int n2 = n >> 1; - const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f; - float c, s, f0, f1, f2, f3; - int i, j; - - /* pre rotation (or something like that) */ - for (i=1; i < n2; i++) { - j = (n - i); - c = q->exptab[i].re; - s = -q->exptab[i].im; - f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain; - f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain; - f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain; - f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain; - q->fft.complex[i].re = s * f0 - c * f1 + f2; - q->fft.complex[i].im = c * f0 + s * f1 + f3; - q->fft.complex[j].re = -s * f0 + c * f1 + f2; - q->fft.complex[j].im = c * f0 + s * f1 - f3; - } - - q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0; - q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0; - q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0; - q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0; - - ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex); - ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex); + const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; + int i; + q->fft.complex[channel][0].re *= 2.0f; + q->fft.complex[channel][0].im = 0.0f; + 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)[i]; + q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; } @@ -1669,7 +1617,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) 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_window, &dither_state, samples_ptr, q->nb_channels, q->sb_samples[ch][(8 * index) + i]); samples_ptr += 32 * q->nb_channels; @@ -1690,15 +1638,15 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) * * @param q context */ -void qdm2_init(QDM2Context *q) { - static int inited = 0; +static av_cold void qdm2_init(QDM2Context *q) { + static int initialized = 0; - if (inited != 0) + if (initialized != 0) return; - inited = 1; + initialized = 1; qdm2_init_vlc(); - ff_mpa_synth_init(mpa_window); + ff_mpa_synth_init(ff_mpa_synth_window); softclip_table_init(); rnd_table_init(); init_noise_samples(); @@ -1737,7 +1685,7 @@ static void dump_context(QDM2Context *q) for (i = q->fft_tone_start; i < q->fft_tone_end; i++) { FFTTone *t = &q->fft_tones[i]; - + av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); // PRINT(" level", t->level); @@ -1756,28 +1704,26 @@ static void dump_context(QDM2Context *q) /** * Init parameters from codec extradata */ -static int qdm2_decode_init(AVCodecContext *avctx) +static av_cold int qdm2_decode_init(AVCodecContext *avctx) { QDM2Context *s = avctx->priv_data; uint8_t *extradata; int extradata_size; int tmp_val, tmp, size; - int i; - float alpha; - + /* extradata parsing - + Structure: wave { frma (QDM2) QDCA QDCP } - + 32 size (including this field) 32 tag (=frma) 32 type (=QDM2 or QDMC) - + 32 size (including this field, in bytes) 32 tag (=QDCA) // maybe mandatory parameters 32 unknown (=1) @@ -1787,7 +1733,7 @@ static int qdm2_decode_init(AVCodecContext *avctx) 32 block size (=4096) 32 frame size (=256) (for one channel) 32 packet size (=1300) - + 32 size (including this field, in bytes) 32 tag (=QDCP) // maybe some tuneable parameters 32 float1 (=1.0) @@ -1834,7 +1780,7 @@ static int qdm2_decode_init(AVCodecContext *avctx) extradata += 8; extradata_size -= 8; - size = BE_32(extradata); + size = AV_RB32(extradata); if(size > extradata_size){ av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", @@ -1844,30 +1790,29 @@ static int qdm2_decode_init(AVCodecContext *avctx) extradata += 4; av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); - if (BE_32(extradata) != MKBETAG('Q','D','C','A')) { + if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); return -1; } extradata += 8; - avctx->channels = s->nb_channels = s->channels = BE_32(extradata); + avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); extradata += 4; - avctx->sample_rate = BE_32(extradata); + avctx->sample_rate = AV_RB32(extradata); extradata += 4; - avctx->bit_rate = BE_32(extradata); + avctx->bit_rate = AV_RB32(extradata); extradata += 4; - s->group_size = BE_32(extradata); + s->group_size = AV_RB32(extradata); extradata += 4; - s->fft_size = BE_32(extradata); + s->fft_size = AV_RB32(extradata); extradata += 4; - s->checksum_size = BE_32(extradata); - extradata += 4; + s->checksum_size = AV_RB32(extradata); s->fft_order = av_log2(s->fft_size) + 1; s->fft_frame_size = 2 * s->fft_size; // complex has two floats @@ -1876,12 +1821,9 @@ static int qdm2_decode_init(AVCodecContext *avctx) s->group_order = av_log2(s->group_size) + 1; s->frame_size = s->group_size / 16; // 16 iterations per super block - if (s->fft_order == 8) - s->sub_sampling = 1; - else - s->sub_sampling = 2; + s->sub_sampling = s->fft_order - 7; s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); - + switch ((s->sub_sampling * 2 + s->channels - 1)) { case 0: tmp = 40; break; case 1: tmp = 48; break; @@ -1899,11 +1841,11 @@ static int qdm2_decode_init(AVCodecContext *avctx) s->cm_table_select = tmp_val; if (s->sub_sampling == 0) - tmp = 16000; + tmp = 7999; else tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; /* - 0: 16000 -> 1 + 0: 7999 -> 0 1: 20000 -> 2 2: 28000 -> 2 */ @@ -1914,39 +1856,38 @@ static int qdm2_decode_init(AVCodecContext *avctx) else s->coeff_per_sb_select = 2; - if (s->fft_order != 8 && s->fft_order != 9) + // Fail on unknown fft order + if ((s->fft_order < 7) || (s->fft_order > 9)) { av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); - - ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1); - - for (i = 1; i < (1 << (s->fft_order - 2)); i++) { - alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1)); - s->exptab[i].re = cos(alpha); - s->exptab[i].im = sin(alpha); + return -1; } + ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); + qdm2_init(s); - + + avctx->sample_fmt = AV_SAMPLE_FMT_S16; + // dump_context(s); return 0; } -static int qdm2_decode_close(AVCodecContext *avctx) +static av_cold int qdm2_decode_close(AVCodecContext *avctx) { QDM2Context *s = avctx->priv_data; - ff_fft_end(&s->fft_ctx); - + ff_rdft_end(&s->rdft_ctx); + return 0; } -void qdm2_decode (QDM2Context *q, 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); - + /* select input buffer */ q->compressed_data = in; q->compressed_size = q->checksum_size; @@ -1960,11 +1901,11 @@ void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) /* decode block of QDM2 compressed data */ if (q->sub_packet == 0) { q->has_errors = 0; // zero it for a new super block - av_log(NULL,AV_LOG_DEBUG,"Super block follows\n"); + av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); qdm2_decode_super_block(q); } - /* parse sub packets */ + /* parse subpackets */ if (!q->has_errors) { if (q->sub_packet == 2) qdm2_decode_fft_packets(q); @@ -1978,7 +1919,7 @@ void qdm2_decode (QDM2Context *q, 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; } } @@ -1999,40 +1940,48 @@ void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) out[i] = value; } + + return 0; } static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - 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 == NULL) || (buf_size < s->checksum_size)) + if(!buf) return 0; - - *data_size = s->channels * s->frame_size * sizeof(int16_t); + if(buf_size < s->checksum_size) + return -1; 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, .close = qdm2_decode_close, .decode = qdm2_decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), };