X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=9c79ddff1e5dc46158db4cd3d37b8aec213d4da6;hb=2912e87a6c9264d556734e2bf94a99c64cf9b102;hp=98bec5cca45dbdfa4e994b6c35e3e33e89fc93b3;hpb=efce1a8fea2d7fb3272992042b9d56f58deb9834;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 98bec5cca45..9c79ddff1e5 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -5,24 +5,25 @@ * 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 + * 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 distortions @@ -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) { \ @@ -96,16 +90,20 @@ typedef struct { /** * 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(16, 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]; @@ -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(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 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 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 void qdm2_init_vlc(void) +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) @@ -403,7 +374,7 @@ static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) * * @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 @@ -465,7 +436,7 @@ 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 @@ -538,7 +509,7 @@ static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_ 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; @@ -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++) @@ -972,7 +942,6 @@ 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 @@ -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); @@ -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 subpackets ordered by type, largest type first */ for (i = 0, max = 256; i < q->sub_packets_B; i++) { - QDM2SubPacket *packet; + QDM2SubPacket *packet= NULL; /* find subpacket with largest type less than max */ - for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) { + 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; - - /* prerotation (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; + ff_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 */ -static 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(); @@ -1756,14 +1704,12 @@ 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 @@ -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 @@ -1911,38 +1856,34 @@ static int qdm2_decode_init(AVCodecContext *avctx) else s->coeff_per_sb_select = 2; - // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[] + // 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); return -1; } - 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); - } + 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; } -static 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); @@ -1978,7 +1919,7 @@ static 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 @@ static 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"), };