X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=4d3b3915fbd2a3157d0b86b7515a14cd02492980;hb=cc5e9e5ff052fe31aa757de79f2d11fb21df3fba;hp=a3373a16d963b98550f666ec1597bfcbe2f0892f;hpb=5ef251e50437ce84a00735c5cac8dd836fb032e9;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index a3373a16d96..4d3b3915fbd 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. */ @@ -34,22 +35,21 @@ #include #include -#define ALT_BITSTREAM_READER_LE +#define 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) { \ @@ -69,14 +69,13 @@ do { \ #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) -#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) - #define SAMPLES_NEEDED \ av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); #define SAMPLES_NEEDED_2(why) \ av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); +#define QDM2_MAX_FRAME_SIZE 512 typedef int8_t sb_int8_array[2][30][64]; @@ -122,13 +121,15 @@ typedef struct { } FFTCoefficient; typedef struct { - DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]); + DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; } QDM2FFT; /** * QDM2 decoder context */ typedef struct { + AVFrame frame; + /// Parameters from codec header, do not change during playback int nb_channels; ///< number of channels int channels; ///< number of channels @@ -139,7 +140,6 @@ typedef struct { /// Parameters built from header parameters, do not change during playback int group_order; ///< order of frame group int fft_order; ///< order of FFT (actually fftorder+1) - int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) int frame_size; ///< size of data frame int frequency_range; int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ @@ -169,12 +169,14 @@ typedef struct { /// I/O data const uint8_t *compressed_data; int compressed_size; - float output_buffer[1024]; + float output_buffer[QDM2_MAX_FRAME_SIZE * 2]; /// 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]; @@ -197,8 +199,6 @@ typedef struct { } QDM2Context; -static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; - static VLC vlc_tab_level; static VLC vlc_tab_diff; static VLC vlc_tab_run; @@ -213,148 +213,124 @@ 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 +382,7 @@ 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 +433,7 @@ 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 +479,8 @@ 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]; } } @@ -522,7 +498,7 @@ static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_ int j,k; int ch; int run, case_val; - int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4}; + static const int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4}; for (ch = 0; ch < channels; ch++) { for (j = 0; j < 64; ) { @@ -707,8 +683,7 @@ 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++) @@ -813,10 +788,10 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l else if (sb >= 24) joined_stereo = 1; else - joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; + joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0; if (joined_stereo) { - if (BITS_LEFT(length,gb) >= 16) + if (get_bits_left(gb) >= 16) for (j = 0; j < 16; j++) sign_bits[j] = get_bits1 (gb); @@ -829,14 +804,14 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l } for (ch = 0; ch < channels; ch++) { - zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; + zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; type34_predictor = 0.0; type34_first = 1; for (j = 0; j < 128; ) { switch (q->coding_method[ch][sb][j / 2]) { case 8: - if (BITS_LEFT(length,gb) >= 10) { + if (get_bits_left(gb) >= 10) { if (zero_encoding) { for (k = 0; k < 5; k++) { if ((j + 2 * k) >= 128) @@ -858,7 +833,7 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l break; case 10: - if (BITS_LEFT(length,gb) >= 1) { + if (get_bits_left(gb) >= 1) { float f = 0.81; if (get_bits1(gb)) @@ -872,7 +847,7 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l break; case 16: - if (BITS_LEFT(length,gb) >= 10) { + if (get_bits_left(gb) >= 10) { if (zero_encoding) { for (k = 0; k < 5; k++) { if ((j + k) >= 128) @@ -892,7 +867,7 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l break; case 24: - if (BITS_LEFT(length,gb) >= 7) { + if (get_bits_left(gb) >= 7) { n = get_bits(gb, 7); for (k = 0; k < 3; k++) samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; @@ -904,24 +879,32 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l break; case 30: - if (BITS_LEFT(length,gb) >= 4) - samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; - else + if (get_bits_left(gb) >= 4) { + unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); + if (index < FF_ARRAY_ELEMS(type30_dequant)) { + samples[0] = type30_dequant[index]; + } else + samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); + } else samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); run = 1; break; case 34: - if (BITS_LEFT(length,gb) >= 7) { + if (get_bits_left(gb) >= 7) { if (type34_first) { type34_div = (float)(1 << get_bits(gb, 2)); samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; type34_predictor = samples[0]; type34_first = 0; } else { - samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; - type34_predictor = samples[0]; + unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); + if (index < FF_ARRAY_ELEMS(type34_delta)) { + samples[0] = type34_delta[index] / type34_div + type34_predictor; + type34_predictor = samples[0]; + } else + samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); } } else { samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); @@ -945,11 +928,11 @@ 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; @@ -964,27 +947,25 @@ 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 */ -static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) +static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb) { int i, k, run, level, diff; - if (BITS_LEFT(length,gb) < 16) + if (get_bits_left(gb) < 16) return; level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); quantized_coeffs[0] = level; for (i = 0; i < 7; ) { - if (BITS_LEFT(length,gb) < 16) + if (get_bits_left(gb) < 16) break; run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; - if (BITS_LEFT(length,gb) < 16) + if (get_bits_left(gb) < 16) break; diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); @@ -1004,16 +985,15 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext * * @param q context * @param gb bitreader context - * @param length packet length in bits */ -static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) +static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb) { int sb, j, k, n, ch; for (ch = 0; ch < q->nb_channels; ch++) { - init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); + init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); - if (BITS_LEFT(length,gb) < 16) { + if (get_bits_left(gb) < 16) { memset(q->quantized_coeffs[ch][0], 0, 8); break; } @@ -1024,11 +1004,11 @@ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, i for (sb = 0; sb < n; sb++) for (ch = 0; ch < q->nb_channels; ch++) for (j = 0; j < 8; j++) { - if (BITS_LEFT(length,gb) < 1) + if (get_bits_left(gb) < 1) break; if (get_bits1(gb)) { for (k=0; k < 8; k++) { - if (BITS_LEFT(length,gb) < 16) + if (get_bits_left(gb) < 16) break; q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); } @@ -1042,7 +1022,7 @@ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, i for (sb = 0; sb < n; sb++) for (ch = 0; ch < q->nb_channels; ch++) { - if (BITS_LEFT(length,gb) < 16) + if (get_bits_left(gb) < 16) break; q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); if (sb > 19) @@ -1057,7 +1037,7 @@ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, i for (sb = 0; sb < n; sb++) for (ch = 0; ch < q->nb_channels; ch++) for (j = 0; j < 8; j++) { - if (BITS_LEFT(length,gb) < 16) + if (get_bits_left(gb) < 16) break; q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; } @@ -1106,16 +1086,14 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) * * @param q context * @param node pointer to node with packet - * @param length packet length in bits */ -static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) +static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; - init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); - - if (length != 0) { - init_tone_level_dequantization(q, &gb, length); + if (node) { + init_get_bits(&gb, node->packet->data, node->packet->size * 8); + init_tone_level_dequantization(q, &gb); fill_tone_level_array(q, 1); } else { fill_tone_level_array(q, 0); @@ -1128,13 +1106,17 @@ static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length * * @param q context * @param node pointer to node with packet - * @param length packet length in bit */ -static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) +static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; + int length = 0; + + if (node) { + length = node->packet->size * 8; + init_get_bits(&gb, node->packet->data, length); + } - init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); if (length >= 32) { int c = get_bits (&gb, 13); @@ -1152,13 +1134,17 @@ 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 bits */ -static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) +static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; + int length = 0; + + if (node) { + length = node->packet->size * 8; + init_get_bits(&gb, node->packet->data, length); + } - init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); } @@ -1178,21 +1164,21 @@ static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); if (nodes[1] != NULL) - process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); + process_subpacket_10(q, nodes[1]); else - process_subpacket_10(q, NULL, 0); + process_subpacket_10(q, NULL); nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) - process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); + process_subpacket_11(q, nodes[2]); else - process_subpacket_11(q, NULL, 0); + process_subpacket_11(q, NULL); nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) - process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); + process_subpacket_12(q, nodes[3]); else - process_subpacket_12(q, NULL, 0); + process_subpacket_12(q, NULL); } @@ -1232,7 +1218,8 @@ 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); @@ -1313,9 +1300,9 @@ static void qdm2_decode_super_block (QDM2Context *q) process_synthesis_subpackets(q, q->sub_packet_list_D); q->do_synth_filter = 1; } else if (q->do_synth_filter) { - process_subpacket_10(q, NULL, 0); - process_subpacket_11(q, NULL, 0); - process_subpacket_12(q, NULL, 0); + process_subpacket_10(q, NULL); + process_subpacket_11(q, NULL); + process_subpacket_12(q, NULL); } /* **************************************************************** */ } @@ -1377,6 +1364,8 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext * return; local_int_14 = (offset >> local_int_8); + if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) + return; if (q->nb_channels > 1) { channel = get_bits1(gb); @@ -1607,13 +1596,17 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) { const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; + float *out = q->output_buffer + channel; 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; + for (i = 0; i < FFALIGN(q->fft_size, 8); i++) { + out[0] += q->fft.complex[channel][i].re * gain; + out[q->channels] += q->fft.complex[channel][i].im * gain; + out += 2 * q->channels; + } } @@ -1623,7 +1616,6 @@ 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 +1627,12 @@ 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 +1644,7 @@ 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 +1661,7 @@ 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(); @@ -1677,52 +1670,6 @@ static av_cold void qdm2_init(QDM2Context *q) { } -#if 0 -static void dump_context(QDM2Context *q) -{ - int i; -#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); - PRINT("compressed_data",q->compressed_data); - PRINT("compressed_size",q->compressed_size); - PRINT("frame_size",q->frame_size); - PRINT("checksum_size",q->checksum_size); - PRINT("channels",q->channels); - PRINT("nb_channels",q->nb_channels); - PRINT("fft_frame_size",q->fft_frame_size); - PRINT("fft_size",q->fft_size); - PRINT("sub_sampling",q->sub_sampling); - PRINT("fft_order",q->fft_order); - PRINT("group_order",q->group_order); - PRINT("group_size",q->group_size); - PRINT("sub_packet",q->sub_packet); - PRINT("frequency_range",q->frequency_range); - PRINT("has_errors",q->has_errors); - PRINT("fft_tone_end",q->fft_tone_end); - PRINT("fft_tone_start",q->fft_tone_start); - PRINT("fft_coefs_index",q->fft_coefs_index); - PRINT("coeff_per_sb_select",q->coeff_per_sb_select); - PRINT("cm_table_select",q->cm_table_select); - PRINT("noise_idx",q->noise_idx); - - 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); - PRINT(" phase", t->phase); - PRINT(" phase_shift", t->phase_shift); - PRINT(" duration", t->duration); - PRINT(" samples_im", t->samples_im); - PRINT(" samples_re", t->samples_re); - PRINT(" table", t->table); - } - -} -#endif - - /** * Init parameters from codec extradata */ @@ -1821,6 +1768,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); extradata += 4; + if (s->channels > MPA_MAX_CHANNELS) + return AVERROR_INVALIDDATA; avctx->sample_rate = AV_RB32(extradata); extradata += 4; @@ -1835,14 +1784,18 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) extradata += 4; s->checksum_size = AV_RB32(extradata); - extradata += 4; + if (s->checksum_size >= 1U << 28) { + av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size); + return AVERROR_INVALIDDATA; + } s->fft_order = av_log2(s->fft_size) + 1; - s->fft_frame_size = 2 * s->fft_size; // complex has two floats // something like max decodable tones s->group_order = av_log2(s->group_size) + 1; s->frame_size = s->group_size / 16; // 16 iterations per super block + if (s->frame_size > QDM2_MAX_FRAME_SIZE) + return AVERROR_INVALIDDATA; s->sub_sampling = s->fft_order - 7; s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); @@ -1885,13 +1838,16 @@ 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; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; -// dump_context(s); return 0; } @@ -1906,7 +1862,7 @@ 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); @@ -1915,8 +1871,6 @@ static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) q->compressed_data = in; q->compressed_size = q->checksum_size; -// dump_context(q); - /* copy old block, clear new block of output samples */ memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); @@ -1942,7 +1896,7 @@ 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; } } @@ -1963,43 +1917,54 @@ 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) +static int qdm2_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; QDM2Context *s = avctx->priv_data; + int16_t *out; + int i, ret; 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); + /* get output buffer */ + s->frame.nb_samples = 16 * s->frame_size; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + out = (int16_t *)s->frame.data[0]; - // 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; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + + return s->checksum_size; } -AVCodec qdm2_decoder = +AVCodec ff_qdm2_decoder = { - .name = "qdm2", - .type = CODEC_TYPE_AUDIO, - .id = CODEC_ID_QDM2, + .name = "qdm2", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_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"), + .init = qdm2_decode_init, + .close = qdm2_decode_close, + .decode = qdm2_decode_frame, + .capabilities = CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), };