X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=668e513a661fb1c782852b6b48d58abfa17f6db8;hb=a594f17f83a1ffdc1eec18818208fe39487dd5d7;hp=e0674beda7fffa1f7fe34a51876de0d81c91460c;hpb=1c7a8c17ff97ea4b6b10f1c4ff9ff4de30e3665a;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index e0674beda7f..668e513a661 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -5,26 +5,28 @@ * 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 * especially on files encoded with 16 or 8 subbands. */ @@ -33,25 +35,19 @@ #include #include -#define ALT_BITSTREAM_READER_LE +#include "libavutil/channel_layout.h" + +#define BITSTREAM_READER_LE #include "avcodec.h" #include "bitstream.h" -#include "dsputil.h" - -#ifdef CONFIG_MPEGAUDIO_HP -#define USE_HIGHPRECISION -#endif - +#include "internal.h" #include "mpegaudio.h" +#include "mpegaudiodsp.h" +#include "rdft.h" +#include "vlc.h" #include "qdm2data.h" - -#undef NDEBUG -#include - - -#define SOFTCLIP_THRESHOLD 27600 -#define HARDCLIP_THRESHOLD 35716 +#include "qdm2_tablegen.h" #define QDM2_LIST_ADD(list, size, packet) \ @@ -73,21 +69,20 @@ 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]; /** * Subpacket */ -typedef struct { +typedef struct QDM2SubPacket { int type; ///< subpacket type unsigned int size; ///< subpacket size const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) @@ -96,16 +91,20 @@ 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 { +typedef struct QDM2Complex { + float re; + float im; +} QDM2Complex; + +typedef struct FFTTone { float level; - float *samples_im; - float *samples_re; - float *table; + QDM2Complex *complex; + const float *table; int phase; int phase_shift; int duration; @@ -113,7 +112,7 @@ typedef struct { short cutoff; } FFTTone; -typedef struct { +typedef struct FFTCoefficient { int16_t sub_packet; uint8_t channel; int16_t offset; @@ -121,21 +120,14 @@ typedef struct { uint8_t phase; } 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]; +typedef struct QDM2FFT { + DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; } QDM2FFT; /** * QDM2 decoder context */ -typedef struct { +typedef struct QDM2Context { /// Parameters from codec header, do not change during playback int nb_channels; ///< number of channels int channels; ///< number of channels @@ -146,7 +138,6 @@ 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% */ @@ -170,19 +161,20 @@ 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]; + float output_buffer[QDM2_MAX_FRAME_SIZE * 2]; /// Synthesis filter - MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16))); + MPADSPContext mpadsp; + DECLARE_ALIGNED(32, float, 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(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]; @@ -205,8 +197,6 @@ 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; @@ -221,231 +211,239 @@ 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 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 +}; -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 VLC_TYPE qdm2_table[3838][2]; + + 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); } - -/* for floating point to fixed point conversion */ -static float f2i_scale = (float) (1 << (FRAC_BITS - 15)); - - -static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) +static int qdm2_get_vlc(BitstreamContext *bc, VLC *vlc, int flag, int depth) { int value; - value = get_vlc2(gb, vlc->table, vlc->bits, depth); + value = bitstream_read_vlc(bc, vlc->table, vlc->bits, depth); /* stage-2, 3 bits exponent escape sequence */ if (value-- == 0) - value = get_bits (gb, get_bits (gb, 3) + 1); + value = bitstream_read(bc, bitstream_read(bc, 3) + 1); /* stage-3, optional */ if (flag) { int tmp = vlc_stage3_values[value]; if ((value & ~3) > 0) - tmp += get_bits (gb, (value >> 2)); + tmp += bitstream_read(bc, value >> 2); value = tmp; } return value; } - -static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) +static int qdm2_get_se_vlc(VLC *vlc, BitstreamContext *bc, int depth) { - int value = qdm2_get_vlc (gb, vlc, 0, depth); + int value = qdm2_get_vlc(bc, vlc, 0, depth); return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); } - /** * QDM2 checksum * - * @param data pointer to data to be checksum'ed + * @param data pointer to data to be checksummed * @param length data length * @param value checksum value * * @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++) + for (i = 0; i < length; i++) value -= data[i]; return (uint16_t)(value & 0xffff); } - /** - * 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 bc bitreader context * @param sub_packet packet under analysis */ -static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) +static void qdm2_decode_sub_packet_header(BitstreamContext *bc, + QDM2SubPacket *sub_packet) { - sub_packet->type = get_bits (gb, 8); + sub_packet->type = bitstream_read(bc, 8); if (sub_packet->type == 0) { sub_packet->size = 0; sub_packet->data = NULL; } else { - sub_packet->size = get_bits (gb, 8); + sub_packet->size = bitstream_read(bc, 8); - if (sub_packet->type & 0x80) { - sub_packet->size <<= 8; - sub_packet->size |= get_bits (gb, 8); - sub_packet->type &= 0x7f; - } + if (sub_packet->type & 0x80) { + sub_packet->size <<= 8; + sub_packet->size |= bitstream_read(bc, 8); + sub_packet->type &= 0x7f; + } - if (sub_packet->type == 0x7f) - sub_packet->type |= (get_bits (gb, 8) << 8); + if (sub_packet->type == 0x7f) + sub_packet->type |= bitstream_read(bc, 8) << 8; - sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data + // FIXME: this depends on bitreader-internal data + sub_packet->data = &bc->buffer[bitstream_tell(bc) / 8]; } - 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); + av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n", + sub_packet->type, sub_packet->size, bitstream_tell(bc) / 8); } - /** * Return node pointer to first packet of requested type in list. * @@ -453,9 +451,10 @@ static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub * @param type type of searched subpacket * @return node pointer for subpacket if found, else NULL */ -static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) +static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, + int type) { - while (list != NULL && list->packet != NULL) { + while (list && list->packet) { if (list->packet->type == type) return list; list = list->next; @@ -463,14 +462,13 @@ static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int return NULL; } - /** - * 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 */ -static void average_quantized_coeffs (QDM2Context *q) +static void average_quantized_coeffs(QDM2Context *q) { int i, j, n, ch, sum; @@ -487,12 +485,11 @@ static void average_quantized_coeffs (QDM2Context *q) if (sum > 0) sum--; - for (j=0; j < 8; j++) + for (j = 0; j < 8; j++) q->quantized_coeffs[ch][i][j] = sum; } } - /** * Build subband samples with noise weighted by q->tone_level. * Called by synthfilt_build_sb_samples. @@ -500,7 +497,7 @@ static void average_quantized_coeffs (QDM2Context *q) * @param q context * @param sb subband index */ -static void build_sb_samples_from_noise (QDM2Context *q, int sb) +static void build_sb_samples_from_noise(QDM2Context *q, int sb) { int ch, j; @@ -509,14 +506,16 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) if (!q->nb_channels) return; - for (ch = 0; ch < q->nb_channels; ch++) + 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]; } + } } - /** * Called while processing data from subpackets 11 and 12. * Used after making changes to coding_method array. @@ -525,44 +524,65 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) * @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 int fix_coding_method_array(int sb, int channels, + sb_int8_array coding_method) { - int j,k; + 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}; for (ch = 0; ch < channels; ch++) { for (j = 0; j < 64; ) { - if((coding_method[ch][sb][j] - 8) > 22) { - run = 1; + if (coding_method[ch][sb][j] < 8) + return -1; + if ((coding_method[ch][sb][j] - 8) > 22) { + run = 1; case_val = 8; } else { - switch (switchtable[coding_method[ch][sb][j]]) { - case 0: run = 10; case_val = 10; break; - case 1: run = 1; case_val = 16; break; - case 2: run = 5; case_val = 24; break; - case 3: run = 3; case_val = 30; break; - case 4: run = 1; case_val = 30; break; - case 5: run = 1; case_val = 8; break; - default: run = 1; case_val = 8; break; + 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; + case 3: run = 3; + case_val = 30; + break; + case 4: run = 1; + case_val = 30; + break; + case 5: run = 1; + case_val = 8; + break; + default: run = 1; + case_val = 8; + break; } } - for (k = 0; k < run; k++) - if (j + k < 128) - if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) + for (k = 0; k < run; k++) { + if (j + k < 128) { + if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) { if (k > 0) { - SAMPLES_NEEDED + SAMPLES_NEEDED //not debugged, almost never used - memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); - memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); + memset(&coding_method[ch][sb][j + k], case_val, + k *sizeof(int8_t)); + memset(&coding_method[ch][sb][j + k], case_val, + 3 * sizeof(int8_t)); } + } + } + } j += run; } } + return 0; } - /** * Related to synthesis filter * Called by process_subpacket_10 @@ -570,15 +590,11 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) * @param q context * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 */ -static void fill_tone_level_array (QDM2Context *q, int flag) +static void fill_tone_level_array(QDM2Context *q, int flag) { int i, sb, ch, sb_used; int tmp, tab; - // This should never happen - if (q->nb_channels <= 0) - return; - for (ch = 0; ch < q->nb_channels; ch++) for (sb = 0; sb < 30; sb++) for (i = 0; i < 8; i++) { @@ -646,16 +662,14 @@ static void fill_tone_level_array (QDM2Context *q, int flag) } } } - - return; } - /** * Related to synthesis filter * Called by process_subpacket_11 * 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 + * 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_temp @@ -665,25 +679,24 @@ static void fill_tone_level_array (QDM2Context *q, int flag) * @param superblocktype_2_3 flag based on superblock packet type * @param cm_table_select q->cm_table_select */ -static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, - sb_int8_array coding_method, int nb_channels, - int c, int superblocktype_2_3, int cm_table_select) +static void fill_coding_method_array(sb_int8_array tone_level_idx, + sb_int8_array tone_level_idx_temp, + sb_int8_array coding_method, + int nb_channels, + int c, int superblocktype_2_3, + int cm_table_select) { int ch, sb, j; int tmp, acc, esp_40, comp; int add1, add2, add3, add4; int64_t multres; - // This should never happen - if (nb_channels <= 0) - return; - if (!superblocktype_2_3) { /* This case is untested, no samples available */ 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; @@ -710,94 +723,93 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra } tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; } - acc = 0; - for (ch = 0; ch < nb_channels; ch++) - 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++) - for (sb = 0; sb < 30; sb++) - for (j = 0; j < 64; j++) { - comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; - if (comp < 0) - comp += 0xff; - comp /= 256; // signed shift - switch(sb) { - case 0: - if (comp < 30) - comp = 30; - comp += 15; - break; - case 1: - if (comp < 24) - comp = 24; - comp += 10; - break; - case 2: - case 3: - case 4: - if (comp < 16) - comp = 16; - } - if (comp <= 5) - tmp = 0; - else if (comp <= 10) - tmp = 10; - else if (comp <= 16) - tmp = 16; - else if (comp <= 24) - tmp = -1; - else - tmp = 0; - coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; + + acc = 0; + for (ch = 0; ch < nb_channels; ch++) + for (sb = 0; sb < 30; sb++) + for (j = 0; j < 64; j++) + acc += tone_level_idx_temp[ch][sb][j]; + + multres = 0x66666667LL * (acc * 10); + esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); + for (ch = 0; ch < nb_channels; ch++) + for (sb = 0; sb < 30; sb++) + for (j = 0; j < 64; j++) { + comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; + if (comp < 0) + comp += 0xff; + comp /= 256; // signed shift + switch(sb) { + case 0: + if (comp < 30) + comp = 30; + comp += 15; + break; + case 1: + if (comp < 24) + comp = 24; + comp += 10; + break; + case 2: + case 3: + case 4: + if (comp < 16) + comp = 16; } + if (comp <= 5) + tmp = 0; + else if (comp <= 10) + tmp = 10; + else if (comp <= 16) + tmp = 16; + else if (comp <= 24) + tmp = -1; + else + tmp = 0; + coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; + } + for (sb = 0; sb < 30; sb++) + fix_coding_method_array(sb, nb_channels, coding_method); + for (ch = 0; ch < nb_channels; ch++) for (sb = 0; sb < 30; sb++) - fix_coding_method_array(sb, nb_channels, coding_method); - for (ch = 0; ch < nb_channels; ch++) - for (sb = 0; sb < 30; sb++) - for (j = 0; j < 64; j++) - if (sb >= 10) { - if (coding_method[ch][sb][j] < 10) - coding_method[ch][sb][j] = 10; + for (j = 0; j < 64; j++) + if (sb >= 10) { + if (coding_method[ch][sb][j] < 10) + coding_method[ch][sb][j] = 10; + } else { + if (sb >= 2) { + if (coding_method[ch][sb][j] < 16) + coding_method[ch][sb][j] = 16; } else { - if (sb >= 2) { - if (coding_method[ch][sb][j] < 16) - coding_method[ch][sb][j] = 16; - } else { - if (coding_method[ch][sb][j] < 30) - coding_method[ch][sb][j] = 30; - } + if (coding_method[ch][sb][j] < 30) + coding_method[ch][sb][j] = 30; } + } } else { // superblocktype_2_3 != 0 for (ch = 0; ch < nb_channels; ch++) for (sb = 0; sb < 30; sb++) for (j = 0; j < 64; j++) coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; } - - return; } - /** - * - * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 - * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used + * Called by process_subpacket_11 to process more data from subpacket 11 + * with sb 0-8. + * Called by process_subpacket_12 to process data from subpacket 12 with + * sb 8-sb_used. * * @param q context - * @param gb bitreader context + * @param bc bitreader context * @param length packet length in bits * @param sb_min lower subband processed (sb_min included) * @param sb_max higher subband processed (sb_max excluded) */ -static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) +static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc, + int length, int sb_min, int sb_max) { int sb, j, k, n, ch, run, channels; - int joined_stereo, zero_encoding, chs; + int joined_stereo, zero_encoding; int type34_first; float type34_div = 0; float type34_predictor; @@ -806,14 +818,12 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l if (length == 0) { // If no data use noise for (sb=sb_min; sb < sb_max; sb++) - build_sb_samples_from_noise (q, sb); + build_sb_samples_from_noise(q, sb); return; } for (sb = sb_min; sb < sb_max; sb++) { - FIX_NOISE_IDX(q->noise_idx); - channels = q->nb_channels; if (q->nb_channels <= 1 || sb < 12) @@ -821,38 +831,43 @@ 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 = (bitstream_bits_left(bc) >= 1) ? bitstream_read_bit(bc) : 0; if (joined_stereo) { - if (BITS_LEFT(length,gb) >= 16) + if (bitstream_bits_left(bc) >= 16) for (j = 0; j < 16; j++) - sign_bits[j] = get_bits1 (gb); + sign_bits[j] = bitstream_read_bit(bc); for (j = 0; j < 64; j++) if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; - fix_coding_method_array(sb, q->nb_channels, q->coding_method); + if (fix_coding_method_array(sb, q->nb_channels, + q->coding_method)) { + build_sb_samples_from_noise(q, sb); + continue; + } channels = 1; } for (ch = 0; ch < channels; ch++) { - zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; + FIX_NOISE_IDX(q->noise_idx); + zero_encoding = (bitstream_bits_left(bc) >= 1) ? bitstream_read_bit(bc) : 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 (bitstream_bits_left(bc) >= 10) { if (zero_encoding) { for (k = 0; k < 5; k++) { if ((j + 2 * k) >= 128) break; - samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; + samples[2 * k] = bitstream_read_bit(bc) ? dequant_1bit[joined_stereo][2 * bitstream_read_bit(bc)] : 0; } } else { - n = get_bits(gb, 8); + n = bitstream_read(bc, 8); for (k = 0; k < 5; k++) samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; } @@ -866,10 +881,10 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l break; case 10: - if (BITS_LEFT(length,gb) >= 1) { + if (bitstream_bits_left(bc) >= 1) { float f = 0.81; - if (get_bits1(gb)) + if (bitstream_read_bit(bc)) f = -f; f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; samples[0] = f; @@ -880,15 +895,15 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l break; case 16: - if (BITS_LEFT(length,gb) >= 10) { + if (bitstream_bits_left(bc) >= 10) { if (zero_encoding) { for (k = 0; k < 5; k++) { if ((j + k) >= 128) break; - samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; + samples[k] = (bitstream_read_bit(bc) == 0) ? 0 : dequant_1bit[joined_stereo][2 * bitstream_read_bit(bc)]; } } else { - n = get_bits (gb, 8); + n = bitstream_read (bc, 8); for (k = 0; k < 5; k++) samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; } @@ -900,8 +915,8 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l break; case 24: - if (BITS_LEFT(length,gb) >= 7) { - n = get_bits(gb, 7); + if (bitstream_bits_left(bc) >= 7) { + n = bitstream_read(bc, 7); for (k = 0; k < 3; k++) samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; } else { @@ -912,24 +927,32 @@ 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 (bitstream_bits_left(bc) >= 4) { + unsigned index = qdm2_get_vlc(bc, &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 (bitstream_bits_left(bc) >= 7) { if (type34_first) { - type34_div = (float)(1 << get_bits(gb, 2)); - samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; + type34_div = (float)(1 << bitstream_read(bc, 2)); + samples[0] = ((float)bitstream_read(bc, 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(bc, &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); @@ -944,20 +967,22 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l } if (joined_stereo) { - float tmp[10][MPA_MAX_CHANNELS]; - - for (k = 0; k < run; k++) { - tmp[k][0] = samples[k]; - tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; + for (k = 0; k < run && j + k < 128; k++) { + q->sb_samples[0][j + k][sb] = + q->tone_level[0][sb][(j + k) / 2] * samples[k]; + if (q->nb_channels == 2) { + if (sign_bits[(j + k) / 8]) + q->sb_samples[1][j + k][sb] = + q->tone_level[1][sb][(j + k) / 2] * -samples[k]; + else + q->sb_samples[1][j + k][sb] = + q->tone_level[1][sb][(j + k) / 2] * samples[k]; + } } - 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); } 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; @@ -966,35 +991,35 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l } // subband loop } - /** - * 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] + * 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. * - * @param q context * @param quantized_coeffs pointer to quantized_coeffs[ch][0] - * @param gb bitreader context - * @param length packet length in bits + * @param bc bitreader context */ -static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) +static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs, + BitstreamContext *bc) { int i, k, run, level, diff; - if (BITS_LEFT(length,gb) < 16) + if (bitstream_bits_left(bc) < 16) return; - level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); + level = qdm2_get_vlc(bc, &vlc_tab_level, 0, 2); quantized_coeffs[0] = level; for (i = 0; i < 7; ) { - if (BITS_LEFT(length,gb) < 16) + if (bitstream_bits_left(bc) < 16) break; - run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; + run = qdm2_get_vlc(bc, &vlc_tab_run, 0, 1) + 1; - if (BITS_LEFT(length,gb) < 16) + if (bitstream_bits_left(bc) < 16) break; - diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); + diff = qdm2_get_se_vlc(&vlc_tab_diff, bc, 2); for (k = 1; k <= run; k++) quantized_coeffs[i + k] = (level + ((k * diff) / run)); @@ -1004,24 +1029,23 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext } } - /** * Related to synthesis filter, process data from packet 10 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 - * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 + * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with + * data from packet 10 * * @param q context - * @param gb bitreader context - * @param length packet length in bits + * @param bc bitreader context */ -static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) +static void init_tone_level_dequantization(QDM2Context *q, BitstreamContext *bc) { 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], bc); - if (BITS_LEFT(length,gb) < 16) { + if (bitstream_bits_left(bc) < 16) { memset(q->quantized_coeffs[ch][0], 0, 8); break; } @@ -1032,13 +1056,13 @@ 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 (bitstream_bits_left(bc) < 1) break; - if (get_bits1(gb)) { + if (bitstream_read_bit(bc)) { for (k=0; k < 8; k++) { - if (BITS_LEFT(length,gb) < 16) + if (bitstream_bits_left(bc) < 16) break; - q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); + q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_hi1, 0, 2); } } else { for (k=0; k < 8; k++) @@ -1050,9 +1074,9 @@ 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 (bitstream_bits_left(bc) < 16) break; - q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); + q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_hi2, 0, 2); if (sb > 19) q->tone_level_idx_hi2[ch][sb] -= 16; else @@ -1065,9 +1089,9 @@ 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 (bitstream_bits_left(bc) < 16) break; - q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; + q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; } } @@ -1077,29 +1101,29 @@ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, i * @param q context * @param node pointer to node with packet */ -static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) +static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) { - GetBitContext gb; + BitstreamContext bc; int i, j, k, n, ch, run, level, diff; - init_get_bits(&gb, node->packet->data, node->packet->size*8); + bitstream_init8(&bc, node->packet->data, node->packet->size); - n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function + n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; for (i = 1; i < n; i++) - for (ch=0; ch < q->nb_channels; ch++) { - level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); + for (ch = 0; ch < q->nb_channels; ch++) { + level = qdm2_get_vlc(&bc, &vlc_tab_level, 0, 2); q->quantized_coeffs[ch][i][0] = level; for (j = 0; j < (8 - 1); ) { - run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; - diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); + run = qdm2_get_vlc(&bc, &vlc_tab_run, 0, 1) + 1; + diff = qdm2_get_se_vlc(&vlc_tab_diff, &bc, 2); for (k = 1; k <= run; k++) - q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); + q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); level += diff; - j += run; + j += run; } } @@ -1108,66 +1132,71 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) q->quantized_coeffs[ch][0][i] = 0; } - /** * Process subpacket 10 if not null, else * * @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)); + BitstreamContext bc; - if (length != 0) { - init_tone_level_dequantization(q, &gb, length); + if (node) { + bitstream_init8(&bc, node->packet->data, node->packet->size); + init_tone_level_dequantization(q, &bc); fill_tone_level_array(q, 1); } else { fill_tone_level_array(q, 0); } } - /** * Process subpacket 11 * * @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; + BitstreamContext bc; + int length = 0; + + if (node) { + length = node->packet->size * 8; + bitstream_init(&bc, 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); + int c = bitstream_read(&bc, 13); if (c > 3) - fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, - q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); + fill_coding_method_array(q->tone_level_idx, + q->tone_level_idx_temp, q->coding_method, + q->nb_channels, 8 * c, + q->superblocktype_2_3, q->cm_table_select); } - synthfilt_build_sb_samples(q, &gb, length, 0, 8); + synthfilt_build_sb_samples(q, &bc, length, 0, 8); } - /** * Process subpacket 12 * * @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; + BitstreamContext bc; + int length = 0; + + if (node) { + length = node->packet->size * 8; + bitstream_init(&bc, 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)); + synthfilt_build_sb_samples(q, &bc, length, 8, QDM2_SB_USED(q->sub_sampling)); } /* @@ -1176,42 +1205,41 @@ static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length * @param q context * @param list list with synthesis filter packets (list D) */ -static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) +static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) { QDM2SubPNode *nodes[4]; nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); - if (nodes[0] != NULL) + if (nodes[0]) process_subpacket_9(q, nodes[0]); 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); + if (nodes[1]) + 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)); + if (nodes[0] && nodes[1] && nodes[2]) + 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)); + if (nodes[0] && nodes[1] && nodes[3]) + process_subpacket_12(q, nodes[3]); else - process_subpacket_12(q, NULL, 0); + process_subpacket_12(q, NULL); } - /* * Decode superblock, fill packet lists. * * @param q context */ -static void qdm2_decode_super_block (QDM2Context *q) +static void qdm2_decode_super_block(QDM2Context *q) { - GetBitContext gb; + BitstreamContext bc; QDM2SubPacket header, *packet; int i, packet_bytes, sub_packet_size, sub_packets_D; unsigned int next_index = 0; @@ -1221,32 +1249,33 @@ static void qdm2_decode_super_block (QDM2Context *q) memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); q->sub_packets_B = 0; - sub_packets_D = 0; + sub_packets_D = 0; average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] - init_get_bits(&gb, q->compressed_data, q->compressed_size*8); - qdm2_decode_sub_packet_header(&gb, &header); + bitstream_init8(&bc, q->compressed_data, q->compressed_size); + qdm2_decode_sub_packet_header(&bc, &header); if (header.type < 2 || header.type >= 8) { q->has_errors = 1; - av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); + av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); return; } q->superblocktype_2_3 = (header.type == 2 || header.type == 3); - packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); + packet_bytes = (q->compressed_size - bitstream_tell(&bc) / 8); - init_get_bits(&gb, header.data, header.size*8); + bitstream_init8(&bc, header.data, header.size); 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 * bitstream_read(&bc, 8); + csum += 2 * bitstream_read(&bc, 8); csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); if (csum != 0) { q->has_errors = 1; - av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); + av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); return; } } @@ -1261,14 +1290,19 @@ static void qdm2_decode_super_block (QDM2Context *q) for (i = 0; packet_bytes > 0; i++) { int j; + if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { + SAMPLES_NEEDED_2("too many packet bytes"); + return; + } + q->sub_packet_list_A[i].next = NULL; if (i > 0) { q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; /* seek to next block */ - init_get_bits(&gb, header.data, header.size*8); - skip_bits(&gb, next_index*8); + bitstream_init8(&bc, header.data, header.size); + bitstream_skip(&bc, next_index * 8); if (next_index >= header.size) break; @@ -1276,8 +1310,8 @@ static void qdm2_decode_super_block (QDM2Context *q) /* decode subpacket */ packet = &q->sub_packets[i]; - qdm2_decode_sub_packet_header(&gb, packet); - next_index = packet->size + get_bits_count(&gb) / 8; + qdm2_decode_sub_packet_header(&bc, packet); + next_index = packet->size + bitstream_tell(&bc) / 8; sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; if (packet->type == 0) @@ -1303,79 +1337,78 @@ static void qdm2_decode_super_block (QDM2Context *q) QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); } else if (packet->type == 13) { for (j = 0; j < 6; j++) - q->fft_level_exp[j] = get_bits(&gb, 6); + q->fft_level_exp[j] = bitstream_read(&bc, 6); } else if (packet->type == 14) { for (j = 0; j < 6; j++) - q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); + q->fft_level_exp[j] = qdm2_get_vlc(&bc, &fft_level_exp_vlc, 0, 2); } else if (packet->type == 15) { SAMPLES_NEEDED_2("packet type 15") return; - } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { + } else if (packet->type >= 16 && packet->type < 48 && + !fft_subpackets[packet->type - 16]) { /* packets for FFT */ QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); } } // Packet bytes loop -/* **************************************************************** */ - if (q->sub_packet_list_D[0].packet != NULL) { + if (q->sub_packet_list_D[0].packet) { 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); } -/* **************************************************************** */ } - -static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, - int offset, int duration, int channel, - int exp, int phase) +static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, + int offset, int duration, int channel, + int exp, int phase) { if (q->fft_coefs_min_index[duration] < 0) q->fft_coefs_min_index[duration] = q->fft_coefs_index; - q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); + q->fft_coefs[q->fft_coefs_index].sub_packet = + ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); q->fft_coefs[q->fft_coefs_index].channel = channel; - q->fft_coefs[q->fft_coefs_index].offset = offset; - q->fft_coefs[q->fft_coefs_index].exp = exp; - q->fft_coefs[q->fft_coefs_index].phase = phase; + q->fft_coefs[q->fft_coefs_index].offset = offset; + q->fft_coefs[q->fft_coefs_index].exp = exp; + q->fft_coefs[q->fft_coefs_index].phase = phase; q->fft_coefs_index++; } - -static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) +static void qdm2_fft_decode_tones(QDM2Context *q, int duration, + BitstreamContext *bc, int b) { int channel, stereo, phase, exp; - int local_int_4, local_int_8, stereo_phase, local_int_10; + int local_int_4, local_int_8, stereo_phase, local_int_10; int local_int_14, stereo_exp, local_int_20, local_int_28; int n, offset; - local_int_4 = 0; + local_int_4 = 0; local_int_28 = 0; local_int_20 = 2; - local_int_8 = (4 - duration); + local_int_8 = (4 - duration); local_int_10 = 1 << (q->group_order - duration - 1); - offset = 1; + offset = 1; while (1) { if (q->superblocktype_2_3) { - while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { + while ((n = qdm2_get_vlc(bc, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { offset = 1; if (n == 0) { - local_int_4 += local_int_10; + local_int_4 += local_int_10; local_int_28 += (1 << local_int_8); } else { - local_int_4 += 8*local_int_10; + local_int_4 += 8 * local_int_10; local_int_28 += (8 << local_int_8); } } offset += (n - 2); } else { - offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); + offset += qdm2_get_vlc(bc, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); while (offset >= (local_int_10 - 1)) { - offset += (1 - (local_int_10 - 1)); + offset += (1 - (local_int_10 - 1)); local_int_4 += local_int_10; local_int_28 += (1 << local_int_8); } @@ -1385,26 +1418,28 @@ 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); - stereo = get_bits1(gb); + channel = bitstream_read_bit(bc); + stereo = bitstream_read_bit(bc); } else { channel = 0; - stereo = 0; + stereo = 0; } - exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); + exp = qdm2_get_vlc(bc, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; - exp = (exp < 0) ? 0 : exp; + exp = (exp < 0) ? 0 : exp; - phase = get_bits(gb, 3); - stereo_exp = 0; + phase = bitstream_read(bc, 3); + stereo_exp = 0; stereo_phase = 0; if (stereo) { - stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); - stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); + stereo_exp = (exp - qdm2_get_vlc(bc, &fft_stereo_exp_vlc, 0, 1)); + stereo_phase = (phase - qdm2_get_vlc(bc, &fft_stereo_phase_vlc, 0, 1)); if (stereo_phase < 0) stereo_phase += 8; } @@ -1412,38 +1447,39 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext * if (q->frequency_range > (local_int_14 + 1)) { int sub_packet = (local_int_20 + local_int_28); - qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); + qdm2_fft_init_coefficient(q, sub_packet, offset, duration, + channel, exp, phase); if (stereo) - qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); + qdm2_fft_init_coefficient(q, sub_packet, offset, duration, + 1 - channel, + stereo_exp, stereo_phase); } - offset++; } } - -static void qdm2_decode_fft_packets (QDM2Context *q) +static void qdm2_decode_fft_packets(QDM2Context *q) { int i, j, min, max, value, type, unknown_flag; - GetBitContext gb; + BitstreamContext bc; - if (q->sub_packet_list_B[0].packet == NULL) + if (!q->sub_packet_list_B[0].packet) return; - /* reset minimum indices for FFT coefficients */ + /* reset minimum indexes for FFT coefficients */ q->fft_coefs_index = 0; - for (i=0; i < 5; i++) + 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; + min = value; packet = q->sub_packet_list_B[j].packet; } } @@ -1451,11 +1487,16 @@ static void qdm2_decode_fft_packets (QDM2Context *q) max = min; /* check for errors (?) */ - if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) + if (!packet) + return; + + if (i == 0 && + (packet->type < 16 || packet->type >= 48 || + fft_subpackets[packet->type - 16])) return; /* decode FFT tones */ - init_get_bits (&gb, packet->data, packet->size*8); + bitstream_init8(&bc, packet->data, packet->size); if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) unknown_flag = 1; @@ -1468,19 +1509,19 @@ static void qdm2_decode_fft_packets (QDM2Context *q) int duration = q->sub_sampling + 5 - (type & 15); if (duration >= 0 && duration < 4) - qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); + qdm2_fft_decode_tones(q, duration, &bc, 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, &bc, 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] = bitstream_read(&bc, 6); + for (j = 0; j < 4; j++) + qdm2_fft_decode_tones(q, j, &bc, 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) @@ -1491,60 +1532,59 @@ static void qdm2_decode_fft_packets (QDM2Context *q) q->fft_coefs_max_index[j] = q->fft_coefs_index; } - -static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) +static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) { - float level, f[6]; - int i; - QDM2Complex c; - const double iscale = 2.0*M_PI / 512.0; + float level, f[6]; + int i; + QDM2Complex c; + const double iscale = 2.0 * M_PI / 512.0; tone->phase += tone->phase_shift; /* calculate current level (maximum amplitude) of tone */ level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; - c.im = level * sin(tone->phase*iscale); - c.re = level * cos(tone->phase*iscale); + c.im = level * sin(tone->phase * iscale); + c.re = level * cos(tone->phase * iscale); /* 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]; - f[2] = 1.0 - tone->table[2] - tone->table[3]; - f[3] = tone->table[1] + tone->table[4] - 1.0; - f[4] = tone->table[0] - tone->table[1]; - f[5] = tone->table[2]; + f[0] = tone->table[3] - tone->table[0]; + f[2] = 1.0 - tone->table[2] - tone->table[3]; + f[3] = tone->table[1] + tone->table[4] - 1.0; + 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]; } } /* copy the tone if it has not yet died out */ if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { - memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); - q->fft_tone_end = (q->fft_tone_end + 1) % 1000; + memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); + q->fft_tone_end = (q->fft_tone_end + 1) % 1000; } } - -static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) +static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) { int i, j, ch; 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 +1602,10 @@ 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 +1635,8 @@ 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; @@ -1610,50 +1649,28 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) } } - -static void qdm2_calculate_fft (QDM2Context *q, int channel, 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; + float *out = q->output_buffer + channel; + 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]; + 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; + } } - /** * @param q context * @param index subpacket number */ -static void qdm2_synthesis_filter (QDM2Context *q, int index) +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 */ @@ -1661,17 +1678,18 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) for (ch = 0; ch < q->channels; ch++) for (i = 0; i < 8; i++) - for (k=sb_used; k < SBLIMIT; k++) + for (k = sb_used; k < SBLIMIT; k++) 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, - samples_ptr, q->nb_channels, - q->sb_samples[ch][(8 * index) + i]); + 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; } } @@ -1681,89 +1699,31 @@ 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]; } - /** * Init static data (does not depend on specific file) * * @param q context */ -void qdm2_init(QDM2Context *q) { - static int inited = 0; - - if (inited != 0) - return; - inited = 1; - +static av_cold void qdm2_init_static_data(AVCodec *codec) { 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(); - - av_log(NULL, AV_LOG_DEBUG, "init done\n"); -} - - -#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 */ -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 @@ -1801,10 +1761,10 @@ static int qdm2_decode_init(AVCodecContext *avctx) if (!avctx->extradata || (avctx->extradata_size < 48)) { av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); - return -1; + return AVERROR_INVALIDDATA; } - extradata = avctx->extradata; + extradata = avctx->extradata; extradata_size = avctx->extradata_size; while (extradata_size > 7) { @@ -1817,64 +1777,72 @@ static int qdm2_decode_init(AVCodecContext *avctx) if (extradata_size < 12) { av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", extradata_size); - return -1; + return AVERROR_INVALIDDATA; } if (memcmp(extradata, "frmaQDM", 7)) { av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); - return -1; + return AVERROR_INVALIDDATA; } if (extradata[7] == 'C') { // s->is_qdmc = 1; - av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); - return -1; + avpriv_report_missing_feature(avctx, "QDMC version 1"); + return AVERROR_PATCHWELCOME; } 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", extradata_size, size); - return -1; + return AVERROR_INVALIDDATA; } 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; + return AVERROR_INVALIDDATA; } extradata += 8; - avctx->channels = s->nb_channels = s->channels = BE_32(extradata); + avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); extradata += 4; + if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) + return AVERROR_INVALIDDATA; + avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO : + AV_CH_LAYOUT_MONO; - 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); + 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)); @@ -1911,38 +1879,34 @@ 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; + avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order); + return AVERROR_PATCHWELCOME; } - - 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); + if (s->fft_size != (1 << (s->fft_order - 1))) { + av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); + return AVERROR_INVALIDDATA; } - qdm2_init(s); + ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); + ff_mpadsp_init(&s->mpadsp); + + 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); @@ -1951,8 +1915,6 @@ void qdm2_decode (QDM2Context *q, 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)); @@ -1976,9 +1938,9 @@ void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) for (ch = 0; ch < q->channels; ch++) { qdm2_calculate_fft(q, ch, q->sub_packet); - if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { + if (!q->has_errors && q->sub_packet_list_C[0].packet) { SAMPLES_NEEDED_2("has errors, and C list is not empty") - return; + return -1; } } @@ -1988,7 +1950,7 @@ void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) q->sub_packet = (q->sub_packet + 1) % 16; - /* clip and convert output float[] to 16bit signed samples */ + /* clip and convert output float[] to 16-bit signed samples */ for (i = 0; i < frame_size; i++) { int value = (int)q->output_buffer[i]; @@ -1999,40 +1961,53 @@ 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) +static int qdm2_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { + AVFrame *frame = data; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; QDM2Context *s = avctx->priv_data; + int16_t *out; + int i, ret; - if((buf == NULL) || (buf_size < s->checksum_size)) + 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 */ + frame->nb_samples = 16 * s->frame_size; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + out = (int16_t *)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 ((ret = qdm2_decode(s, buf, out)) < 0) + return ret; + out += s->channels * s->frame_size; } - return 0; + *got_frame_ptr = 1; + + return s->checksum_size; } -AVCodec qdm2_decoder = -{ - .name = "qdm2", - .type = CODEC_TYPE_AUDIO, - .id = CODEC_ID_QDM2, - .priv_data_size = sizeof(QDM2Context), - .init = qdm2_decode_init, - .close = qdm2_decode_close, - .decode = qdm2_decode_frame, +AVCodec ff_qdm2_decoder = { + .name = "qdm2", + .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_QDM2, + .priv_data_size = sizeof(QDM2Context), + .init = qdm2_decode_init, + .init_static_data = qdm2_init_static_data, + .close = qdm2_decode_close, + .decode = qdm2_decode_frame, + .capabilities = AV_CODEC_CAP_DR1, };