X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=114b9781dbe1cd3748f41da79001f8f53717fac9;hb=7019f1a510ee69df946294a88e178b119cb6fb4a;hp=f385077c3a4db90369321c15493eda324952c7c1;hpb=db795a1c57e13345a53fcb7533549ab6c539efde;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index f385077c3a4..114b9781dbe 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -5,28 +5,29 @@ * Copyright (c) 2005 Alex Beregszaszi * Copyright (c) 2005 Roberto Togni * - * This library is free software; you can redistribute it and/or + * This file is part of FFmpeg. + * + * FFmpeg 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, + * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file qdm2.c * QDM2 decoder * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni - * The decoder is not perfect yet, there are still some distorions expecially - * on files encoded with 16 or 8 subbands + * The decoder is not perfect yet, there are still some distortions + * especially on files encoded with 16 or 8 subbands. */ #include @@ -94,18 +95,18 @@ typedef struct { } QDM2SubPacket; /** - * A node in subpacket list + * A node in the subpacket list */ -typedef struct _QDM2SubPNode { +typedef struct QDM2SubPNode { QDM2SubPacket *packet; ///< packet - struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node + struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node } QDM2SubPNode; typedef struct { float level; float *samples_im; float *samples_re; - float *table; + const float *table; int phase; int phase_shift; int duration; @@ -127,7 +128,7 @@ typedef struct { } QDM2Complex; typedef struct { - QDM2Complex complex[256 + 1] __attribute__((aligned(16))); + DECLARE_ALIGNED_16(QDM2Complex, complex[256 + 1]); float samples_im[MPA_MAX_CHANNELS][256]; float samples_re[MPA_MAX_CHANNELS][256]; } QDM2FFT; @@ -175,14 +176,14 @@ typedef struct { QDM2FFT fft; /// I/O data - uint8_t *compressed_data; + const uint8_t *compressed_data; int compressed_size; float output_buffer[1024]; /// Synthesis filter - MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16))); + DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]); int synth_buf_offset[MPA_MAX_CHANNELS]; - int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16))); + DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]); /// Mixed temporary data used in decoding float tone_level[MPA_MAX_CHANNELS][30][64]; @@ -196,12 +197,12 @@ typedef struct { int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; // Flags - int has_errors; ///< packet have errors + int has_errors; ///< packet has errors int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type int do_synth_filter; ///< used to perform or skip synthesis filter int sub_packet; - int noise_idx; ///< Index for dithering noise table + int noise_idx; ///< index for dithering noise table } QDM2Context; @@ -227,10 +228,10 @@ 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 DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); -static void softclip_table_init() { +static void softclip_table_init(void) { int i; double dfl = SOFTCLIP_THRESHOLD - 32767; float delta = 1.0 / -dfl; @@ -240,7 +241,7 @@ static void softclip_table_init() { // random generated table -static void rnd_table_init() { +static void rnd_table_init(void) { int i,j; uint32_t ldw,hdw; uint64_t tmp64_1; @@ -276,7 +277,7 @@ static void rnd_table_init() { } -static void init_noise_samples() { +static void init_noise_samples(void) { int i; int random_seed = 0; float delta = 1.0 / 16384.0; @@ -287,7 +288,7 @@ static void init_noise_samples() { } -static void qdm2_init_vlc() +static void qdm2_init_vlc(void) { init_vlc (&vlc_tab_level, 8, 24, vlc_tab_level_huffbits, 1, 1, @@ -401,9 +402,9 @@ static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) * @param length data length * @param value checksum value * - * @return 0 if checksum is ok + * @return 0 if checksum is OK */ -static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) { +static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { int i; for (i=0; i < length; i++) @@ -414,7 +415,7 @@ static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) { /** - * Fills a QDM2SubPacket structure with packet type, size, and data pointer + * Fills a QDM2SubPacket structure with packet type, size, and data pointer. * * @param gb bitreader context * @param sub_packet packet under analysis @@ -441,15 +442,15 @@ static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data } - av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n", + av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); } /** - * Return node pointer to first packet of requested type in list + * Return node pointer to first packet of requested type in list. * - * @param list list of subpacket to be scanned + * @param list list of subpackets to be scanned * @param type type of searched subpacket * @return node pointer for subpacket if found, else NULL */ @@ -465,8 +466,8 @@ static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int /** - * Replaces 8 elements with their average value - * Called by qdm2_decode_superblock before starting subblocks decoding + * Replaces 8 elements with their average value. + * Called by qdm2_decode_superblock before starting subblock decoding. * * @param q context */ @@ -494,8 +495,8 @@ static void average_quantized_coeffs (QDM2Context *q) /** - * Build subband samples with noise weighted by q->tone_level - * Called by synthfilt_build_sb_samples + * Build subband samples with noise weighted by q->tone_level. + * Called by synthfilt_build_sb_samples. * * @param q context * @param sb subband index @@ -518,14 +519,14 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) /** - * Called while processing data from subpackets 11 and 12 - * Used after making changes to coding_method array + * Called while processing data from subpackets 11 and 12. + * Used after making changes to coding_method array. * * @param sb subband index * @param channels number of channels * @param coding_method q->coding_method[0][0][0] */ - void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) +static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) { int j,k; int ch; @@ -538,7 +539,7 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) run = 1; case_val = 8; } else { - switch (switchtable[coding_method[ch][sb][j]]) { + switch (switchtable[coding_method[ch][sb][j]-8]) { case 0: run = 10; case_val = 10; break; case 1: run = 1; case_val = 16; break; case 2: run = 5; case_val = 24; break; @@ -657,7 +658,7 @@ static void fill_tone_level_array (QDM2Context *q, int flag) * c is built with data from subpacket 11 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples * - * @param tone_level_idx + * @param tone_level_idx * @param tone_level_idx_temp * @param coding_method q->coding_method[0][0][0] * @param nb_channels number of channels @@ -790,7 +791,7 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra * * @param q context * @param gb bitreader context - * @param length packet length in bit + * @param length packet length in bits * @param sb_min lower subband processed (sb_min included) * @param sb_max higher subband processed (sb_max excluded) */ @@ -916,7 +917,7 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; else samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); - + run = 1; break; @@ -968,14 +969,14 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l /** - * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]) + * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). * This is similar to process_subpacket_9, but for a single channel and for element [0] - * same VLC tables as process_subpacket_9 are used + * same VLC tables as process_subpacket_9 are used. * * @param q context * @param quantized_coeffs pointer to quantized_coeffs[ch][0] * @param gb bitreader context - * @param length packet length in bit + * @param length packet length in bits */ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) { @@ -995,10 +996,10 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext if (BITS_LEFT(length,gb) < 16) break; diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); - + for (k = 1; k <= run; k++) quantized_coeffs[i + k] = (level + ((k * diff) / run)); - + level += diff; i += run; } @@ -1012,7 +1013,7 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext * * @param q context * @param gb bitreader context - * @param length packet length in bit + * @param length packet length in bits */ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) { @@ -1114,7 +1115,7 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) * * @param q context * @param node pointer to node with packet - * @param length packet length in bit + * @param length packet length in bits */ static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) { @@ -1160,7 +1161,7 @@ static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length * * @param q context * @param node pointer to node with packet - * @param length packet length in bit + * @param length packet length in bits */ static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) { @@ -1205,7 +1206,7 @@ static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) /* - * Decode superblock, fill packet lists + * Decode superblock, fill packet lists. * * @param q context */ @@ -1274,7 +1275,7 @@ static void qdm2_decode_super_block (QDM2Context *q) break; } - /* decode sub packet */ + /* decode subpacket */ packet = &q->sub_packets[i]; qdm2_decode_sub_packet_header(&gb, packet); next_index = packet->size + get_bits_count(&gb) / 8; @@ -1291,10 +1292,10 @@ static void qdm2_decode_super_block (QDM2Context *q) packet_bytes -= sub_packet_size; - /* add sub packet to 'all sub packets' list */ + /* add subpacket to 'all subpackets' list */ q->sub_packet_list_A[i].packet = packet; - /* add sub packet to related list */ + /* add subpacket to related list */ if (packet->type == 8) { SAMPLES_NEEDED_2("packet type 8"); return; @@ -1430,16 +1431,16 @@ static void qdm2_decode_fft_packets (QDM2Context *q) if (q->sub_packet_list_B[0].packet == NULL) return; - /* reset minimum indices for FFT coefficients */ + /* reset minimum indexes for FFT coefficients */ q->fft_coefs_index = 0; for (i=0; i < 5; i++) q->fft_coefs_min_index[i] = -1; - /* process sub packets ordered by type, largest type first */ + /* process subpackets ordered by type, largest type first */ for (i = 0, max = 256; i < q->sub_packets_B; i++) { QDM2SubPacket *packet; - /* find sub packet with largest type less than max */ + /* find subpacket with largest type less than max */ for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) { value = q->sub_packet_list_B[j].packet->type; if (value > min && value < max) { @@ -1470,17 +1471,17 @@ static void qdm2_decode_fft_packets (QDM2Context *q) if (duration >= 0 && duration < 4) qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); } else if (type == 31) { - for (i=0; i < 4; i++) - qdm2_fft_decode_tones(q, i, &gb, unknown_flag); + for (j=0; j < 4; j++) + qdm2_fft_decode_tones(q, j, &gb, unknown_flag); } else if (type == 46) { - for (i=0; i < 6; i++) - q->fft_level_exp[i] = get_bits(&gb, 6); - for (i=0; i < 4; i++) - qdm2_fft_decode_tones(q, i, &gb, unknown_flag); + for (j=0; j < 6; j++) + q->fft_level_exp[j] = get_bits(&gb, 6); + for (j=0; j < 4; j++) + qdm2_fft_decode_tones(q, j, &gb, unknown_flag); } } // Loop on B packets - /* calculate maximum indices for FFT coefficients */ + /* calculate maximum indexes for FFT coefficients */ for (i = 0, j = -1; i < 5; i++) if (q->fft_coefs_min_index[i] >= 0) { if (j >= 0) @@ -1597,7 +1598,7 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 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.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; @@ -1619,7 +1620,7 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) float c, s, f0, f1, f2, f3; int i, j; - /* pre rotation (or something like that) */ + /* prerotation (or something like that) */ for (i=1; i < n2; i++) { j = (n - i); c = q->exptab[i].re; @@ -1690,12 +1691,12 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) * * @param q context */ -void qdm2_init(QDM2Context *q) { - static int inited = 0; +static void qdm2_init(QDM2Context *q) { + static int initialized = 0; - if (inited != 0) + if (initialized != 0) return; - inited = 1; + initialized = 1; qdm2_init_vlc(); ff_mpa_synth_init(mpa_window); @@ -1737,7 +1738,7 @@ static void dump_context(QDM2Context *q) for (i = q->fft_tone_start; i < q->fft_tone_end; i++) { FFTTone *t = &q->fft_tones[i]; - + av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); // PRINT(" level", t->level); @@ -1764,20 +1765,20 @@ static int qdm2_decode_init(AVCodecContext *avctx) int tmp_val, tmp, size; int i; float alpha; - + /* extradata parsing - + Structure: wave { frma (QDM2) QDCA QDCP } - + 32 size (including this field) 32 tag (=frma) 32 type (=QDM2 or QDMC) - + 32 size (including this field, in bytes) 32 tag (=QDCA) // maybe mandatory parameters 32 unknown (=1) @@ -1787,7 +1788,7 @@ static int qdm2_decode_init(AVCodecContext *avctx) 32 block size (=4096) 32 frame size (=256) (for one channel) 32 packet size (=1300) - + 32 size (including this field, in bytes) 32 tag (=QDCP) // maybe some tuneable parameters 32 float1 (=1.0) @@ -1834,7 +1835,7 @@ static int qdm2_decode_init(AVCodecContext *avctx) extradata += 8; extradata_size -= 8; - size = BE_32(extradata); + size = AV_RB32(extradata); if(size > extradata_size){ av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", @@ -1844,29 +1845,29 @@ static int qdm2_decode_init(AVCodecContext *avctx) extradata += 4; av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); - if (BE_32(extradata) != MKBETAG('Q','D','C','A')) { + if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); return -1; } extradata += 8; - avctx->channels = s->nb_channels = s->channels = BE_32(extradata); + avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); extradata += 4; - avctx->sample_rate = BE_32(extradata); + avctx->sample_rate = AV_RB32(extradata); extradata += 4; - avctx->bit_rate = BE_32(extradata); + avctx->bit_rate = AV_RB32(extradata); extradata += 4; - s->group_size = BE_32(extradata); + s->group_size = AV_RB32(extradata); extradata += 4; - s->fft_size = BE_32(extradata); + s->fft_size = AV_RB32(extradata); extradata += 4; - s->checksum_size = BE_32(extradata); + s->checksum_size = AV_RB32(extradata); extradata += 4; s->fft_order = av_log2(s->fft_size) + 1; @@ -1876,12 +1877,9 @@ static int qdm2_decode_init(AVCodecContext *avctx) s->group_order = av_log2(s->group_size) + 1; s->frame_size = s->group_size / 16; // 16 iterations per super block - if (s->fft_order == 8) - s->sub_sampling = 1; - else - s->sub_sampling = 2; + s->sub_sampling = s->fft_order - 7; s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); - + switch ((s->sub_sampling * 2 + s->channels - 1)) { case 0: tmp = 40; break; case 1: tmp = 48; break; @@ -1899,11 +1897,11 @@ static int qdm2_decode_init(AVCodecContext *avctx) s->cm_table_select = tmp_val; if (s->sub_sampling == 0) - tmp = 16000; + tmp = 7999; else tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; /* - 0: 16000 -> 1 + 0: 7999 -> 0 1: 20000 -> 2 2: 28000 -> 2 */ @@ -1914,8 +1912,11 @@ static int qdm2_decode_init(AVCodecContext *avctx) else s->coeff_per_sb_select = 2; - if (s->fft_order != 8 && s->fft_order != 9) + // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[] + if ((s->fft_order < 7) || (s->fft_order > 9)) { av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); + return -1; + } ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1); @@ -1926,7 +1927,7 @@ static int qdm2_decode_init(AVCodecContext *avctx) } qdm2_init(s); - + // dump_context(s); return 0; } @@ -1937,16 +1938,16 @@ static int qdm2_decode_close(AVCodecContext *avctx) QDM2Context *s = avctx->priv_data; ff_fft_end(&s->fft_ctx); - + return 0; } -void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) +static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) { int ch, i; const int frame_size = (q->frame_size * q->channels); - + /* select input buffer */ q->compressed_data = in; q->compressed_size = q->checksum_size; @@ -1960,11 +1961,11 @@ void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) /* decode block of QDM2 compressed data */ if (q->sub_packet == 0) { q->has_errors = 0; // zero it for a new super block - av_log(NULL,AV_LOG_DEBUG,"Super block follows\n"); + av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); qdm2_decode_super_block(q); } - /* parse sub packets */ + /* parse subpackets */ if (!q->has_errors) { if (q->sub_packet == 2) qdm2_decode_fft_packets(q); @@ -2004,12 +2005,14 @@ void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - uint8_t *buf, int buf_size) + const uint8_t *buf, int buf_size) { QDM2Context *s = avctx->priv_data; - 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); @@ -2035,4 +2038,5 @@ AVCodec qdm2_decoder = .init = qdm2_decode_init, .close = qdm2_decode_close, .decode = qdm2_decode_frame, + .long_name = "QDesign Music Codec 2", };