diff options
Diffstat (limited to 'libavcodec/qdm2.c')
-rw-r--r-- | libavcodec/qdm2.c | 301 |
1 files changed, 91 insertions, 210 deletions
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index b33e7c6..074aafd 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -5,20 +5,20 @@ * Copyright (c) 2005 Alex Beregszaszi * Copyright (c) 2005 Roberto Togni * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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 2.1 of the License, or (at your option) any later version. * - * Libav 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -44,10 +44,8 @@ #include "mpegaudiodsp.h" #include "mpegaudio.h" -#include "qdm2data.h" #include "qdm2_tablegen.h" - #define QDM2_LIST_ADD(list, size, packet) \ do { \ if (size > 0) { \ @@ -165,7 +163,7 @@ typedef struct QDM2Context { /// I/O data const uint8_t *compressed_data; int compressed_size; - float output_buffer[QDM2_MAX_FRAME_SIZE * 2]; + float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; /// Synthesis filter MPADSPContext mpadsp; @@ -194,173 +192,11 @@ typedef struct QDM2Context { int noise_idx; ///< index for dithering noise table } QDM2Context; - -static VLC vlc_tab_level; -static VLC vlc_tab_diff; -static VLC vlc_tab_run; -static VLC fft_level_exp_alt_vlc; -static VLC fft_level_exp_vlc; -static VLC fft_stereo_exp_vlc; -static VLC fft_stereo_phase_vlc; -static VLC vlc_tab_tone_level_idx_hi1; -static VLC vlc_tab_tone_level_idx_mid; -static VLC vlc_tab_tone_level_idx_hi2; -static VLC vlc_tab_type30; -static VLC vlc_tab_type34; -static VLC vlc_tab_fft_tone_offset[5]; - -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 av_cold void qdm2_init_vlc(void) -{ - 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); -} - -static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) +static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth) { int value; @@ -372,7 +208,14 @@ static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) /* stage-3, optional */ if (flag) { - int tmp = vlc_stage3_values[value]; + int tmp; + + if (value >= 60) { + av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); + return 0; + } + + tmp= vlc_stage3_values[value]; if ((value & ~3) > 0) tmp += get_bits(gb, (value >> 2)); @@ -382,7 +225,7 @@ static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) return value; } -static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth) +static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth) { int value = qdm2_get_vlc(gb, vlc, 0, depth); @@ -691,7 +534,8 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx, if (!superblocktype_2_3) { /* This case is untested, no samples available */ - SAMPLES_NEEDED + avpriv_request_sample(NULL, "!superblocktype_2_3"); + return; for (ch = 0; ch < nb_channels; ch++) for (sb = 0; sb < 30; sb++) { for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer @@ -802,7 +646,7 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx, * @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, +static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) { int sb, j, k, n, ch, run, channels; @@ -810,14 +654,15 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, int type34_first; float type34_div = 0; float type34_predictor; - float samples[10], sign_bits[16]; + float samples[10]; + int sign_bits[16] = {0}; if (length == 0) { // If no data use noise for (sb=sb_min; sb < sb_max; sb++) build_sb_samples_from_noise(q, sb); - return; + return 0; } for (sb = sb_min; sb < sb_max; sb++) { @@ -841,6 +686,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, if (fix_coding_method_array(sb, q->nb_channels, q->coding_method)) { + av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); build_sb_samples_from_noise(q, sb); continue; } @@ -865,6 +711,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, } } else { n = get_bits(gb, 8); + if (n >= 243) { + av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); + return AVERROR_INVALIDDATA; + } + for (k = 0; k < 5; k++) samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; } @@ -901,6 +752,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, } } else { n = get_bits (gb, 8); + if (n >= 243) { + av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); + return AVERROR_INVALIDDATA; + } + for (k = 0; k < 5; k++) samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; } @@ -914,6 +770,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, case 24: if (get_bits_left(gb) >= 7) { n = get_bits(gb, 7); + if (n >= 125) { + av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); + return AVERROR_INVALIDDATA; + } + for (k = 0; k < 3; k++) samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; } else { @@ -926,10 +787,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, case 30: if (get_bits_left(gb) >= 4) { unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); - if (index < FF_ARRAY_ELEMS(type30_dequant)) { - samples[0] = type30_dequant[index]; - } else - samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); + if (index >= FF_ARRAY_ELEMS(type30_dequant)) { + av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index); + return AVERROR_INVALIDDATA; + } + samples[0] = type30_dequant[index]; } else samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); @@ -945,11 +807,12 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, type34_first = 0; } else { unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); - if (index < FF_ARRAY_ELEMS(type34_delta)) { - samples[0] = type34_delta[index] / type34_div + type34_predictor; - type34_predictor = samples[0]; - } else - samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); + if (index >= FF_ARRAY_ELEMS(type34_delta)) { + av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index); + return AVERROR_INVALIDDATA; + } + samples[0] = type34_delta[index] / type34_div + type34_predictor; + type34_predictor = samples[0]; } } else { samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); @@ -986,6 +849,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, } // j loop } // channel loop } // subband loop + return 0; } /** @@ -998,24 +862,27 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, * @param quantized_coeffs pointer to quantized_coeffs[ch][0] * @param gb bitreader context */ -static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs, +static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, GetBitContext *gb) { int i, k, run, level, diff; if (get_bits_left(gb) < 16) - return; + return -1; level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); quantized_coeffs[0] = level; for (i = 0; i < 7; ) { if (get_bits_left(gb) < 16) - break; + return -1; run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; + if (i + run >= 8) + return -1; + if (get_bits_left(gb) < 16) - break; + return -1; diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); for (k = 1; k <= run; k++) @@ -1024,6 +891,7 @@ static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs, level += diff; i += run; } + return 0; } /** @@ -1098,7 +966,7 @@ static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb) * @param q context * @param node pointer to node with packet */ -static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) +static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; int i, j, k, n, ch, run, level, diff; @@ -1116,6 +984,9 @@ static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); + if (j + run >= 8) + return -1; + for (k = 1; k <= run; k++) q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); @@ -1127,6 +998,8 @@ static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) for (ch = 0; ch < q->nb_channels; ch++) for (i = 0; i < 8; i++) q->quantized_coeffs[ch][0][i] = 0; + + return 0; } /** @@ -1196,7 +1069,7 @@ static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); } -/* +/** * Process new subpackets for synthesis filter * * @param q context @@ -1229,7 +1102,7 @@ static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) process_subpacket_12(q, NULL); } -/* +/** * Decode superblock, fill packet lists. * * @param q context @@ -1389,9 +1262,14 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration, local_int_10 = 1 << (q->group_order - duration - 1); offset = 1; - while (1) { + while (get_bits_left(gb)>0) { if (q->superblocktype_2_3) { while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { + if (get_bits_left(gb)<0) { + if(local_int_4 < q->group_size) + av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n"); + return; + } offset = 1; if (n == 0) { local_int_4 += local_int_10; @@ -1704,12 +1582,19 @@ static void qdm2_synthesis_filter(QDM2Context *q, int index) * * @param q context */ -static av_cold void qdm2_init_static_data(AVCodec *codec) { +static av_cold void qdm2_init_static_data(void) { + static int done; + + if(done) + return; + qdm2_init_vlc(); ff_mpa_synth_init_float(ff_mpa_synth_window_float); softclip_table_init(); rnd_table_init(); init_noise_samples(); + + done = 1; } /** @@ -1722,6 +1607,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) int extradata_size; int tmp_val, tmp, size; + qdm2_init_static_data(); + /* extradata parsing Structure: @@ -1810,8 +1697,10 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); extradata += 4; - if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) + if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) { + av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); return AVERROR_INVALIDDATA; + } avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO; @@ -1838,6 +1727,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) // 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; @@ -1860,18 +1750,9 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; s->cm_table_select = tmp_val; - if (s->sub_sampling == 0) - tmp = 7999; - else - tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; - /* - 0: 7999 -> 0 - 1: 20000 -> 2 - 2: 28000 -> 2 - */ - if (tmp < 8000) + if (avctx->bit_rate <= 8000) s->coeff_per_sb_select = 0; - else if (tmp <= 16000) + else if (avctx->bit_rate < 16000) s->coeff_per_sb_select = 1; else s->coeff_per_sb_select = 2; @@ -1908,6 +1789,9 @@ 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); + if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) + return -1; + /* select input buffer */ q->compressed_data = in; q->compressed_size = q->checksum_size; @@ -1979,10 +1863,8 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *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"); + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; - } out = (int16_t *)frame->data[0]; for (i = 0; i < 16; i++) { @@ -2003,7 +1885,6 @@ AVCodec ff_qdm2_decoder = { .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, |