X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=797a464b841ad5a5cbb0cd916a0be8ae639cf6d0;hb=67e19a9e4b6b48ecedc6a64210ef6f69edce9119;hp=e1b67d0c19176945409acafe068d498623aa30ba;hpb=fd76c37fd9f564b4e979fbe20ecfcfad13f8b4f4;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index e1b67d0c191..797a464b841 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -23,7 +23,7 @@ */ /** - * @file qdm2.c + * @file libavcodec/qdm2.c * QDM2 decoder * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni * The decoder is not perfect yet, there are still some distortions @@ -36,13 +36,9 @@ #define ALT_BITSTREAM_READER_LE #include "avcodec.h" -#include "bitstream.h" +#include "get_bits.h" #include "dsputil.h" - -#ifdef CONFIG_MPEGAUDIO_HP -#define USE_HIGHPRECISION -#endif - +#include "fft.h" #include "mpegaudio.h" #include "qdm2data.h" @@ -102,10 +98,14 @@ typedef struct QDM2SubPNode { struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node } QDM2SubPNode; +typedef struct { + float re; + float im; +} QDM2Complex; + typedef struct { float level; - float *samples_im; - float *samples_re; + QDM2Complex *complex; const float *table; int phase; int phase_shift; @@ -123,14 +123,7 @@ typedef struct { } FFTCoefficient; typedef struct { - float re; - float im; -} QDM2Complex; - -typedef struct { - DECLARE_ALIGNED_16(QDM2Complex, complex[256 + 1]); - float samples_im[MPA_MAX_CHANNELS][256]; - float samples_re[MPA_MAX_CHANNELS][256]; + DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; } QDM2FFT; /** @@ -171,8 +164,7 @@ 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 @@ -181,9 +173,9 @@ typedef struct { float output_buffer[1024]; /// Synthesis filter - DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]); + DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; int synth_buf_offset[MPA_MAX_CHANNELS]; - DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]); + DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; /// Mixed temporary data used in decoding float tone_level[MPA_MAX_CHANNELS][30][64]; @@ -228,10 +220,8 @@ static uint8_t random_dequant_index[256][5]; static uint8_t random_dequant_type24[128][3]; static float noise_samples[128]; -static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); - -static void softclip_table_init(void) { +static av_cold void softclip_table_init(void) { int i; double dfl = SOFTCLIP_THRESHOLD - 32767; float delta = 1.0 / -dfl; @@ -241,7 +231,7 @@ static void softclip_table_init(void) { // random generated table -static void rnd_table_init(void) { +static av_cold void rnd_table_init(void) { int i,j; uint32_t ldw,hdw; uint64_t tmp64_1; @@ -277,7 +267,7 @@ static void rnd_table_init(void) { } -static void init_noise_samples(void) { +static av_cold void init_noise_samples(void) { int i; int random_seed = 0; float delta = 1.0 / 16384.0; @@ -287,76 +277,122 @@ static void init_noise_samples(void) { } } +static const uint16_t qdm2_vlc_offs[] = { + 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, +}; -static void qdm2_init_vlc(void) +static av_cold void qdm2_init_vlc(void) { - init_vlc (&vlc_tab_level, 8, 24, - vlc_tab_level_huffbits, 1, 1, - vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_diff, 8, 37, - vlc_tab_diff_huffbits, 1, 1, - vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_run, 5, 6, - vlc_tab_run_huffbits, 1, 1, - vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_level_exp_alt_vlc, 8, 28, - fft_level_exp_alt_huffbits, 1, 1, - fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_level_exp_vlc, 8, 20, - fft_level_exp_huffbits, 1, 1, - fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_stereo_exp_vlc, 6, 7, - fft_stereo_exp_huffbits, 1, 1, - fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&fft_stereo_phase_vlc, 6, 9, - fft_stereo_phase_huffbits, 1, 1, - fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, - vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, - vlc_tab_tone_level_idx_mid_huffbits, 1, 1, - vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, - vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_type30, 6, 9, - vlc_tab_type30_huffbits, 1, 1, - vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_type34, 5, 10, - vlc_tab_type34_huffbits, 1, 1, - vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, - vlc_tab_fft_tone_offset_0_huffbits, 1, 1, - vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, - vlc_tab_fft_tone_offset_1_huffbits, 1, 1, - vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, - vlc_tab_fft_tone_offset_2_huffbits, 1, 1, - vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, - vlc_tab_fft_tone_offset_3_huffbits, 1, 1, - vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); - - init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, - vlc_tab_fft_tone_offset_4_huffbits, 1, 1, - vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); + static int vlcs_initialized = 0; + static VLC_TYPE qdm2_table[3838][2]; + + if (!vlcs_initialized) { + + vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; + vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; + init_vlc (&vlc_tab_level, 8, 24, + vlc_tab_level_huffbits, 1, 1, + vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; + vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; + init_vlc (&vlc_tab_diff, 8, 37, + vlc_tab_diff_huffbits, 1, 1, + vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; + vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; + init_vlc (&vlc_tab_run, 5, 6, + vlc_tab_run_huffbits, 1, 1, + vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; + fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; + init_vlc (&fft_level_exp_alt_vlc, 8, 28, + fft_level_exp_alt_huffbits, 1, 1, + fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + + fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; + fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; + init_vlc (&fft_level_exp_vlc, 8, 20, + fft_level_exp_huffbits, 1, 1, + fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; + fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; + init_vlc (&fft_stereo_exp_vlc, 6, 7, + fft_stereo_exp_huffbits, 1, 1, + fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; + fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; + init_vlc (&fft_stereo_phase_vlc, 6, 9, + fft_stereo_phase_huffbits, 1, 1, + fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; + vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; + init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, + vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, + vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; + vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; + init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, + vlc_tab_tone_level_idx_mid_huffbits, 1, 1, + vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; + vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; + init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, + vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, + vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; + vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; + init_vlc (&vlc_tab_type30, 6, 9, + vlc_tab_type30_huffbits, 1, 1, + vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; + vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; + init_vlc (&vlc_tab_type34, 5, 10, + vlc_tab_type34_huffbits, 1, 1, + vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; + vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; + init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, + vlc_tab_fft_tone_offset_0_huffbits, 1, 1, + vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; + vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; + init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, + vlc_tab_fft_tone_offset_1_huffbits, 1, 1, + vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; + vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; + init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, + vlc_tab_fft_tone_offset_2_huffbits, 1, 1, + vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; + vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; + init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, + vlc_tab_fft_tone_offset_3_huffbits, 1, 1, + vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; + vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; + init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, + vlc_tab_fft_tone_offset_4_huffbits, 1, 1, + vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + + vlcs_initialized=1; + } } @@ -716,8 +752,7 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra for (sb = 0; sb < 30; sb++) for (j = 0; j < 64; j++) acc += tone_level_idx_temp[ch][sb][j]; - if (acc) - tmp = c * 256 / (acc & 0xffff); + multres = 0x66666667 * (acc * 10); esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); for (ch = 0; ch < nb_channels; ch++) @@ -1512,10 +1547,10 @@ static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) /* generate FFT coefficients for tone */ if (tone->duration >= 3 || tone->cutoff >= 3) { - tone->samples_im[0] += c.im; - tone->samples_re[0] += c.re; - tone->samples_im[1] -= c.im; - tone->samples_re[1] -= c.re; + tone->complex[0].im += c.im; + tone->complex[0].re += c.re; + tone->complex[1].im -= c.im; + tone->complex[1].re -= c.re; } else { f[1] = -tone->table[4]; f[0] = tone->table[3] - tone->table[0]; @@ -1524,12 +1559,12 @@ static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) f[4] = tone->table[0] - tone->table[1]; f[5] = tone->table[2]; for (i = 0; i < 2; i++) { - tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i]; - tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); + tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; + tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); } for (i = 0; i < 4; i++) { - tone->samples_re[i] += c.re * f[i+2]; - tone->samples_im[i] += c.im * f[i+2]; + tone->complex[i].re += c.re * f[i+2]; + tone->complex[i].im += c.im * f[i+2]; } } @@ -1547,8 +1582,7 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) const double iscale = 0.25 * M_PI; for (ch = 0; ch < q->channels; ch++) { - memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float)); - memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float)); + memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); } @@ -1566,10 +1600,10 @@ 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 */ @@ -1599,8 +1633,7 @@ 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.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); @@ -1617,37 +1650,14 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) { - const int n = 1 << (q->fft_order - 1); - const int n2 = n >> 1; - const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f; - float c, s, f0, f1, f2, f3; - int i, j; - - /* prerotation (or something like that) */ - for (i=1; i < n2; i++) { - j = (n - i); - c = q->exptab[i].re; - s = -q->exptab[i].im; - f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain; - f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain; - f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain; - f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain; - q->fft.complex[i].re = s * f0 - c * f1 + f2; - q->fft.complex[i].im = c * f0 + s * f1 + f3; - q->fft.complex[j].re = -s * f0 + c * f1 + f2; - q->fft.complex[j].im = c * f0 + s * f1 - f3; - } - - q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0; - q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0; - q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0; - q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0; - - ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex); - ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex); + const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; + int i; + q->fft.complex[channel][0].re *= 2.0f; + q->fft.complex[channel][0].im = 0.0f; + ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); /* add samples to output buffer */ for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) - q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i]; + q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; } @@ -1673,7 +1683,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) for (i = 0; i < 8; i++) { ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), - mpa_window, &dither_state, + ff_mpa_synth_window, &dither_state, samples_ptr, q->nb_channels, q->sb_samples[ch][(8 * index) + i]); samples_ptr += 32 * q->nb_channels; @@ -1694,7 +1704,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) * * @param q context */ -static void qdm2_init(QDM2Context *q) { +static av_cold void qdm2_init(QDM2Context *q) { static int initialized = 0; if (initialized != 0) @@ -1702,7 +1712,7 @@ static void qdm2_init(QDM2Context *q) { initialized = 1; qdm2_init_vlc(); - ff_mpa_synth_init(mpa_window); + ff_mpa_synth_init(ff_mpa_synth_window); softclip_table_init(); rnd_table_init(); init_noise_samples(); @@ -1760,14 +1770,12 @@ static void dump_context(QDM2Context *q) /** * Init parameters from codec extradata */ -static int qdm2_decode_init(AVCodecContext *avctx) +static av_cold int qdm2_decode_init(AVCodecContext *avctx) { QDM2Context *s = avctx->priv_data; uint8_t *extradata; int extradata_size; int tmp_val, tmp, size; - int i; - float alpha; /* extradata parsing @@ -1871,7 +1879,6 @@ static int qdm2_decode_init(AVCodecContext *avctx) extradata += 4; s->checksum_size = AV_RB32(extradata); - extradata += 4; s->fft_order = av_log2(s->fft_size) + 1; s->fft_frame_size = 2 * s->fft_size; // complex has two floats @@ -1915,19 +1922,13 @@ static int qdm2_decode_init(AVCodecContext *avctx) else s->coeff_per_sb_select = 2; - // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[] + // Fail on unknown fft order if ((s->fft_order < 7) || (s->fft_order > 9)) { av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); return -1; } - ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1); - - for (i = 1; i < (1 << (s->fft_order - 2)); i++) { - alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1)); - s->exptab[i].re = cos(alpha); - s->exptab[i].im = sin(alpha); - } + ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); qdm2_init(s); @@ -1938,11 +1939,11 @@ static int qdm2_decode_init(AVCodecContext *avctx) } -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; } @@ -2010,8 +2011,10 @@ static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - const uint8_t *buf, int buf_size) + AVPacket *avpkt) { + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; QDM2Context *s = avctx->priv_data; if(!buf)