* You should have received a copy of the GNU Lesser General Public
* 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
+ * @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
#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 "mpegaudio.h"
#include "qdm2data.h"
/**
* 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 re;
+ float im;
+} QDM2Complex;
+
typedef struct {
float level;
- float *samples_im;
- float *samples_re;
- float *table;
+ QDM2Complex *complex;
+ const float *table;
int phase;
int phase_shift;
int duration;
} 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];
+ DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]);
} QDM2FFT;
/**
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];
/// 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];
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(void) {
+static av_cold void softclip_table_init(void) {
int i;
double dfl = SOFTCLIP_THRESHOLD - 32767;
float delta = 1.0 / -dfl;
// 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;
}
-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;
}
}
+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;
+ }
}
/* for floating point to fixed point conversion */
-static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
+static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
*
* @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++)
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;
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++)
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 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;
max = min;
/* check for errors (?) */
+ if (!packet)
+ return;
+
if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
return;
}
} // 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)
/* 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[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];
}
}
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));
}
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 */
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;
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;
}
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;
*
* @param q context
*/
-static void qdm2_init(QDM2Context *q) {
- static int inited = 0;
+static av_cold 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);
+ ff_mpa_synth_init(ff_mpa_synth_window);
softclip_table_init();
rnd_table_init();
init_noise_samples();
/**
* 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
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
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, IRDFT);
qdm2_init(s);
+ avctx->sample_fmt = 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;
}
-static 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);
static int qdm2_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
- 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)
.init = qdm2_decode_init,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
};
projects / ffmpeg / blobdiff