* 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
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
*/
/**
/**
* 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;
} 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;
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 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;
// 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;
}
-static void init_noise_samples() {
+static void init_noise_samples(void) {
int i;
int random_seed = 0;
float delta = 1.0 / 16384.0;
}
-static void qdm2_init_vlc()
+static void qdm2_init_vlc(void)
{
init_vlc (&vlc_tab_level, 8, 24,
vlc_tab_level_huffbits, 1, 1,
*
* @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++)
* @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;
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;
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;
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)
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;
*
* @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);
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 += 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;
}
-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)
+ 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);
.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