* Copyright (c) 2005 Alex Beregszaszi
* Copyright (c) 2005 Roberto Togni
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* @file
* QDM2 decoder
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
+ *
* The decoder is not perfect yet, there are still some distortions
* especially on files encoded with 16 or 8 subbands.
*/
#include <stddef.h>
#include <stdio.h>
-#define ALT_BITSTREAM_READER_LE
+#include "libavutil/channel_layout.h"
+
+#define BITSTREAM_READER_LE
#include "avcodec.h"
-#include "get_bits.h"
-#include "dsputil.h"
-#include "fft.h"
+#include "bitstream.h"
+#include "internal.h"
#include "mpegaudio.h"
+#include "mpegaudiodsp.h"
+#include "rdft.h"
#include "qdm2data.h"
#include "qdm2_tablegen.h"
-#undef NDEBUG
-#include <assert.h>
-
#define QDM2_LIST_ADD(list, size, packet) \
do { \
#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
-#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
-
#define SAMPLES_NEEDED \
av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
#define SAMPLES_NEEDED_2(why) \
av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
+#define QDM2_MAX_FRAME_SIZE 512
typedef int8_t sb_int8_array[2][30][64];
/**
* Subpacket
*/
-typedef struct {
+typedef struct QDM2SubPacket {
int type; ///< subpacket type
unsigned int size; ///< subpacket size
const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
} QDM2SubPNode;
-typedef struct {
+typedef struct QDM2Complex {
float re;
float im;
} QDM2Complex;
-typedef struct {
+typedef struct FFTTone {
float level;
QDM2Complex *complex;
const float *table;
short cutoff;
} FFTTone;
-typedef struct {
+typedef struct FFTCoefficient {
int16_t sub_packet;
uint8_t channel;
int16_t offset;
uint8_t phase;
} FFTCoefficient;
-typedef struct {
- DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
+typedef struct QDM2FFT {
+ DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
* QDM2 decoder context
*/
-typedef struct {
+typedef struct QDM2Context {
/// Parameters from codec header, do not change during playback
int nb_channels; ///< number of channels
int channels; ///< number of channels
/// Parameters built from header parameters, do not change during playback
int group_order; ///< order of frame group
int fft_order; ///< order of FFT (actually fftorder+1)
- int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
int frame_size; ///< size of data frame
int frequency_range;
int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
/// I/O data
const uint8_t *compressed_data;
int compressed_size;
- float output_buffer[1024];
+ float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
/// Synthesis filter
- DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
+ MPADSPContext mpadsp;
+ DECLARE_ALIGNED(32, float, 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(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+ DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
/// Mixed temporary data used in decoding
float tone_level[MPA_MAX_CHANNELS][30][64];
} QDM2Context;
-static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
-
static VLC vlc_tab_level;
static VLC vlc_tab_diff;
static VLC vlc_tab_run;
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 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;
- }
+ 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);
}
-
-/* for floating point to fixed point conversion */
-static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
-
-
-static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
+static int qdm2_get_vlc(BitstreamContext *bc, VLC *vlc, int flag, int depth)
{
int value;
- value = get_vlc2(gb, vlc->table, vlc->bits, depth);
+ value = bitstream_read_vlc(bc, vlc->table, vlc->bits, depth);
/* stage-2, 3 bits exponent escape sequence */
if (value-- == 0)
- value = get_bits (gb, get_bits (gb, 3) + 1);
+ value = bitstream_read(bc, bitstream_read(bc, 3) + 1);
/* stage-3, optional */
if (flag) {
int tmp = vlc_stage3_values[value];
if ((value & ~3) > 0)
- tmp += get_bits (gb, (value >> 2));
+ tmp += bitstream_read(bc, value >> 2);
value = tmp;
}
return value;
}
-
-static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
+static int qdm2_get_se_vlc(VLC *vlc, BitstreamContext *bc, int depth)
{
- int value = qdm2_get_vlc (gb, vlc, 0, depth);
+ int value = qdm2_get_vlc(bc, vlc, 0, depth);
return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
}
-
/**
* QDM2 checksum
*
- * @param data pointer to data to be checksum'ed
+ * @param data pointer to data to be checksummed
* @param length data length
* @param value checksum value
*
* @return 0 if checksum is OK
*/
-static uint16_t qdm2_packet_checksum (const 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++)
+ for (i = 0; i < length; i++)
value -= data[i];
return (uint16_t)(value & 0xffff);
}
-
/**
* Fill a QDM2SubPacket structure with packet type, size, and data pointer.
*
- * @param gb bitreader context
+ * @param bc bitreader context
* @param sub_packet packet under analysis
*/
-static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
+static void qdm2_decode_sub_packet_header(BitstreamContext *bc,
+ QDM2SubPacket *sub_packet)
{
- sub_packet->type = get_bits (gb, 8);
+ sub_packet->type = bitstream_read(bc, 8);
if (sub_packet->type == 0) {
sub_packet->size = 0;
sub_packet->data = NULL;
} else {
- sub_packet->size = get_bits (gb, 8);
+ sub_packet->size = bitstream_read(bc, 8);
- if (sub_packet->type & 0x80) {
- sub_packet->size <<= 8;
- sub_packet->size |= get_bits (gb, 8);
- sub_packet->type &= 0x7f;
- }
+ if (sub_packet->type & 0x80) {
+ sub_packet->size <<= 8;
+ sub_packet->size |= bitstream_read(bc, 8);
+ sub_packet->type &= 0x7f;
+ }
- if (sub_packet->type == 0x7f)
- sub_packet->type |= (get_bits (gb, 8) << 8);
+ if (sub_packet->type == 0x7f)
+ sub_packet->type |= bitstream_read(bc, 8) << 8;
- sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
+ // FIXME: this depends on bitreader-internal data
+ sub_packet->data = &bc->buffer[bitstream_tell(bc) / 8];
}
- 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);
+ av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
+ sub_packet->type, sub_packet->size, bitstream_tell(bc) / 8);
}
-
/**
* Return node pointer to first packet of requested type in list.
*
* @param type type of searched subpacket
* @return node pointer for subpacket if found, else NULL
*/
-static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
+static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
+ int type)
{
- while (list != NULL && list->packet != NULL) {
+ while (list && list->packet) {
if (list->packet->type == type)
return list;
list = list->next;
return NULL;
}
-
/**
* Replace 8 elements with their average value.
* Called by qdm2_decode_superblock before starting subblock decoding.
*
* @param q context
*/
-static void average_quantized_coeffs (QDM2Context *q)
+static void average_quantized_coeffs(QDM2Context *q)
{
int i, j, n, ch, sum;
if (sum > 0)
sum--;
- for (j=0; j < 8; j++)
+ for (j = 0; j < 8; j++)
q->quantized_coeffs[ch][i][j] = sum;
}
}
-
/**
* Build subband samples with noise weighted by q->tone_level.
* Called by synthfilt_build_sb_samples.
* @param q context
* @param sb subband index
*/
-static void build_sb_samples_from_noise (QDM2Context *q, int sb)
+static void build_sb_samples_from_noise(QDM2Context *q, int sb)
{
int ch, j;
if (!q->nb_channels)
return;
- for (ch = 0; ch < q->nb_channels; ch++)
+ for (ch = 0; ch < q->nb_channels; ch++) {
for (j = 0; j < 64; j++) {
- q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
- q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
+ q->sb_samples[ch][j * 2][sb] =
+ SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
+ q->sb_samples[ch][j * 2 + 1][sb] =
+ SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
}
+ }
}
-
/**
* Called while processing data from subpackets 11 and 12.
* Used after making changes to coding_method array.
* @param channels number of channels
* @param coding_method q->coding_method[0][0][0]
*/
-static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
+static int fix_coding_method_array(int sb, int channels,
+ sb_int8_array coding_method)
{
- int j,k;
+ int j, k;
int ch;
int run, case_val;
- 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};
for (ch = 0; ch < channels; ch++) {
for (j = 0; j < 64; ) {
- if((coding_method[ch][sb][j] - 8) > 22) {
- run = 1;
+ if (coding_method[ch][sb][j] < 8)
+ return -1;
+ if ((coding_method[ch][sb][j] - 8) > 22) {
+ run = 1;
case_val = 8;
} else {
- 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;
- case 3: run = 3; case_val = 30; break;
- case 4: run = 1; case_val = 30; break;
- case 5: run = 1; case_val = 8; break;
- default: run = 1; case_val = 8; break;
+ 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;
+ case 3: run = 3;
+ case_val = 30;
+ break;
+ case 4: run = 1;
+ case_val = 30;
+ break;
+ case 5: run = 1;
+ case_val = 8;
+ break;
+ default: run = 1;
+ case_val = 8;
+ break;
}
}
- for (k = 0; k < run; k++)
- if (j + k < 128)
- if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
+ for (k = 0; k < run; k++) {
+ if (j + k < 128) {
+ if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
if (k > 0) {
- SAMPLES_NEEDED
+ SAMPLES_NEEDED
//not debugged, almost never used
- memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
- memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
+ memset(&coding_method[ch][sb][j + k], case_val,
+ k *sizeof(int8_t));
+ memset(&coding_method[ch][sb][j + k], case_val,
+ 3 * sizeof(int8_t));
}
+ }
+ }
+ }
j += run;
}
}
+ return 0;
}
-
/**
* Related to synthesis filter
* Called by process_subpacket_10
* @param q context
* @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
*/
-static void fill_tone_level_array (QDM2Context *q, int flag)
+static void fill_tone_level_array(QDM2Context *q, int flag)
{
int i, sb, ch, sb_used;
int tmp, tab;
- // This should never happen
- if (q->nb_channels <= 0)
- return;
-
for (ch = 0; ch < q->nb_channels; ch++)
for (sb = 0; sb < 30; sb++)
for (i = 0; i < 8; i++) {
}
}
}
-
- return;
}
-
/**
* Related to synthesis filter
* Called by process_subpacket_11
* 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
+ * 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_temp
* @param superblocktype_2_3 flag based on superblock packet type
* @param cm_table_select q->cm_table_select
*/
-static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
- sb_int8_array coding_method, int nb_channels,
- int c, int superblocktype_2_3, int cm_table_select)
+static void fill_coding_method_array(sb_int8_array tone_level_idx,
+ sb_int8_array tone_level_idx_temp,
+ sb_int8_array coding_method,
+ int nb_channels,
+ int c, int superblocktype_2_3,
+ int cm_table_select)
{
int ch, sb, j;
int tmp, acc, esp_40, comp;
int add1, add2, add3, add4;
int64_t multres;
- // This should never happen
- if (nb_channels <= 0)
- return;
-
if (!superblocktype_2_3) {
/* This case is untested, no samples available */
SAMPLES_NEEDED
}
tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
}
- acc = 0;
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- acc += tone_level_idx_temp[ch][sb][j];
-
- multres = 0x66666667 * (acc * 10);
- esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++) {
- comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
- if (comp < 0)
- comp += 0xff;
- comp /= 256; // signed shift
- switch(sb) {
- case 0:
- if (comp < 30)
- comp = 30;
- comp += 15;
- break;
- case 1:
- if (comp < 24)
- comp = 24;
- comp += 10;
- break;
- case 2:
- case 3:
- case 4:
- if (comp < 16)
- comp = 16;
- }
- if (comp <= 5)
- tmp = 0;
- else if (comp <= 10)
- tmp = 10;
- else if (comp <= 16)
- tmp = 16;
- else if (comp <= 24)
- tmp = -1;
- else
- tmp = 0;
- coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+
+ acc = 0;
+ for (ch = 0; ch < nb_channels; ch++)
+ for (sb = 0; sb < 30; sb++)
+ for (j = 0; j < 64; j++)
+ acc += tone_level_idx_temp[ch][sb][j];
+
+ multres = 0x66666667LL * (acc * 10);
+ esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
+ for (ch = 0; ch < nb_channels; ch++)
+ for (sb = 0; sb < 30; sb++)
+ for (j = 0; j < 64; j++) {
+ comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
+ if (comp < 0)
+ comp += 0xff;
+ comp /= 256; // signed shift
+ switch(sb) {
+ case 0:
+ if (comp < 30)
+ comp = 30;
+ comp += 15;
+ break;
+ case 1:
+ if (comp < 24)
+ comp = 24;
+ comp += 10;
+ break;
+ case 2:
+ case 3:
+ case 4:
+ if (comp < 16)
+ comp = 16;
}
+ if (comp <= 5)
+ tmp = 0;
+ else if (comp <= 10)
+ tmp = 10;
+ else if (comp <= 16)
+ tmp = 16;
+ else if (comp <= 24)
+ tmp = -1;
+ else
+ tmp = 0;
+ coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+ }
+ for (sb = 0; sb < 30; sb++)
+ fix_coding_method_array(sb, nb_channels, coding_method);
+ for (ch = 0; ch < nb_channels; ch++)
for (sb = 0; sb < 30; sb++)
- fix_coding_method_array(sb, nb_channels, coding_method);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- if (sb >= 10) {
- if (coding_method[ch][sb][j] < 10)
- coding_method[ch][sb][j] = 10;
+ for (j = 0; j < 64; j++)
+ if (sb >= 10) {
+ if (coding_method[ch][sb][j] < 10)
+ coding_method[ch][sb][j] = 10;
+ } else {
+ if (sb >= 2) {
+ if (coding_method[ch][sb][j] < 16)
+ coding_method[ch][sb][j] = 16;
} else {
- if (sb >= 2) {
- if (coding_method[ch][sb][j] < 16)
- coding_method[ch][sb][j] = 16;
- } else {
- if (coding_method[ch][sb][j] < 30)
- coding_method[ch][sb][j] = 30;
- }
+ if (coding_method[ch][sb][j] < 30)
+ coding_method[ch][sb][j] = 30;
}
+ }
} else { // superblocktype_2_3 != 0
for (ch = 0; ch < nb_channels; ch++)
for (sb = 0; sb < 30; sb++)
for (j = 0; j < 64; j++)
coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
}
-
- return;
}
-
/**
- *
- * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
- * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
+ * Called by process_subpacket_11 to process more data from subpacket 11
+ * with sb 0-8.
+ * Called by process_subpacket_12 to process data from subpacket 12 with
+ * sb 8-sb_used.
*
* @param q context
- * @param gb bitreader context
+ * @param bc bitreader context
* @param length packet length in bits
* @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, int length, int sb_min, int sb_max)
+static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
+ int length, int sb_min, int sb_max)
{
int sb, j, k, n, ch, run, channels;
- int joined_stereo, zero_encoding, chs;
+ int joined_stereo, zero_encoding;
int type34_first;
float type34_div = 0;
float type34_predictor;
if (length == 0) {
// If no data use noise
for (sb=sb_min; sb < sb_max; sb++)
- build_sb_samples_from_noise (q, sb);
+ build_sb_samples_from_noise(q, sb);
return;
}
for (sb = sb_min; sb < sb_max; sb++) {
- FIX_NOISE_IDX(q->noise_idx);
-
channels = q->nb_channels;
if (q->nb_channels <= 1 || sb < 12)
else if (sb >= 24)
joined_stereo = 1;
else
- joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
+ joined_stereo = (bitstream_bits_left(bc) >= 1) ? bitstream_read_bit(bc) : 0;
if (joined_stereo) {
- if (BITS_LEFT(length,gb) >= 16)
+ if (bitstream_bits_left(bc) >= 16)
for (j = 0; j < 16; j++)
- sign_bits[j] = get_bits1 (gb);
+ sign_bits[j] = bitstream_read_bit(bc);
for (j = 0; j < 64; j++)
if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
- fix_coding_method_array(sb, q->nb_channels, q->coding_method);
+ if (fix_coding_method_array(sb, q->nb_channels,
+ q->coding_method)) {
+ build_sb_samples_from_noise(q, sb);
+ continue;
+ }
channels = 1;
}
for (ch = 0; ch < channels; ch++) {
- zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
+ FIX_NOISE_IDX(q->noise_idx);
+ zero_encoding = (bitstream_bits_left(bc) >= 1) ? bitstream_read_bit(bc) : 0;
type34_predictor = 0.0;
type34_first = 1;
for (j = 0; j < 128; ) {
switch (q->coding_method[ch][sb][j / 2]) {
case 8:
- if (BITS_LEFT(length,gb) >= 10) {
+ if (bitstream_bits_left(bc) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + 2 * k) >= 128)
break;
- samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
+ samples[2 * k] = bitstream_read_bit(bc) ? dequant_1bit[joined_stereo][2 * bitstream_read_bit(bc)] : 0;
}
} else {
- n = get_bits(gb, 8);
+ n = bitstream_read(bc, 8);
for (k = 0; k < 5; k++)
samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
}
break;
case 10:
- if (BITS_LEFT(length,gb) >= 1) {
+ if (bitstream_bits_left(bc) >= 1) {
float f = 0.81;
- if (get_bits1(gb))
+ if (bitstream_read_bit(bc))
f = -f;
f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
samples[0] = f;
break;
case 16:
- if (BITS_LEFT(length,gb) >= 10) {
+ if (bitstream_bits_left(bc) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + k) >= 128)
break;
- samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
+ samples[k] = (bitstream_read_bit(bc) == 0) ? 0 : dequant_1bit[joined_stereo][2 * bitstream_read_bit(bc)];
}
} else {
- n = get_bits (gb, 8);
+ n = bitstream_read (bc, 8);
for (k = 0; k < 5; k++)
samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
}
break;
case 24:
- if (BITS_LEFT(length,gb) >= 7) {
- n = get_bits(gb, 7);
+ if (bitstream_bits_left(bc) >= 7) {
+ n = bitstream_read(bc, 7);
for (k = 0; k < 3; k++)
samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
} else {
break;
case 30:
- if (BITS_LEFT(length,gb) >= 4)
- samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
- else
+ if (bitstream_bits_left(bc) >= 4) {
+ unsigned index = qdm2_get_vlc(bc, &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);
+ } else
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
run = 1;
break;
case 34:
- if (BITS_LEFT(length,gb) >= 7) {
+ if (bitstream_bits_left(bc) >= 7) {
if (type34_first) {
- type34_div = (float)(1 << get_bits(gb, 2));
- samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
+ type34_div = (float)(1 << bitstream_read(bc, 2));
+ samples[0] = ((float)bitstream_read(bc, 5) - 16.0) / 15.0;
type34_predictor = samples[0];
type34_first = 0;
} else {
- samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
- type34_predictor = samples[0];
+ unsigned index = qdm2_get_vlc(bc, &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);
}
} else {
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
}
if (joined_stereo) {
- float tmp[10][MPA_MAX_CHANNELS];
-
- for (k = 0; k < run; k++) {
- tmp[k][0] = samples[k];
- tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
+ for (k = 0; k < run && j + k < 128; k++) {
+ q->sb_samples[0][j + k][sb] =
+ q->tone_level[0][sb][(j + k) / 2] * samples[k];
+ if (q->nb_channels == 2) {
+ if (sign_bits[(j + k) / 8])
+ q->sb_samples[1][j + k][sb] =
+ q->tone_level[1][sb][(j + k) / 2] * -samples[k];
+ else
+ q->sb_samples[1][j + k][sb] =
+ q->tone_level[1][sb][(j + k) / 2] * samples[k];
+ }
}
- for (chs = 0; chs < q->nb_channels; chs++)
- for (k = 0; k < run; k++)
- if ((j + k) < 128)
- q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
} else {
for (k = 0; k < run; k++)
if ((j + k) < 128)
- q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
+ q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
}
j += run;
} // subband loop
}
-
/**
- * 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]
+ * 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.
*
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
- * @param gb bitreader context
- * @param length packet length in bits
+ * @param bc bitreader context
*/
-static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
+static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
+ BitstreamContext *bc)
{
int i, k, run, level, diff;
- if (BITS_LEFT(length,gb) < 16)
+ if (bitstream_bits_left(bc) < 16)
return;
- level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
+ level = qdm2_get_vlc(bc, &vlc_tab_level, 0, 2);
quantized_coeffs[0] = level;
for (i = 0; i < 7; ) {
- if (BITS_LEFT(length,gb) < 16)
+ if (bitstream_bits_left(bc) < 16)
break;
- run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
+ run = qdm2_get_vlc(bc, &vlc_tab_run, 0, 1) + 1;
- if (BITS_LEFT(length,gb) < 16)
+ if (bitstream_bits_left(bc) < 16)
break;
- diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
+ diff = qdm2_get_se_vlc(&vlc_tab_diff, bc, 2);
for (k = 1; k <= run; k++)
quantized_coeffs[i + k] = (level + ((k * diff) / run));
}
}
-
/**
* Related to synthesis filter, process data from packet 10
* Init part of quantized_coeffs via function init_quantized_coeffs_elem0
- * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
+ * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
+ * data from packet 10
*
* @param q context
- * @param gb bitreader context
- * @param length packet length in bits
+ * @param bc bitreader context
*/
-static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
+static void init_tone_level_dequantization(QDM2Context *q, BitstreamContext *bc)
{
int sb, j, k, n, ch;
for (ch = 0; ch < q->nb_channels; ch++) {
- init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
+ init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], bc);
- if (BITS_LEFT(length,gb) < 16) {
+ if (bitstream_bits_left(bc) < 16) {
memset(q->quantized_coeffs[ch][0], 0, 8);
break;
}
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 1)
+ if (bitstream_bits_left(bc) < 1)
break;
- if (get_bits1(gb)) {
+ if (bitstream_read_bit(bc)) {
for (k=0; k < 8; k++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (bitstream_bits_left(bc) < 16)
break;
- q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
+ q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_hi1, 0, 2);
}
} else {
for (k=0; k < 8; k++)
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (bitstream_bits_left(bc) < 16)
break;
- q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
+ q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_hi2, 0, 2);
if (sb > 19)
q->tone_level_idx_hi2[ch][sb] -= 16;
else
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (bitstream_bits_left(bc) < 16)
break;
- q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
+ q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
}
}
* @param q context
* @param node pointer to node with packet
*/
-static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
+static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
{
- GetBitContext gb;
+ BitstreamContext bc;
int i, j, k, n, ch, run, level, diff;
- init_get_bits(&gb, node->packet->data, node->packet->size*8);
+ bitstream_init(&bc, node->packet->data, node->packet->size * 8);
- n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
+ n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
for (i = 1; i < n; i++)
- for (ch=0; ch < q->nb_channels; ch++) {
- level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
+ for (ch = 0; ch < q->nb_channels; ch++) {
+ level = qdm2_get_vlc(&bc, &vlc_tab_level, 0, 2);
q->quantized_coeffs[ch][i][0] = level;
for (j = 0; j < (8 - 1); ) {
- run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
- diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
+ run = qdm2_get_vlc(&bc, &vlc_tab_run, 0, 1) + 1;
+ diff = qdm2_get_se_vlc(&vlc_tab_diff, &bc, 2);
for (k = 1; k <= run; k++)
- q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
+ q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
level += diff;
- j += run;
+ j += run;
}
}
q->quantized_coeffs[ch][0][i] = 0;
}
-
/**
* Process subpacket 10 if not null, else
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bits
*/
-static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
{
- GetBitContext gb;
-
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
+ BitstreamContext bc;
- if (length != 0) {
- init_tone_level_dequantization(q, &gb, length);
+ if (node) {
+ bitstream_init(&bc, node->packet->data, node->packet->size * 8);
+ init_tone_level_dequantization(q, &bc);
fill_tone_level_array(q, 1);
} else {
fill_tone_level_array(q, 0);
}
}
-
/**
* Process subpacket 11
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bit
*/
-static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
{
- GetBitContext gb;
+ BitstreamContext bc;
+ int length = 0;
+
+ if (node) {
+ length = node->packet->size * 8;
+ bitstream_init(&bc, node->packet->data, length);
+ }
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
if (length >= 32) {
- int c = get_bits (&gb, 13);
+ int c = bitstream_read(&bc, 13);
if (c > 3)
- fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
- q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
+ fill_coding_method_array(q->tone_level_idx,
+ q->tone_level_idx_temp, q->coding_method,
+ q->nb_channels, 8 * c,
+ q->superblocktype_2_3, q->cm_table_select);
}
- synthfilt_build_sb_samples(q, &gb, length, 0, 8);
+ synthfilt_build_sb_samples(q, &bc, length, 0, 8);
}
-
/**
* Process subpacket 12
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bits
*/
-static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
{
- GetBitContext gb;
+ BitstreamContext bc;
+ int length = 0;
+
+ if (node) {
+ length = node->packet->size * 8;
+ bitstream_init(&bc, node->packet->data, length);
+ }
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
- synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
+ synthfilt_build_sb_samples(q, &bc, length, 8, QDM2_SB_USED(q->sub_sampling));
}
/*
* @param q context
* @param list list with synthesis filter packets (list D)
*/
-static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
+static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
{
QDM2SubPNode *nodes[4];
nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
- if (nodes[0] != NULL)
+ if (nodes[0])
process_subpacket_9(q, nodes[0]);
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
- if (nodes[1] != NULL)
- process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
+ if (nodes[1])
+ process_subpacket_10(q, nodes[1]);
else
- process_subpacket_10(q, NULL, 0);
+ process_subpacket_10(q, NULL);
nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
- if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
- process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
+ if (nodes[0] && nodes[1] && nodes[2])
+ process_subpacket_11(q, nodes[2]);
else
- process_subpacket_11(q, NULL, 0);
+ process_subpacket_11(q, NULL);
nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
- if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
- process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
+ if (nodes[0] && nodes[1] && nodes[3])
+ process_subpacket_12(q, nodes[3]);
else
- process_subpacket_12(q, NULL, 0);
+ process_subpacket_12(q, NULL);
}
-
/*
* Decode superblock, fill packet lists.
*
* @param q context
*/
-static void qdm2_decode_super_block (QDM2Context *q)
+static void qdm2_decode_super_block(QDM2Context *q)
{
- GetBitContext gb;
+ BitstreamContext bc;
QDM2SubPacket header, *packet;
int i, packet_bytes, sub_packet_size, sub_packets_D;
unsigned int next_index = 0;
memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
q->sub_packets_B = 0;
- sub_packets_D = 0;
+ sub_packets_D = 0;
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
- init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
- qdm2_decode_sub_packet_header(&gb, &header);
+ bitstream_init(&bc, q->compressed_data, q->compressed_size * 8);
+ qdm2_decode_sub_packet_header(&bc, &header);
if (header.type < 2 || header.type >= 8) {
q->has_errors = 1;
- av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
+ av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
return;
}
q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
- packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
+ packet_bytes = (q->compressed_size - bitstream_tell(&bc) / 8);
- init_get_bits(&gb, header.data, header.size*8);
+ bitstream_init(&bc, header.data, header.size * 8);
if (header.type == 2 || header.type == 4 || header.type == 5) {
- int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
+ int csum = 257 * bitstream_read(&bc, 8);
+ csum += 2 * bitstream_read(&bc, 8);
csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
if (csum != 0) {
q->has_errors = 1;
- av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
+ av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
return;
}
}
for (i = 0; packet_bytes > 0; i++) {
int j;
+ if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
+ SAMPLES_NEEDED_2("too many packet bytes");
+ return;
+ }
+
q->sub_packet_list_A[i].next = NULL;
if (i > 0) {
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
/* seek to next block */
- init_get_bits(&gb, header.data, header.size*8);
- skip_bits(&gb, next_index*8);
+ bitstream_init(&bc, header.data, header.size * 8);
+ bitstream_skip(&bc, next_index * 8);
if (next_index >= header.size)
break;
/* decode subpacket */
packet = &q->sub_packets[i];
- qdm2_decode_sub_packet_header(&gb, packet);
- next_index = packet->size + get_bits_count(&gb) / 8;
+ qdm2_decode_sub_packet_header(&bc, packet);
+ next_index = packet->size + bitstream_tell(&bc) / 8;
sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
if (packet->type == 0)
QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
} else if (packet->type == 13) {
for (j = 0; j < 6; j++)
- q->fft_level_exp[j] = get_bits(&gb, 6);
+ q->fft_level_exp[j] = bitstream_read(&bc, 6);
} else if (packet->type == 14) {
for (j = 0; j < 6; j++)
- q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
+ q->fft_level_exp[j] = qdm2_get_vlc(&bc, &fft_level_exp_vlc, 0, 2);
} else if (packet->type == 15) {
SAMPLES_NEEDED_2("packet type 15")
return;
- } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
+ } else if (packet->type >= 16 && packet->type < 48 &&
+ !fft_subpackets[packet->type - 16]) {
/* packets for FFT */
QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
}
} // Packet bytes loop
-/* **************************************************************** */
- if (q->sub_packet_list_D[0].packet != NULL) {
+ if (q->sub_packet_list_D[0].packet) {
process_synthesis_subpackets(q, q->sub_packet_list_D);
q->do_synth_filter = 1;
} else if (q->do_synth_filter) {
- process_subpacket_10(q, NULL, 0);
- process_subpacket_11(q, NULL, 0);
- process_subpacket_12(q, NULL, 0);
+ process_subpacket_10(q, NULL);
+ process_subpacket_11(q, NULL);
+ process_subpacket_12(q, NULL);
}
-/* **************************************************************** */
}
-
-static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
- int offset, int duration, int channel,
- int exp, int phase)
+static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
+ int offset, int duration, int channel,
+ int exp, int phase)
{
if (q->fft_coefs_min_index[duration] < 0)
q->fft_coefs_min_index[duration] = q->fft_coefs_index;
- q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
+ q->fft_coefs[q->fft_coefs_index].sub_packet =
+ ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
q->fft_coefs[q->fft_coefs_index].channel = channel;
- q->fft_coefs[q->fft_coefs_index].offset = offset;
- q->fft_coefs[q->fft_coefs_index].exp = exp;
- q->fft_coefs[q->fft_coefs_index].phase = phase;
+ q->fft_coefs[q->fft_coefs_index].offset = offset;
+ q->fft_coefs[q->fft_coefs_index].exp = exp;
+ q->fft_coefs[q->fft_coefs_index].phase = phase;
q->fft_coefs_index++;
}
-
-static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
+static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
+ BitstreamContext *bc, int b)
{
int channel, stereo, phase, exp;
- int local_int_4, local_int_8, stereo_phase, local_int_10;
+ int local_int_4, local_int_8, stereo_phase, local_int_10;
int local_int_14, stereo_exp, local_int_20, local_int_28;
int n, offset;
- local_int_4 = 0;
+ local_int_4 = 0;
local_int_28 = 0;
local_int_20 = 2;
- local_int_8 = (4 - duration);
+ local_int_8 = (4 - duration);
local_int_10 = 1 << (q->group_order - duration - 1);
- offset = 1;
+ offset = 1;
while (1) {
if (q->superblocktype_2_3) {
- while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
+ while ((n = qdm2_get_vlc(bc, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
offset = 1;
if (n == 0) {
- local_int_4 += local_int_10;
+ local_int_4 += local_int_10;
local_int_28 += (1 << local_int_8);
} else {
- local_int_4 += 8*local_int_10;
+ local_int_4 += 8 * local_int_10;
local_int_28 += (8 << local_int_8);
}
}
offset += (n - 2);
} else {
- offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
+ offset += qdm2_get_vlc(bc, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
while (offset >= (local_int_10 - 1)) {
- offset += (1 - (local_int_10 - 1));
+ offset += (1 - (local_int_10 - 1));
local_int_4 += local_int_10;
local_int_28 += (1 << local_int_8);
}
return;
local_int_14 = (offset >> local_int_8);
+ if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
+ return;
if (q->nb_channels > 1) {
- channel = get_bits1(gb);
- stereo = get_bits1(gb);
+ channel = bitstream_read_bit(bc);
+ stereo = bitstream_read_bit(bc);
} else {
channel = 0;
- stereo = 0;
+ stereo = 0;
}
- exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
+ exp = qdm2_get_vlc(bc, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
- exp = (exp < 0) ? 0 : exp;
+ exp = (exp < 0) ? 0 : exp;
- phase = get_bits(gb, 3);
- stereo_exp = 0;
+ phase = bitstream_read(bc, 3);
+ stereo_exp = 0;
stereo_phase = 0;
if (stereo) {
- stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
- stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
+ stereo_exp = (exp - qdm2_get_vlc(bc, &fft_stereo_exp_vlc, 0, 1));
+ stereo_phase = (phase - qdm2_get_vlc(bc, &fft_stereo_phase_vlc, 0, 1));
if (stereo_phase < 0)
stereo_phase += 8;
}
if (q->frequency_range > (local_int_14 + 1)) {
int sub_packet = (local_int_20 + local_int_28);
- qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
+ qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
+ channel, exp, phase);
if (stereo)
- qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
+ qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
+ 1 - channel,
+ stereo_exp, stereo_phase);
}
-
offset++;
}
}
-
-static void qdm2_decode_fft_packets (QDM2Context *q)
+static void qdm2_decode_fft_packets(QDM2Context *q)
{
int i, j, min, max, value, type, unknown_flag;
- GetBitContext gb;
+ BitstreamContext bc;
- if (q->sub_packet_list_B[0].packet == NULL)
+ if (!q->sub_packet_list_B[0].packet)
return;
/* reset minimum indexes for FFT coefficients */
q->fft_coefs_index = 0;
- for (i=0; i < 5; i++)
+ 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= NULL;
+ QDM2SubPacket *packet = NULL;
/* find subpacket with largest type less than max */
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;
+ min = value;
packet = q->sub_packet_list_B[j].packet;
}
}
if (!packet)
return;
- if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
+ if (i == 0 &&
+ (packet->type < 16 || packet->type >= 48 ||
+ fft_subpackets[packet->type - 16]))
return;
/* decode FFT tones */
- init_get_bits (&gb, packet->data, packet->size*8);
+ bitstream_init(&bc, packet->data, packet->size * 8);
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
unknown_flag = 1;
int duration = q->sub_sampling + 5 - (type & 15);
if (duration >= 0 && duration < 4)
- qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
+ qdm2_fft_decode_tones(q, duration, &bc, unknown_flag);
} else if (type == 31) {
- for (j=0; j < 4; j++)
- qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
+ for (j = 0; j < 4; j++)
+ qdm2_fft_decode_tones(q, j, &bc, unknown_flag);
} else if (type == 46) {
- 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);
+ for (j = 0; j < 6; j++)
+ q->fft_level_exp[j] = bitstream_read(&bc, 6);
+ for (j = 0; j < 4; j++)
+ qdm2_fft_decode_tones(q, j, &bc, unknown_flag);
}
} // Loop on B packets
q->fft_coefs_max_index[j] = q->fft_coefs_index;
}
-
-static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
+static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
{
- float level, f[6];
- int i;
- QDM2Complex c;
- const double iscale = 2.0*M_PI / 512.0;
+ float level, f[6];
+ int i;
+ QDM2Complex c;
+ const double iscale = 2.0 * M_PI / 512.0;
tone->phase += tone->phase_shift;
/* calculate current level (maximum amplitude) of tone */
level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
- c.im = level * sin(tone->phase*iscale);
- c.re = level * cos(tone->phase*iscale);
+ c.im = level * sin(tone->phase * iscale);
+ c.re = level * cos(tone->phase * iscale);
/* generate FFT coefficients for tone */
if (tone->duration >= 3 || tone->cutoff >= 3) {
tone->complex[1].re -= c.re;
} else {
f[1] = -tone->table[4];
- f[0] = tone->table[3] - tone->table[0];
- f[2] = 1.0 - tone->table[2] - tone->table[3];
- f[3] = tone->table[1] + tone->table[4] - 1.0;
- f[4] = tone->table[0] - tone->table[1];
- f[5] = tone->table[2];
+ f[0] = tone->table[3] - tone->table[0];
+ f[2] = 1.0 - tone->table[2] - tone->table[3];
+ f[3] = tone->table[1] + tone->table[4] - 1.0;
+ f[4] = tone->table[0] - tone->table[1];
+ f[5] = tone->table[2];
for (i = 0; i < 2; 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]);
+ 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->complex[i].re += c.re * f[i+2];
- tone->complex[i].im += c.im * f[i+2];
+ tone->complex[i].re += c.re * f[i + 2];
+ tone->complex[i].im += c.im * f[i + 2];
}
}
/* copy the tone if it has not yet died out */
if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
- memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
- q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
+ memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
+ q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
}
}
-
-static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
+static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
{
int i, j, ch;
const double iscale = 0.25 * M_PI;
}
}
-
-static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
+static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
{
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
+ float *out = q->output_buffer + channel;
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]);
+ q->fft.complex[channel][0].im = 0.0f;
+ q->rdft_ctx.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[channel])[i] * gain;
+ for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
+ out[0] += q->fft.complex[channel][i].re * gain;
+ out[q->channels] += q->fft.complex[channel][i].im * gain;
+ out += 2 * q->channels;
+ }
}
-
/**
* @param q context
* @param index subpacket number
*/
-static void qdm2_synthesis_filter (QDM2Context *q, int index)
+static void qdm2_synthesis_filter(QDM2Context *q, int index)
{
- OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
/* copy sb_samples */
for (ch = 0; ch < q->channels; ch++)
for (i = 0; i < 8; i++)
- for (k=sb_used; k < SBLIMIT; k++)
+ for (k = sb_used; k < SBLIMIT; k++)
q->sb_samples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < q->nb_channels; ch++) {
- OUT_INT *samples_ptr = samples + ch;
+ float *samples_ptr = q->samples + ch;
for (i = 0; i < 8; i++) {
- ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
- ff_mpa_synth_window, &dither_state,
- samples_ptr, q->nb_channels,
- q->sb_samples[ch][(8 * index) + i]);
+ ff_mpa_synth_filter_float(&q->mpadsp,
+ q->synth_buf[ch], &(q->synth_buf_offset[ch]),
+ ff_mpa_synth_window_float, &dither_state,
+ samples_ptr, q->nb_channels,
+ q->sb_samples[ch][(8 * index) + i]);
samples_ptr += 32 * q->nb_channels;
}
}
for (ch = 0; ch < q->channels; ch++)
for (i = 0; i < q->frame_size; i++)
- q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
+ q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
}
-
/**
* Init static data (does not depend on specific file)
*
* @param q context
*/
-static av_cold void qdm2_init(QDM2Context *q) {
- static int initialized = 0;
-
- if (initialized != 0)
- return;
- initialized = 1;
-
+static av_cold void qdm2_init_static_data(AVCodec *codec) {
qdm2_init_vlc();
- ff_mpa_synth_init(ff_mpa_synth_window);
+ ff_mpa_synth_init_float(ff_mpa_synth_window_float);
softclip_table_init();
rnd_table_init();
init_noise_samples();
-
- av_log(NULL, AV_LOG_DEBUG, "init done\n");
-}
-
-
-#if 0
-static void dump_context(QDM2Context *q)
-{
- int i;
-#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
- PRINT("compressed_data",q->compressed_data);
- PRINT("compressed_size",q->compressed_size);
- PRINT("frame_size",q->frame_size);
- PRINT("checksum_size",q->checksum_size);
- PRINT("channels",q->channels);
- PRINT("nb_channels",q->nb_channels);
- PRINT("fft_frame_size",q->fft_frame_size);
- PRINT("fft_size",q->fft_size);
- PRINT("sub_sampling",q->sub_sampling);
- PRINT("fft_order",q->fft_order);
- PRINT("group_order",q->group_order);
- PRINT("group_size",q->group_size);
- PRINT("sub_packet",q->sub_packet);
- PRINT("frequency_range",q->frequency_range);
- PRINT("has_errors",q->has_errors);
- PRINT("fft_tone_end",q->fft_tone_end);
- PRINT("fft_tone_start",q->fft_tone_start);
- PRINT("fft_coefs_index",q->fft_coefs_index);
- PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
- PRINT("cm_table_select",q->cm_table_select);
- PRINT("noise_idx",q->noise_idx);
-
- 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);
- PRINT(" phase", t->phase);
- PRINT(" phase_shift", t->phase_shift);
- PRINT(" duration", t->duration);
- PRINT(" samples_im", t->samples_im);
- PRINT(" samples_re", t->samples_re);
- PRINT(" table", t->table);
- }
-
}
-#endif
-
/**
* Init parameters from codec extradata
if (!avctx->extradata || (avctx->extradata_size < 48)) {
av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
- extradata = avctx->extradata;
+ extradata = avctx->extradata;
extradata_size = avctx->extradata_size;
while (extradata_size > 7) {
if (extradata_size < 12) {
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
extradata_size);
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (memcmp(extradata, "frmaQDM", 7)) {
av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (extradata[7] == 'C') {
// s->is_qdmc = 1;
- av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
- return -1;
+ avpriv_report_missing_feature(avctx, "QDMC version 1");
+ return AVERROR_PATCHWELCOME;
}
extradata += 8;
if(size > extradata_size){
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
extradata_size, size);
- return -1;
+ return AVERROR_INVALIDDATA;
}
extradata += 4;
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
extradata += 8;
avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
extradata += 4;
+ if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS)
+ return AVERROR_INVALIDDATA;
+ avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
+ AV_CH_LAYOUT_MONO;
avctx->sample_rate = AV_RB32(extradata);
extradata += 4;
extradata += 4;
s->checksum_size = AV_RB32(extradata);
+ if (s->checksum_size >= 1U << 28) {
+ av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
+ return AVERROR_INVALIDDATA;
+ }
s->fft_order = av_log2(s->fft_size) + 1;
- s->fft_frame_size = 2 * s->fft_size; // complex has two floats
// 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;
s->sub_sampling = s->fft_order - 7;
s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
// 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;
+ avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
+ return AVERROR_PATCHWELCOME;
+ }
+ if (s->fft_size != (1 << (s->fft_order - 1))) {
+ av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
+ return AVERROR_INVALIDDATA;
}
ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
+ ff_mpadsp_init(&s->mpadsp);
- qdm2_init(s);
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- avctx->sample_fmt = SAMPLE_FMT_S16;
-
-// dump_context(s);
return 0;
}
-
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
{
QDM2Context *s = avctx->priv_data;
return 0;
}
-
-static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
+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);
q->compressed_data = in;
q->compressed_size = q->checksum_size;
-// dump_context(q);
-
/* copy old block, clear new block of output samples */
memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
for (ch = 0; ch < q->channels; ch++) {
qdm2_calculate_fft(q, ch, q->sub_packet);
- if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
+ if (!q->has_errors && q->sub_packet_list_C[0].packet) {
SAMPLES_NEEDED_2("has errors, and C list is not empty")
- return;
+ return -1;
}
}
q->sub_packet = (q->sub_packet + 1) % 16;
- /* clip and convert output float[] to 16bit signed samples */
+ /* clip and convert output float[] to 16-bit signed samples */
for (i = 0; i < frame_size; i++) {
int value = (int)q->output_buffer[i];
out[i] = value;
}
-}
+ return 0;
+}
-static int qdm2_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
+ int16_t *out;
+ int i, ret;
if(!buf)
return 0;
if(buf_size < s->checksum_size)
return -1;
- *data_size = s->channels * s->frame_size * sizeof(int16_t);
-
- av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
- buf_size, buf, s->checksum_size, data, *data_size);
-
- qdm2_decode(s, buf, 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");
+ return ret;
+ }
+ out = (int16_t *)frame->data[0];
- // reading only when next superblock found
- if (s->sub_packet == 0) {
- return s->checksum_size;
+ for (i = 0; i < 16; i++) {
+ if ((ret = qdm2_decode(s, buf, out)) < 0)
+ return ret;
+ out += s->channels * s->frame_size;
}
- return 0;
+ *got_frame_ptr = 1;
+
+ return s->checksum_size;
}
-AVCodec qdm2_decoder =
-{
- .name = "qdm2",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_QDM2,
- .priv_data_size = sizeof(QDM2Context),
- .init = qdm2_decode_init,
- .close = qdm2_decode_close,
- .decode = qdm2_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
+AVCodec ff_qdm2_decoder = {
+ .name = "qdm2",
+ .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .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,
};
projects / ffmpeg / blobdiff