38     512, 12, 
tab.
lsp08, 1, 5, 3, 3, 
tab.
shape08, 8, 28, 20, 6, 40
 
   47     512, 16, 
tab.
lsp11, 1, 6, 4, 3, 
tab.
shape11, 9, 36, 30, 7, 90
 
   56     512, 16, 
tab.
lsp11, 1, 6, 4, 3, 
tab.
shape11, 9, 36, 30, 7, 90
 
   65     1024, 16, 
tab.
lsp16, 1, 6, 4, 3, 
tab.
shape16, 9, 56, 60, 7, 180
 
   74     1024, 16, 
tab.
lsp22_1, 1, 6, 4, 3, 
tab.
shape22_1, 9, 56, 36, 7, 144
 
   83     1024, 16, 
tab.
lsp22_1, 1, 6, 4, 3, 
tab.
shape22_1, 9, 56, 36, 7, 144
 
   92     512, 16, 
tab.
lsp22_2, 1, 6, 4, 4, 
tab.
shape22_2, 9, 56, 36, 7, 72
 
  101     2048, 20, 
tab.
lsp44, 1, 6, 4, 4, 
tab.
shape44, 9, 84, 54, 7, 432
 
  110     2048, 20, 
tab.
lsp44, 1, 6, 4, 4, 
tab.
shape44, 9, 84, 54, 7, 432
 
  139     if (x % 400 || b % 5)
 
  144     size = 
tabs[b / 5].size;
 
  145     rtab = 
tabs[b / 5].tab;
 
  146     return x - rtab[size * 
av_log2(2 * (x - 1) / size) + (x - 1) % size];
 
  155                      float ppc_gain, 
float *speech, 
int len)
 
  159     const float *shape_end = shape + 
len;
 
  163     for (i = 0; i < width / 2; i++)
 
  164         speech[i] += ppc_gain * *shape++;
 
  168         for (j = -width / 2; j < (width + 1) / 2; j++)
 
  169             speech[j + center] += ppc_gain * *shape++;
 
  174     for (j = -width / 2; j < (width + 1) / 2 && shape < shape_end; j++)
 
  175         speech[j + center] += ppc_gain * *shape++;
 
  179                        const float *shape, 
float *speech)
 
  186     int period_range = max_period - min_period;
 
  187     float pgain_step = 25000.0 / ((1 << mtab->
pgain_bit) - 1);
 
  188     float ppc_gain   = 1.0 / 8192 *
 
  195     int period = min_period +
 
  200     if (isampf == 22 && ibps == 32) {
 
  211                          int ch, 
float *
out, 
float gain,
 
  216     float *hist     = tctx->
bark_hist[ftype][ch];
 
  217     float val       = ((
const float []) { 0.4, 0.35, 0.28 })[ftype];
 
  222     for (i = 0; i < fw_cb_len; i++)
 
  223         for (j = 0; j < bark_n_coef; j++, idx++) {
 
  224             float tmp2 = mtab->
fmode[ftype].
bark_cb[fw_cb_len * in[j] + i] *
 
  226             float st   = use_hist ? (1.0 - 
val) * tmp2 + val * hist[idx] + 1.0
 
  243     for (i = 0; i < tctx->
n_div[ftype]; i++) {
 
  278     for (i = 0; i < channels; i++)
 
  279         for (j = 0; j < sub; j++)
 
  281                 bits->
bark1[i][j][k] =
 
  284     for (i = 0; i < channels; i++)
 
  285         for (j = 0; j < sub; j++)
 
  289         for (i = 0; i < channels; i++)
 
  292         for (i = 0; i < channels; i++) {
 
  294             for (j = 0; j < sub; j++)
 
  300     for (i = 0; i < channels; i++) {
 
  310         for (i = 0; i < channels; i++) {
 
  332     if (isampf < 8 || isampf > 44) {
 
  360     if (ibps < 8 || ibps > 48) {
 
  365     switch ((isampf << 8) + ibps) {
 
  395                "This version does not support %d kHz - %d kbit/s/ch mode.\n",
 
  409                "VQF TwinVQ should have only one frame per packet\n");
 
static const TwinVQModeTab mode_44_48
 
const char const char void * val
 
#define AVERROR_INVALIDDATA
Invalid data found when processing input. 
 
uint8_t bark_n_bit
number of bits of the BSE coefs 
 
uint8_t ppc_coeffs[TWINVQ_PPC_SHAPE_LEN_MAX]
 
int bits_main_spec_change[4]
 
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits. 
 
const TwinVQModeTab * mtab
 
int64_t bit_rate
the average bitrate 
 
TwinVQFrameData bits[TWINVQ_MAX_FRAMES_PER_PACKET]
 
int p_coef[TWINVQ_CHANNELS_MAX]
 
static av_cold int init(AVCodecContext *avctx)
 
uint8_t bark_n_coef
number of BSE CB coefficients to read 
 
static const TwinVQModeTab mode_22_20
 
static const uint16_t bark_tab_m08_256[]
 
uint16_t size
frame size in samples 
 
#define AV_CH_LAYOUT_STEREO
 
static void dec_bark_env(TwinVQContext *tctx, const uint8_t *in, int use_hist, int ch, float *out, float gain, enum TwinVQFrameType ftype)
 
uint8_t bark_use_hist[TWINVQ_CHANNELS_MAX][TWINVQ_SUBBLOCKS_MAX]
 
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
 
uint8_t lpc_idx1[TWINVQ_CHANNELS_MAX]
 
enum TwinVQFrameType ff_twinvq_wtype_to_ftype_table[]
 
uint8_t sub_gain_bits[TWINVQ_CHANNELS_MAX *TWINVQ_SUBBLOCKS_MAX]
 
static av_cold int twinvq_decode_init(AVCodecContext *avctx)
 
uint8_t lpc_idx2[TWINVQ_CHANNELS_MAX][TWINVQ_LSP_SPLIT_MAX]
 
static const TwinVQModeTab mode_22_32
 
int g_coef[TWINVQ_CHANNELS_MAX]
 
uint8_t ppc_period_bit
number of the bits for the PPC period value 
 
av_cold int ff_twinvq_decode_close(AVCodecContext *avctx)
 
uint8_t gain_bits[TWINVQ_CHANNELS_MAX]
 
uint8_t * extradata
some codecs need / can use extradata like Huffman tables. 
 
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
 
static void decode_ppc(TwinVQContext *tctx, int period_coef, int g_coef, const float *shape, float *speech)
 
static int get_bits_count(const GetBitContext *s)
 
#define TWINVQ_WINDOW_TYPE_BITS
 
Parameters and tables that are different for every combination of bitrate/sample rate. 
 
uint8_t lpc_hist_idx[TWINVQ_CHANNELS_MAX]
 
static const TwinVQModeTab mode_44_40
 
bitstream reader API header. 
 
static const struct @76 tabs[]
 
static const uint16_t bark_tab_s08_64[]
 
Long frame (single sub-block + PPC) 
 
uint8_t bark_env_size
number of distinct bark scale envelope values 
 
#define ROUNDED_DIV(a, b)
 
static float twinvq_mulawinv(float y, float clip, float mu)
 
static const TwinVQModeTab mode_16_16
 
uint8_t main_coeffs[1024]
 
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered. 
 
static const TwinVQModeTab mode_08_08
 
AVCodec ff_twinvq_decoder
 
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
 
static const uint16_t bark_tab_m22_256[]
 
const char * name
Name of the codec implementation. 
 
int(* read_bitstream)(AVCodecContext *avctx, struct TwinVQContext *tctx, const uint8_t *buf, int buf_size)
 
static void read_cb_data(TwinVQContext *tctx, GetBitContext *gb, uint8_t *dst, enum TwinVQFrameType ftype)
 
static const uint16_t bark_tab_l22_512[]
 
uint64_t channel_layout
Audio channel layout. 
 
void(* decode_ppc)(struct TwinVQContext *tctx, int period_coef, int g_coef, const float *shape, float *speech)
 
void(* dec_bark_env)(struct TwinVQContext *tctx, const uint8_t *in, int use_hist, int ch, float *out, float gain, enum TwinVQFrameType ftype)
 
audio channel layout utility functions 
 
uint8_t sub
Number subblocks in each frame. 
 
uint16_t peak_per2wid
constant for peak period to peak width conversion 
 
static int twinvq_read_bitstream(AVCodecContext *avctx, TwinVQContext *tctx, const uint8_t *buf, int buf_size)
 
uint8_t bits_main_spec[2][4][2]
bits for the main codebook 
 
static const TwinVQModeTab mode_11_08
 
static void twinvq_memset_float(float *buf, float val, int size)
 
int ff_twinvq_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
 
Libavcodec external API header. 
 
AVSampleFormat
Audio sample formats. 
 
float bark_hist[3][2][40]
BSE coefficients of last frame. 
 
int sample_rate
samples per second 
 
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext. 
 
main external API structure. 
 
const uint16_t * bark_tab
 
static const TwinVQModeTab mode_11_10
 
enum TwinVQFrameType ftype
 
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
 
const int16_t * bark_cb
codebook for the bark scale envelope (BSE) 
 
static unsigned int get_bits1(GetBitContext *s)
 
static void skip_bits(GetBitContext *s, int n)
 
struct TwinVQFrameMode fmode[3]
frame type-dependent parameters 
 
uint8_t pgain_bit
bits for PPC gain 
 
static const TwinVQModeTab mode_22_24
 
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
 
common internal api header. 
 
uint8_t ppc_shape_len
size of PPC shape CB 
 
#define TWINVQ_SUB_GAIN_BITS
 
int channels
number of audio channels 
 
static const struct twinvq_data tab
 
static enum AVSampleFormat sample_fmts[]
 
static const uint16_t bark_tab_l08_512[]
 
static void add_peak(int period, int width, const float *shape, float ppc_gain, float *speech, int len)
Sum to data a periodic peak of a given period, width and shape. 
 
#define TWINVQ_CHANNELS_MAX
 
#define AV_CH_LAYOUT_MONO
 
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators. 
 
static int very_broken_op(int a, int b)
Evaluate a * b / 400 rounded to the nearest integer. 
 
uint8_t lsp_split
number of CB entries for the LSP decoding 
 
uint8_t bark1[TWINVQ_CHANNELS_MAX][TWINVQ_SUBBLOCKS_MAX][TWINVQ_BARK_N_COEF_MAX]
 
av_cold int ff_twinvq_decode_init(AVCodecContext *avctx)