Medium frame (divided in m
uint8_t bark_n_coef
number of BSE CB coefficients to read
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_dlog(ac->avr,"%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
uint16_t size
frame size in samples
#define FF_ARRAY_ELEMS(a)
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]
static const uint8_t wtype_to_wsize[]
enum TwinVQFrameType ff_twinvq_wtype_to_ftype_table[]
uint8_t sub_gain_bits[TWINVQ_CHANNELS_MAX *TWINVQ_SUBBLOCKS_MAX]
Short frame (divided in n sub-blocks)
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
uint8_t lpc_idx2[TWINVQ_CHANNELS_MAX][TWINVQ_LSP_SPLIT_MAX]
enum AVSampleFormat sample_fmt
audio sample format
const int16_t * ppc_shape_cb
PPC shape CB.
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]
#define TWINVQ_SUBBLOCKS_MAX
static void imdct_output(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float **out, int offset)
static void interpolate(float *out, float v1, float v2, int size)
#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]
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
static void dequant(TwinVQContext *tctx, const uint8_t *cb_bits, float *out, enum TwinVQFrameType ftype, const int16_t *cb0, const int16_t *cb1, int cb_len)
Inverse quantization.
Long frame (single sub-block + PPC)
static float twinvq_mulawinv(float y, float clip, float mu)
static av_cold void init_bitstream_params(TwinVQContext *tctx)
uint8_t main_coeffs[1024]
#define TWINVQ_MAX_FRAMES_PER_PACKET
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the product of two vectors of floats and store the result in a vector of floats...
#define TWINVQ_PPC_SHAPE_CB_SIZE
static void decode_lsp(TwinVQContext *tctx, int lpc_idx1, uint8_t *lpc_idx2, int lpc_hist_idx, float *lsp, float *hist)
void av_log(void *avcl, int level, const char *fmt,...)
static void eval_lpcenv_2parts(TwinVQContext *tctx, enum TwinVQFrameType ftype, const float *buf, float *lpc, int size, int step)
int(* read_bitstream)(AVCodecContext *avctx, struct TwinVQContext *tctx, const uint8_t *buf, int buf_size)
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)
const int16_t * cb0
main codebooks for spectrum data
static void dec_lpc_spectrum_inv(TwinVQContext *tctx, float *lsp, enum TwinVQFrameType ftype, float *lpc)
int bit_rate
the average bitrate
audio channel layout utility functions
uint8_t sub
Number subblocks in each frame.
uint8_t bits_main_spec[2][4][2]
bits for the main codebook
static void twinvq_memset_float(float *buf, float val, int size)
uint8_t n_lsp
number of lsp coefficients
int ff_twinvq_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static void rearrange_lsp(int order, float *lsp, float min_dist)
Rearrange the LSP coefficients so that they have a minimum distance of min_dist.
if(ac->has_optimized_func)
uint8_t ppc_shape_bit
number of bits of the PPC shape CB coeffs
void(* butterflies_float)(float *restrict v1, float *restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
Libavcodec external API header.
static const int16_t cos_tab[COS_TBL_SIZE]
float bark_hist[3][2][40]
BSE coefficients of last frame.
static void permutate_in_line(int16_t *tab, int num_vect, int num_blocks, int block_size, const uint8_t line_len[2], int length_div, enum TwinVQFrameType ftype)
Interpret the data as if it were a num_blocks x line_len[0] matrix and for each line do a cyclic perm...
static void eval_lpcenv(TwinVQContext *tctx, const float *cos_vals, float *lpc)
Evaluate the LPC amplitude spectrum envelope from the line spectrum pairs.
int sample_rate
samples per second
Periodic Peak Component (part of the long frame)
main external API structure.
enum TwinVQFrameType ftype
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
struct TwinVQFrameMode fmode[3]
frame type-dependant parameters
SINETABLE_CONST float *const ff_sine_windows[14]
static void read_and_decode_spectrum(TwinVQContext *tctx, float *out, enum TwinVQFrameType ftype)
#define TWINVQ_PPC_SHAPE_LEN_MAX
float * prev_frame
non-interleaved previous frame
uint8_t pgain_bit
bits for PPC gain
static float eval_lpc_spectrum(const float *lsp, float cos_val, int order)
Evaluate a single LPC amplitude spectrum envelope coefficient from the line spectrum pairs...
static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float *in, float *prev, int ch)
av_cold void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
Initialize a float DSP context.
const float * lspcodebook
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_dlog(ac->avr,"%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-> out
static av_cold int init_mdct_win(TwinVQContext *tctx)
Init IMDCT and windowing tables.
float lsp_hist[2][20]
LSP coefficients of the last frame.
#define TWINVQ_LSP_COEFS_MAX
common internal api header.
#define FF_ALLOC_OR_GOTO(ctx, p, size, label)
uint8_t ppc_shape_len
size of PPC shape CB
#define TWINVQ_SUB_GAIN_BITS
void ff_sort_nearly_sorted_floats(float *vals, int len)
Sort values in ascending order.
int channels
number of audio channels
static void transpose_perm(int16_t *out, int16_t *in, int num_vect, const uint8_t line_len[2], int length_div)
Interpret the input data as in the following table:
static const struct twinvq_data tab
static void eval_lpcenv_or_interp(TwinVQContext *tctx, enum TwinVQFrameType ftype, float *out, const float *in, int size, int step, int part)
Evaluate the LPC amplitude spectrum envelope from the line spectrum pairs.
static float get_cos(int idx, int part, const float *cos_tab, int size)
float * curr_frame
non-interleaved output
#define FFSWAP(type, a, b)
static void dec_gain(TwinVQContext *tctx, enum TwinVQFrameType ftype, float *out)
#define TWINVQ_CHANNELS_MAX
uint8_t ** extended_data
pointers to the data planes/channels.
uint8_t length[4][2]
main codebook stride
This structure stores compressed data.
static av_cold void construct_perm_table(TwinVQContext *tctx, enum TwinVQFrameType ftype)
#define TWINVQ_SUB_AMP_MAX
int nb_samples
number of audio samples (per channel) described by this frame
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
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)