88 unsigned int low = 0, high = size - 1;
91 int index = (low + high) >> 1;
95 return table[high] + error > value ? low : high;
114 float num = 0, den = 0;
140 const float *ortho1,
const float *ortho2,
141 const float *
data,
float *score,
float *gain)
153 g += work[
i] * work[
i];
154 c += data[
i] * work[
i];
181 vect[lag + i] = cb[i];
196 const float *coefs,
float *
data)
199 float score, gain, best_score,
av_uninit(best_gain);
202 gain = best_score = 0;
206 if (score > best_score) {
222 data[i] -= best_gain * work[i];
223 return best_vect - BLOCKSIZE / 2 + 1;
245 const float *ortho2,
float *
data,
int *idx,
249 float g, score, best_score;
252 *idx = *gain = best_score = 0;
257 if (score > best_score) {
279 int cba_idx,
int *cb1_idx,
int *cb2_idx)
291 memcpy(cba_vect, work,
sizeof(cba_vect));
294 data, cb1_idx, &gain);
307 data[i] -= gain * work[i];
308 memcpy(cb1_vect, work,
sizeof(cb1_vect));
314 ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
328 const int16_t *sblock_data,
329 const int16_t *lpc_coefs,
unsigned int rms,
335 int cba_idx, cb1_idx, cb2_idx, gain;
339 float error, best_error;
343 coefs[
i] = lpc_coefs[
i] * (1/4096.0);
353 zero[
i] =
work[LPC_ORDER +
i];
354 data[
i] = sblock_data[
i] - zero[
i];
362 memset(
work, 0, LPC_ORDER *
sizeof(*
work));
371 memcpy(cba,
work + LPC_ORDER,
sizeof(cba));
383 memcpy(cb1,
work + LPC_ORDER,
sizeof(cb1));
387 memcpy(cb2,
work + LPC_ORDER,
sizeof(cb2));
389 best_error = FLT_MAX;
391 for (n = 0; n < 256; n++) {
401 data[
i] = zero[
i] + g[0] * cba[
i] + g[1] * cb1[
i] +
403 error += (data[
i] - sblock_data[
i]) *
404 (data[i] - sblock_data[i]);
408 data[
i] = zero[
i] + g[1] * cb1[
i] + g[2] * cb2[
i];
409 error += (data[
i] - sblock_data[
i]) *
410 (data[i] - sblock_data[i]);
413 if (error < best_error) {
439 unsigned int refl_rms[
NBLOCKS];
440 const int16_t *
samples = frame ? (
const int16_t *)frame->
data[0] :
NULL;
459 energy += (lpc_data[
i] * lpc_data[
i]) >> 4;
464 lpc_data[
i] = samples[j] >> 2;
465 energy += (lpc_data[
i] * lpc_data[
i]) >> 4;
469 memset(&lpc_data[i], 0, (
NBLOCKS * BLOCKSIZE - i) *
sizeof(*lpc_data));
477 block_coefs[
NBLOCKS - 1][i] = -lpc_coefs[LPC_ORDER - 1][i]
478 * (1 << (12 - shift[LPC_ORDER - 1]));
492 memset(lpc_refl, 0,
sizeof(lpc_refl));
504 refl_rms[1] =
ff_interp(ractx, block_coefs[1], 2,
505 energy <= ractx->old_energy,
507 refl_rms[2] =
ff_interp(ractx, block_coefs[2], 3, 0, energy);
513 block_coefs[i], refl_rms[i], &pb);
530 (NBLOCKS * BLOCKSIZE - i) *
sizeof(*ractx->
curr_block));
554 .supported_samplerates = (
const int[]){ 8000, 0 },
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
unsigned int lpc_tables[2][10]
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
static int shift(int a, int b)
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
This structure describes decoded (raw) audio or video data.
const int16_t *const ff_lpc_refl_cb[10]
ptrdiff_t const GLvoid * data
must be printed separately If there s no standard function for printing the type you the WRITE_1D_FUNC_ARGV macro is a very quick way to create one See libavcodec dv_tablegen c for an example The h file This file should the initialization functions should not do and instead of the variable declarations the generated *_tables h file should be included Since that will be generated in the build the path must be i e not Makefile changes To make the automatic table creation work
static int adaptive_cb_search(const int16_t *adapt_cb, float *work, const float *coefs, float *data)
Search the adaptive codebook for the best entry and gain and remove its contribution from input data...
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
int64_t bit_rate
the average bitrate
static av_cold int init(AVCodecContext *avctx)
static int quantize(int value, const int16_t *table, unsigned int size)
Quantize a value by searching a sorted table for the element with the nearest value.
static av_cold int ra144_encode_close(AVCodecContext *avctx)
int16_t adapt_cb[146+2]
Adaptive codebook, its size is two units bigger to avoid a buffer overflow.
static void orthogonalize(float *v, const float *u)
Orthogonalize a vector to another vector.
static void error(const char *err)
static void get_match_score(float *work, const float *coefs, float *vect, const float *ortho1, const float *ortho2, const float *data, float *score, float *gain)
Calculate match score and gain of an LPC-filtered vector with respect to input data, possibly orthogonalizing it to up to two other vectors.
AVCodec ff_ra_144_encoder
static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
#define NBLOCKS
number of subblocks within a block
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
static double cb(void *priv, double x, double y)
av_cold void ff_audiodsp_init(AudioDSPContext *c)
unsigned int lpc_refl_rms[2]
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
unsigned int ff_rms(const int *data)
GLsizei GLboolean const GLfloat * value
#define u(width, name, range_min, range_max)
#define FIXED_CB_SIZE
size of fixed codebooks
const uint16_t ff_cb2_base[128]
static const uint16_t table[]
void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
unsigned int * lpc_coef[2]
LPC coefficients: lpc_coef[0] is the coefficients of the current frame and lpc_coef[1] of the previou...
static const int sizes[][2]
const int8_t ff_cb1_vects[128][40]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int initial_padding
Audio only.
const char * name
Name of the codec implementation.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
static void fixed_cb_search(float *work, const float *coefs, float *data, int cba_idx, int *cb1_idx, int *cb2_idx)
Search the two fixed codebooks for the best entry and gain.
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset)
Copy the last offset values of *source to *target.
void ff_int_to_int16(int16_t *out, const int *inp)
const int16_t ff_gain_val_tab[256][3]
int frame_size
Number of samples per channel in an audio frame.
#define BLOCKSIZE
subblock size in 16-bit words
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
main external API structure.
static av_cold int ra144_encode_init(AVCodecContext *avctx)
Levinson-Durbin recursion.
int16_t buffer_a[FFALIGN(BLOCKSIZE, 16)]
int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx)
Evaluate the reflection coefficients from the filter coefficients.
#define BUFFERSIZE
the size of the adaptive codebook
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
int ff_lpc_calc_coefs(LPCContext *s, const int32_t *samples, int blocksize, int min_order, int max_order, int precision, int32_t coefs[][MAX_LPC_ORDER], int *shift, enum FFLPCType lpc_type, int lpc_passes, int omethod, int min_shift, int max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static void find_best_vect(float *work, const float *coefs, const int8_t cb[][BLOCKSIZE], const float *ortho1, const float *ortho2, float *data, int *idx, float *gain)
Find the best vector of a fixed codebook by applying an LPC filter to codebook entries, possibly orthogonalizing them to up to two other vectors and matching the results with input data.
const int8_t ff_cb2_vects[128][40]
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
unsigned int old_energy
previous frame energy
const int16_t ff_energy_tab[32]
const uint8_t ff_gain_exp_tab[256]
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
int channels
number of audio channels
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
Filter the word “frame” indicates either a video frame or a group of audio samples
const uint16_t ff_cb1_base[128]
int16_t curr_block[NBLOCKS *BLOCKSIZE]
#define FFSWAP(type, a, b)
int ff_irms(AudioDSPContext *adsp, const int16_t *data)
inverse root mean square
#define AV_CH_LAYOUT_MONO
static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
Create a vector from the adaptive codebook at a given lag value.
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
static void ra144_encode_subblock(RA144Context *ractx, const int16_t *sblock_data, const int16_t *lpc_coefs, unsigned int rms, PutBitContext *pb)
Encode a subblock of the current frame.