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85 unsigned int low = 0, high =
size - 1;
88 int index = (low + high) >> 1;
111 float num = 0, den = 0;
137 const float *ortho1,
const float *ortho2,
138 const float *
data,
float *score,
float *gain)
178 vect[lag +
i] =
cb[
i];
193 const float *coefs,
float *
data)
196 float score, gain, best_score,
av_uninit(best_gain);
199 gain = best_score = 0;
203 if (score > best_score) {
242 const float *ortho2,
float *
data,
int *idx,
246 float g, score, best_score;
249 *idx = *gain = best_score = 0;
254 if (score > best_score) {
276 int cba_idx,
int *cb1_idx,
int *cb2_idx)
288 memcpy(cba_vect,
work,
sizeof(cba_vect));
291 data, cb1_idx, &gain);
305 memcpy(cb1_vect,
work,
sizeof(cb1_vect));
311 ortho_cb1 ? cb1_vect :
NULL,
data, cb2_idx, &gain);
325 const int16_t *sblock_data,
326 const int16_t *lpc_coefs,
unsigned int rms,
332 int cba_idx, cb1_idx, cb2_idx, gain;
336 float error, best_error;
340 coefs[
i] = lpc_coefs[
i] * (1/4096.0);
386 best_error = FLT_MAX;
388 for (n = 0; n < 256; n++) {
401 (
data[
i] - sblock_data[
i]);
407 (
data[
i] - sblock_data[
i]);
410 if (
error < best_error) {
427 static const uint8_t
sizes[
LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
428 static const uint8_t bit_sizes[
LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
436 unsigned int refl_rms[
NBLOCKS];
456 energy += (lpc_data[
i] * lpc_data[
i]) >> 4;
462 energy += (lpc_data[
i] * lpc_data[
i]) >> 4;
489 memset(lpc_refl, 0,
sizeof(lpc_refl));
501 refl_rms[1] =
ff_interp(ractx, block_coefs[1], 2,
502 energy <= ractx->old_energy,
504 refl_rms[2] =
ff_interp(ractx, block_coefs[2], 3, 0, energy);
510 block_coefs[
i], refl_rms[
i], &pb);
539 .
p.
name =
"real_144",
551 .p.supported_samplerates = (
const int[]){ 8000, 0 },
static void error(const char *err)
int frame_size
Number of samples per channel in an audio frame.
#define NBLOCKS
number of subblocks within a block
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.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
static double cb(void *priv, double x, double y)
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define u(width, name, range_min, range_max)
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.
#define AV_CH_LAYOUT_MONO
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.
const int16_t ff_energy_tab[32]
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
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.
This structure describes decoded (raw) audio or video data.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
static const uint16_t table[]
av_cold void ff_audiodsp_init(AudioDSPContext *c)
#define FIXED_CB_SIZE
size of fixed codebooks
const int8_t ff_cb2_vects[128][40]
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
#define BUFFERSIZE
the size of the adaptive codebook
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
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,...
int16_t buffer_a[FFALIGN(BLOCKSIZE, 16)]
AVCodec p
The public AVCodec.
const int8_t ff_cb1_vects[128][40]
int initial_padding
Audio only.
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 void create_adapt_vect(float *vect, const int16_t *cb, int lag)
Create a vector from the adaptive codebook at a given lag value.
#define FF_CODEC_ENCODE_CB(func)
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx)
Evaluate the reflection coefficients from the filter coefficients.
const int16_t *const ff_lpc_refl_cb[10]
int ff_irms(AudioDSPContext *adsp, const int16_t *data)
inverse root mean square
const FFCodec ff_ra_144_encoder
int(* init)(AVBSFContext *ctx)
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
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.
int16_t curr_block[NBLOCKS *BLOCKSIZE]
#define CODEC_LONG_NAME(str)
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
const uint16_t ff_cb2_base[128]
static const int sizes[][2]
int64_t bit_rate
the average bitrate
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
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
unsigned int lpc_tables[2][10]
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,...
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
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
An AVChannelLayout holds information about the channel layout of audio data.
static int shift(int a, int b)
const uint16_t ff_cb1_base[128]
void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset)
Copy the last offset values of *source to *target.
unsigned int ff_rms(const int *data)
const int16_t ff_gain_val_tab[256][3]
int nb_samples
number of audio samples (per channel) described by this frame
void ff_int_to_int16(int16_t *out, const int *inp)
#define i(width, name, range_min, range_max)
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
unsigned int old_energy
previous frame energy
static av_cold int ra144_encode_close(AVCodecContext *avctx)
AVSampleFormat
Audio sample formats.
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default value
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
static void orthogonalize(float *v, const float *u)
Orthogonalize a vector to another vector.
#define FFSWAP(type, a, b)
int16_t adapt_cb[146+2]
Adaptive codebook, its size is two units bigger to avoid a buffer overflow.
main external API structure.
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Filter the word “frame” indicates either a video frame or a group of audio samples
static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
#define AV_CHANNEL_LAYOUT_MONO
This structure stores compressed data.
unsigned int lpc_refl_rms[2]
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
#define BLOCKSIZE
subblock size in 16-bit words
@ FF_LPC_TYPE_LEVINSON
Levinson-Durbin recursion.
const uint8_t ff_gain_exp_tab[256]
static av_cold int ra144_encode_init(AVCodecContext *avctx)