43 #if FF_API_OLD_ENCODE_AUDIO
74 #if FF_API_OLD_ENCODE_AUDIO
99 static int quantize(
int value,
const int16_t *table,
unsigned int size)
101 unsigned int low = 0, high = size - 1;
104 int index = (low + high) >> 1;
105 int error = table[
index] - value;
108 return table[high] + error > value ? low : high;
127 float num = 0, den = 0;
153 const float *ortho1,
const float *ortho2,
154 const float *
data,
float *score,
float *gain)
166 g += work[i] * work[i];
167 c += data[i] * work[i];
194 vect[lag + i] = cb[i];
209 const float *coefs,
float *
data)
212 float score, gain, best_score, best_gain;
215 gain = best_score = 0;
219 if (score > best_score) {
235 data[i] -= best_gain * work[i];
236 return best_vect - BLOCKSIZE / 2 + 1;
257 const int8_t cb[][
BLOCKSIZE],
const float *ortho1,
258 const float *ortho2,
float *
data,
int *idx,
262 float g, score, best_score;
265 *idx = *gain = best_score = 0;
270 if (score > best_score) {
292 int cba_idx,
int *cb1_idx,
int *cb2_idx)
304 memcpy(cba_vect, work,
sizeof(cba_vect));
307 data, cb1_idx, &gain);
320 data[i] -= gain * work[i];
321 memcpy(cb1_vect, work,
sizeof(cb1_vect));
327 ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
341 const int16_t *sblock_data,
342 const int16_t *lpc_coefs,
unsigned int rms,
349 int cba_idx, cb1_idx, cb2_idx, gain;
352 float error, best_error;
356 coefs[i] = lpc_coefs[i] * (1/4096.0);
366 zero[i] = work[LPC_ORDER + i];
367 data[i] = sblock_data[i] - zero[i];
375 memset(work, 0, LPC_ORDER *
sizeof(*work));
384 memcpy(cba, work + LPC_ORDER,
sizeof(cba));
387 m[0] = (
ff_irms(cba_vect) * rms) >> 12;
389 fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
396 memcpy(cb1, work + LPC_ORDER,
sizeof(cb1));
400 memcpy(cb2, work + LPC_ORDER,
sizeof(cb2));
402 best_error = FLT_MAX;
404 for (n = 0; n < 256; n++) {
414 data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
416 error += (data[i] - sblock_data[i]) *
417 (data[i] - sblock_data[i]);
421 data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
422 error += (data[i] - sblock_data[i]) *
423 (data[i] - sblock_data[i]);
426 if (error < best_error) {
441 const AVFrame *frame,
int *got_packet_ptr)
452 unsigned int refl_rms[
NBLOCKS];
453 const int16_t *
samples = frame ? (
const int16_t *)frame->
data[0] :
NULL;
474 energy += (lpc_data[i] * lpc_data[i]) >> 4;
479 lpc_data[i] = samples[j] >> 2;
480 energy += (lpc_data[i] * lpc_data[i]) >> 4;
484 memset(&lpc_data[i], 0, (
NBLOCKS * BLOCKSIZE - i) *
sizeof(*lpc_data));
492 block_coefs[
NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<
493 (12 - shift[LPC_ORDER - 1]));
507 memset(lpc_refl, 0,
sizeof(lpc_refl));
519 refl_rms[1] =
ff_interp(ractx, block_coefs[1], 2,
520 energy <= ractx->old_energy,
522 refl_rms[2] =
ff_interp(ractx, block_coefs[2], 3, 0, energy);
528 block_coefs[i], refl_rms[i], &pb);
545 (NBLOCKS * BLOCKSIZE - i) *
sizeof(*ractx->
curr_block));
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.
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]
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...
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 max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
AVFrame * coded_frame
the picture in the bitstream
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)
static void orthogonalize(float *v, const float *u)
Orthogonalize a vector to another vector.
uint16_t adapt_cb[146+2]
Adaptive codebook, its size is two units bigger to avoid a buffer overflow.
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 othogonalizing it to up to 2 other vectors.
AVCodec ff_ra_144_encoder
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
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 FFSWAP(type, a, b)
unsigned int lpc_refl_rms[2]
unsigned int ff_rms(const int *data)
#define FIXED_CB_SIZE
size of fixed codebooks
const uint16_t ff_cb2_base[128]
#define FRAMESIZE
size of encoded frame
void ff_subblock_synthesis(RA144Context *ractx, const uint16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
static int init(AVCodecParserContext *s)
#define CODEC_CAP_SMALL_LAST_FRAME
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. ...
void av_log(void *avcl, int level, const char *fmt,...)
const char * name
Name of the codec implementation.
static void put_bits(PutBitContext *s, int n, unsigned int value)
Write up to 31 bits into a bitstream.
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.
AVFrame * avcodec_alloc_frame(void)
Allocate an AVFrame and set its fields to default values.
int bit_rate
the average bitrate
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset)
Copy the last offset values of *source to *target.
int ff_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
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.
main external API structure.
static void close(AVCodecParserContext *s)
int ff_irms(const int16_t *data)
inverse root mean square
static av_cold int ra144_encode_init(AVCodecContext *avctx)
Levinson-Durbin recursion.
struct RA144Context RA144Context
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)
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 othogonalizing them to up to 2 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]
AVSampleFormat
Audio Sample Formats.
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.
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int *duration)
Remove frame(s) from the queue.
int channels
number of audio channels
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
const uint16_t ff_cb1_base[128]
int16_t curr_block[NBLOCKS *BLOCKSIZE]
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.
if(!(ptr_align%ac->ptr_align)&&samples_align >=aligned_len)