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32 #include "config_components.h"
55 #define IMC_BLOCK_SIZE 64
56 #define IMC_FRAME_ID 0x21
112 #define IMC_VLC_BITS 9
113 #define VLC_TABLES_SIZE 9512
119 return 3.5 * atan((freq / 7500.0) * (freq / 7500.0)) + 13.0 * atan(freq * 0.00076);
124 double freqmin[32], freqmid[32], freqmax[32];
125 double scale = sampling_rate / (256.0 * 2.0 * 2.0);
126 double nyquist_freq = sampling_rate * 0.5;
127 double freq, bark, prev_bark = 0, tf, tb;
130 for (
i = 0;
i < 32;
i++) {
135 tb = bark - prev_bark;
144 while (tf < nyquist_freq) {
156 if (tb <= bark - 0.5)
162 for (
i = 0;
i < 32;
i++) {
164 for (j = 31; j > 0 && freq <= freqmid[j]; j--);
168 for (j = 0; j < 32 && freq >= freqmid[j]; j++);
177 for (
int i = 0;
i < 4 ;
i++) {
178 for (
int j = 0; j < 4; j++) {
193 float scale = 1.0f / (16384);
197 "Strange sample rate of %i, file likely corrupt or "
198 "needing a new table derivation method.\n",
226 for (
i = 0;
i < 30;
i++)
256 float *flcoeffs2,
int *bandWidthT,
257 float *flcoeffs3,
float *flcoeffs5)
262 float snr_limit = 1.e-30;
267 flcoeffs5[
i] = workT2[
i] = 0.0;
269 workT1[
i] = flcoeffs1[
i] * flcoeffs1[
i];
270 flcoeffs3[
i] = 2.0 * flcoeffs2[
i];
273 flcoeffs3[
i] = -30000.0;
275 workT3[
i] = bandWidthT[
i] * workT1[
i] * 0.01;
276 if (workT3[
i] <= snr_limit)
281 for (cnt2 =
i; cnt2 < q->
cyclTab[
i]; cnt2++)
282 flcoeffs5[cnt2] = flcoeffs5[cnt2] + workT3[
i];
283 workT2[cnt2 - 1] = workT2[cnt2 - 1] + workT3[
i];
287 accum = (workT2[
i - 1] + accum) * q->
weights1[
i - 1];
288 flcoeffs5[
i] += accum;
295 for (cnt2 =
i - 1; cnt2 > q->
cyclTab2[
i]; cnt2--)
296 flcoeffs5[cnt2] += workT3[
i];
297 workT2[cnt2+1] += workT3[
i];
303 accum = (workT2[
i+1] + accum) * q->
weights2[
i];
304 flcoeffs5[
i] += accum;
315 const uint8_t *cb_sel;
316 int s = stream_format_code >> 1;
321 if (stream_format_code & 4)
328 if (levlCoeffs[
i] == 17)
345 float *flcoeffs1,
float *flcoeffs2)
351 flcoeffs1[0] = 20000.0 /
exp2 (levlCoeffBuf[0] * 0.18945);
352 flcoeffs2[0] =
log2f(flcoeffs1[0]);
364 else if (
level <= 24)
370 tmp2 += 0.83048 *
level;
379 float *old_floor,
float *flcoeffs1,
389 if (levlCoeffBuf[
i] < 16) {
391 flcoeffs2[
i] = (levlCoeffBuf[
i] - 7) * 0.83048 + flcoeffs2[
i];
393 flcoeffs1[
i] = old_floor[
i];
399 float *flcoeffs1,
float *flcoeffs2)
405 flcoeffs1[
pos] = 20000.0 / pow (2, levlCoeffBuf[0] * 0.18945);
408 tmp2 = flcoeffs2[
pos];
414 level = *levlCoeffBuf++;
416 flcoeffs2[
i] = tmp2 - 1.4533435415 *
level;
424 int stream_format_code,
int freebits,
int flag)
427 const float limit = -1.e20;
436 float lowest = 1.e10;
454 highest = highest * 0.25;
473 if (stream_format_code & 0x2) {
480 for (
i = (stream_format_code & 0x2) ? 4 : 0;
i <
BANDS - 1;
i++) {
489 summa = (summa * 0.5 - freebits) / iacc;
493 rres = summer - freebits;
494 if ((rres >= -8) && (rres <= 8))
500 for (j = (stream_format_code & 0x2) ? 4 : 0; j <
BANDS; j++) {
512 if (freebits < summer)
519 summa = (
float)(summer - freebits) / ((t1 + 1) * iacc) + summa;
522 for (
i = (stream_format_code & 0x2) ? 4 : 0;
i <
BANDS;
i++) {
527 if (freebits > summer) {
536 if (highest <= -1.e20)
543 if (workT[
i] > highest) {
549 if (highest > -1.e20) {
550 workT[found_indx] -= 2.0;
552 workT[found_indx] = -1.e20;
554 for (j =
band_tab[found_indx]; j <
band_tab[found_indx + 1] && (freebits > summer); j++) {
559 }
while (freebits > summer);
561 if (freebits < summer) {
566 if (stream_format_code & 0x2) {
572 while (freebits < summer) {
576 if (workT[
i] < lowest) {
583 workT[low_indx] = lowest + 2.0;
586 workT[low_indx] = 1.e20;
588 for (j =
band_tab[low_indx]; j <
band_tab[low_indx+1] && (freebits < summer); j++) {
669 while (corrected < summer) {
670 if (highest <= -1.e20)
676 if (workT[
i] > highest) {
682 if (highest > -1.e20) {
683 workT[found_indx] -= 2.0;
684 if (++(chctx->
bitsBandT[found_indx]) == 6)
685 workT[found_indx] = -1.e20;
687 for (j =
band_tab[found_indx]; j <
band_tab[found_indx+1] && (corrected < summer); j++) {
698 int stream_format_code)
701 int middle_value, cw_len, max_size;
702 const float *quantizer;
709 if (cw_len <= 0 || chctx->skipFlags[j])
712 max_size = 1 << cw_len;
713 middle_value = max_size >> 1;
740 int i, j, cw_len, cw;
753 "Potential problem on band %i, coefficient %i"
754 ": cw_len=%i\n",
i, j, cw_len);
810 int stream_format_code;
811 int imc_hdr,
i, j,
ret;
820 if (imc_hdr & 0x18) {
827 if (stream_format_code & 0x04)
839 if (stream_format_code & 0x1)
844 if (stream_format_code & 0x1)
847 else if (stream_format_code & 0x4)
863 if (stream_format_code & 0x1) {
890 if (stream_format_code & 0x2) {
897 for (
i = 1;
i < 4;
i++) {
898 if (stream_format_code & 0x1)
911 if (!(stream_format_code & 0x2))
923 if (stream_format_code & 0x1) {
953 memcpy(chctx->
prev_win, q->
temp + 128,
sizeof(
float)*128);
959 int *got_frame_ptr,
AVPacket *avpkt)
961 const uint8_t *buf = avpkt->
data;
962 int buf_size = avpkt->
size;
971 if (buf_size < IMC_BLOCK_SIZE * avctx->ch_layout.nb_channels) {
1022 #if CONFIG_IMC_DECODER
1039 #if CONFIG_IAC_DECODER
const FFCodec ff_iac_decoder
@ AV_SAMPLE_FMT_FLTP
float, planar
#define AV_LOG_WARNING
Something somehow does not look correct.
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static av_cold void flush(AVCodecContext *avctx)
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
void(* butterflies_float)(float *restrict v1, float *restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
int CWlengthT[COEFFS]
how many bits in each codeword
static const float imc_exp_tab[32]
int sample_rate
samples per second
static const uint8_t imc_huffman_lens[4][4][18]
int skipFlagCount[BANDS]
skipped coefficients per band
static int get_bits_count(const GetBitContext *s)
static const float imc_quantizer2[2][56]
This structure describes decoded (raw) audio or video data.
static av_cold void iac_generate_tabs(IMCContext *q, int sampling_rate)
int nb_channels
Number of channels in this layout.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static void imc_calculate_coeffs(IMCContext *q, float *flcoeffs1, float *flcoeffs2, int *bandWidthT, float *flcoeffs3, float *flcoeffs5)
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
static void imc_read_level_coeffs_raw(IMCContext *q, int stream_format_code, int *levlCoeffs)
static VLCElem vlc_tables[VLC_TABLES_SIZE]
static const float *const imc_exp_tab2
static void imc_read_level_coeffs(IMCContext *q, int stream_format_code, int *levlCoeffs)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
AVCodec p
The public AVCodec.
int skipFlagRaw[BANDS]
skip flags are stored in raw form or not
static const int8_t cyclTab[32]
AVChannelLayout ch_layout
Audio channel layout.
static av_cold void imc_init_static(void)
static const float imc_weights2[31]
int flags
AV_CODEC_FLAG_*.
static int ff_thread_once(char *control, void(*routine)(void))
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
For static VLCs, the number of bits can often be hardcoded at each get_vlc2() callsite.
@ AV_TX_FLOAT_MDCT
Standard MDCT with a sample data type of float, double or int32_t, respecively.
static av_cold int imc_decode_close(AVCodecContext *avctx)
#define FF_CODEC_DECODE_CB(func)
#define LOCAL_ALIGNED_16(t, v,...)
static const float imc_weights1[31]
static void imc_get_skip_coeff(IMCContext *q, IMCChannel *chctx)
void(* bswap16_buf)(uint16_t *dst, const uint16_t *src, int len)
#define CODEC_LONG_NAME(str)
int bandFlagsBuf[BANDS]
flags for each band
static void imc_decode_level_coefficients2(IMCContext *q, int *levlCoeffBuf, float *old_floor, float *flcoeffs1, float *flcoeffs2)
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static unsigned int get_bits1(GetBitContext *s)
static const uint16_t band_tab[33]
static av_always_inline int get_vlc2(GetBitContext *s, const VLCElem *table, int bits, int max_depth)
Parse a vlc code.
static const int8_t cyclTab2[32]
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
int bandWidthT[BANDS]
codewords per band
static void imc_decode_level_coefficients_raw(IMCContext *q, int *levlCoeffBuf, float *flcoeffs1, float *flcoeffs2)
static const uint8_t imc_huffman_syms[4][4][18]
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
int(* init)(AVBSFContext *ctx)
#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.
#define DECLARE_ALIGNED(n, t, v)
static const VLCElem * huffman_vlc[4][4]
enum AVSampleFormat sample_fmt
audio sample format
static const float imc_quantizer1[4][8]
static int imc_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
void ff_sine_window_init(float *window, int n)
Generate a sine window.
static void imc_refine_bit_allocation(IMCContext *q, IMCChannel *chctx)
#define i(width, name, range_min, range_max)
static int bit_allocation(IMCContext *q, IMCChannel *chctx, int stream_format_code, int freebits, int flag)
Perform bit allocation depending on bits available.
static int inverse_quant_coeff(IMCContext *q, IMCChannel *chctx, int stream_format_code)
int bitsBandT[BANDS]
how many bits per codeword in band
static const float xTab[14]
AVSampleFormat
Audio sample formats.
const char * name
Name of the codec implementation.
static av_cold int imc_decode_init(AVCodecContext *avctx)
static double limit(double x)
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
static const uint8_t imc_huffman_sizes[4]
#define AV_INPUT_BUFFER_PADDING_SIZE
main external API structure.
int skipFlagBits[BANDS]
bits used to code skip flags
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
int skipFlags[COEFFS]
skip coefficient decoding or not
const FFCodec ff_imc_decoder
const av_cold VLCElem * ff_vlc_init_tables_from_lengths(VLCInitState *state, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags)
static int imc_decode_block(AVCodecContext *avctx, IMCContext *q, int ch)
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define avpriv_request_sample(...)
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
#define AV_CHANNEL_LAYOUT_MONO
static void scale(int *out, const int *in, const int w, const int h, const int shift)
#define VLC_INIT_STATE(_table)
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
static const uint8_t imc_cb_select[4][32]
static void imc_decode_level_coefficients(IMCContext *q, int *levlCoeffBuf, float *flcoeffs1, float *flcoeffs2)
static double freq2bark(double freq)
float mdct_sine_window[COEFFS]
MDCT tables.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static void imc_get_coeffs(AVCodecContext *avctx, IMCContext *q, IMCChannel *chctx)
static void imc_adjust_bit_allocation(IMCContext *q, IMCChannel *chctx, int summer)
Increase highest' band coefficient sizes as some bits won't be used.
int codewords[COEFFS]
raw codewords read from bitstream
int sumLenArr[BANDS]
bits for all coeffs in band