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32 #include "config_components.h"
55 #define IMC_BLOCK_SIZE 64
56 #define IMC_FRAME_ID 0x21
118 #define IMC_VLC_BITS 9
119 #define VLC_TABLES_SIZE 9512
125 return 3.5 * atan((freq / 7500.0) * (freq / 7500.0)) + 13.0 * atan(freq * 0.00076);
130 double freqmin[32], freqmid[32], freqmax[32];
131 double scale = sampling_rate / (256.0 * 2.0 * 2.0);
132 double nyquist_freq = sampling_rate * 0.5;
133 double freq, bark, prev_bark = 0,
tf,
tb;
136 for (
i = 0;
i < 32;
i++) {
141 tb = bark - prev_bark;
150 while (
tf < nyquist_freq) {
162 if (
tb <= bark - 0.5)
168 for (
i = 0;
i < 32;
i++) {
170 for (j = 31; j > 0 && freq <= freqmid[j]; j--);
174 for (j = 0; j < 32 && freq >= freqmid[j]; j++);
182 for (
int i = 0,
offset = 0;
i < 4 ;
i++) {
183 for (
int j = 0; j < 4; j++) {
205 "Strange sample rate of %i, file likely corrupt or "
206 "needing a new table derivation method.\n",
239 r1 = sin((
i * 4.0 + 1.0) / 1024.0 *
M_PI);
240 r2 = cos((
i * 4.0 + 1.0) / 1024.0 *
M_PI);
253 for (
i = 0;
i < 30;
i++)
284 float *flcoeffs2,
int *bandWidthT,
285 float *flcoeffs3,
float *flcoeffs5)
290 float snr_limit = 1.e-30;
295 flcoeffs5[
i] = workT2[
i] = 0.0;
297 workT1[
i] = flcoeffs1[
i] * flcoeffs1[
i];
298 flcoeffs3[
i] = 2.0 * flcoeffs2[
i];
301 flcoeffs3[
i] = -30000.0;
303 workT3[
i] = bandWidthT[
i] * workT1[
i] * 0.01;
304 if (workT3[
i] <= snr_limit)
309 for (cnt2 =
i; cnt2 < q->
cyclTab[
i]; cnt2++)
310 flcoeffs5[cnt2] = flcoeffs5[cnt2] + workT3[
i];
311 workT2[cnt2 - 1] = workT2[cnt2 - 1] + workT3[
i];
315 accum = (workT2[
i - 1] + accum) * q->
weights1[
i - 1];
316 flcoeffs5[
i] += accum;
323 for (cnt2 =
i - 1; cnt2 > q->
cyclTab2[
i]; cnt2--)
324 flcoeffs5[cnt2] += workT3[
i];
325 workT2[cnt2+1] += workT3[
i];
331 accum = (workT2[
i+1] + accum) * q->
weights2[
i];
332 flcoeffs5[
i] += accum;
344 const uint8_t *cb_sel;
347 s = stream_format_code >> 1;
354 if (stream_format_code & 4)
361 if (levlCoeffs[
i] == 17)
378 float *flcoeffs1,
float *flcoeffs2)
384 flcoeffs1[0] = 20000.0 /
exp2 (levlCoeffBuf[0] * 0.18945);
385 flcoeffs2[0] =
log2f(flcoeffs1[0]);
397 else if (
level <= 24)
403 tmp2 += 0.83048 *
level;
412 float *old_floor,
float *flcoeffs1,
422 if (levlCoeffBuf[
i] < 16) {
424 flcoeffs2[
i] = (levlCoeffBuf[
i] - 7) * 0.83048 + flcoeffs2[
i];
426 flcoeffs1[
i] = old_floor[
i];
432 float *flcoeffs1,
float *flcoeffs2)
438 flcoeffs1[
pos] = 20000.0 / pow (2, levlCoeffBuf[0] * 0.18945);
441 tmp2 = flcoeffs2[
pos];
447 level = *levlCoeffBuf++;
449 flcoeffs2[
i] = tmp2 - 1.4533435415 *
level;
457 int stream_format_code,
int freebits,
int flag)
460 const float limit = -1.e20;
469 float lowest = 1.e10;
487 highest = highest * 0.25;
506 if (stream_format_code & 0x2) {
513 for (
i = (stream_format_code & 0x2) ? 4 : 0;
i <
BANDS - 1;
i++) {
522 summa = (summa * 0.5 - freebits) / iacc;
526 rres = summer - freebits;
527 if ((rres >= -8) && (rres <= 8))
533 for (j = (stream_format_code & 0x2) ? 4 : 0; j <
BANDS; j++) {
545 if (freebits < summer)
552 summa = (
float)(summer - freebits) / ((
t1 + 1) * iacc) + summa;
555 for (
i = (stream_format_code & 0x2) ? 4 : 0;
i <
BANDS;
i++) {
560 if (freebits > summer) {
569 if (highest <= -1.e20)
576 if (workT[
i] > highest) {
582 if (highest > -1.e20) {
583 workT[found_indx] -= 2.0;
585 workT[found_indx] = -1.e20;
587 for (j =
band_tab[found_indx]; j <
band_tab[found_indx + 1] && (freebits > summer); j++) {
592 }
while (freebits > summer);
594 if (freebits < summer) {
599 if (stream_format_code & 0x2) {
605 while (freebits < summer) {
609 if (workT[
i] < lowest) {
616 workT[low_indx] = lowest + 2.0;
619 workT[low_indx] = 1.e20;
621 for (j =
band_tab[low_indx]; j <
band_tab[low_indx+1] && (freebits < summer); j++) {
702 while (corrected < summer) {
703 if (highest <= -1.e20)
709 if (workT[
i] > highest) {
715 if (highest > -1.e20) {
716 workT[found_indx] -= 2.0;
717 if (++(chctx->
bitsBandT[found_indx]) == 6)
718 workT[found_indx] = -1.e20;
720 for (j =
band_tab[found_indx]; j <
band_tab[found_indx+1] && (corrected < summer); j++) {
764 int stream_format_code)
767 int middle_value, cw_len, max_size;
768 const float *quantizer;
775 if (cw_len <= 0 || chctx->skipFlags[j])
778 max_size = 1 << cw_len;
779 middle_value = max_size >> 1;
806 int i, j, cw_len, cw;
819 "Potential problem on band %i, coefficient %i"
820 ": cw_len=%i\n",
i, j, cw_len);
876 int stream_format_code;
877 int imc_hdr,
i, j,
ret;
886 if (imc_hdr & 0x18) {
893 if (stream_format_code & 0x04)
905 if (stream_format_code & 0x1)
910 if (stream_format_code & 0x1)
913 else if (stream_format_code & 0x4)
929 if (stream_format_code & 0x1) {
956 if (stream_format_code & 0x2) {
963 for (
i = 1;
i < 4;
i++) {
964 if (stream_format_code & 0x1)
977 if (!(stream_format_code & 0x2))
989 if (stream_format_code & 0x1) {
1022 int *got_frame_ptr,
AVPacket *avpkt)
1024 const uint8_t *buf = avpkt->
data;
1025 int buf_size = avpkt->
size;
1034 if (buf_size < IMC_BLOCK_SIZE * avctx->ch_layout.nb_channels) {
1084 #if CONFIG_IMC_DECODER
1101 #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.
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
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)
#define AV_CHANNEL_LAYOUT_MONO
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)
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
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
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.
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
static av_cold void imc_init_static(void)
static void imc_imdct256(IMCContext *q, IMCChannel *chctx, int channels)
static const float imc_weights2[31]
int flags
AV_CODEC_FLAG_*.
static av_always_inline float scale(float x, float s)
static int ff_thread_once(char *control, void(*routine)(void))
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
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)
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]
int ff_init_vlc_from_lengths(VLC *vlc, 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, void *logctx)
Build VLC decoding tables suitable for use with get_vlc2()
static VLC huffman_vlc[4][4]
static av_always_inline int get_vlc2(GetBitContext *s, const VLCElem *table, int bits, int max_depth)
Parse a vlc code.
FFTComplex samples[COEFFS/2]
static const int8_t cyclTab2[32]
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
int bandWidthT[BANDS]
codewords per band
float last_fft_im[COEFFS]
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.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
An AVChannelLayout holds information about the channel layout of audio data.
enum AVSampleFormat sample_fmt
audio sample format
static const float imc_quantizer1[4][8]
void(* fft_permute)(struct FFTContext *s, FFTComplex *z)
Do the permutation needed BEFORE calling fft_calc().
void(* fft_calc)(struct FFTContext *s, FFTComplex *z)
Do a complex FFT with the parameters defined in ff_fft_init().
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 offset
static int imc_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
#define AV_LOG_INFO
Standard information.
void ff_sine_window_init(float *window, int n)
Generate a sine window.
#define DECLARE_ALIGNED(n, t, v)
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.
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
const char * name
Name of the codec implementation.
static av_cold int imc_decode_init(AVCodecContext *avctx)
static double limit(double x)
#define INIT_VLC_STATIC_OVERLONG
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
int skipFlags[COEFFS]
skip coefficient decoding or not
const FFCodec ff_imc_decoder
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(...)
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