59 #define MONO 0x1000001 60 #define STEREO 0x1000002 61 #define JOINT_STEREO 0x1000003 62 #define MC_COOK 0x2000000 // multichannel Cook, not supported 64 #define SUBBAND_SIZE 20 65 #define MAX_SUBPACKETS 5 90 float mono_previous_buffer1[1024];
91 float mono_previous_buffer2[1024];
101 typedef struct cook {
106 void (*scalar_dequant)(
struct cook *q,
int index,
int quant_index,
107 int *subband_coef_index,
int *subband_coef_sign,
110 void (*decouple)(
struct cook *q,
114 float *decode_buffer,
115 float *mlt_buffer1,
float *mlt_buffer2);
117 void (*imlt_window)(
struct cook *q,
float *buffer1,
118 cook_gains *gains_ptr,
float *previous_buffer);
121 int gain_index,
int gain_index_next);
123 void (*saturate_output)(
struct cook *q,
float *
out);
140 VLC envelope_quant_index[13];
145 float gain_table[23];
151 float decode_buffer_1[1024];
152 float decode_buffer_2[1024];
153 float decode_buffer_0[1060];
169 for (i = -63; i < 64; i++) {
180 for (i = 0; i < 23; i++)
191 for (i = 0; i < 13; i++) {
197 for (i = 0; i < 7; i++) {
227 for (j = 0; j < mlt_size; j++)
244 for (i = 0; i < 5; i++)
250 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4) 251 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes))) 275 static const uint32_t
tab[4] = {
282 uint32_t *obuf = (uint32_t *) out;
289 off = (intptr_t) inbuffer & 3;
290 buf = (
const uint32_t *) (inbuffer - off);
293 for (i = 0; i < bytes / 4; i++)
294 obuf[i] = c ^ buf[i];
313 for (i = 0; i < 13; i++)
315 for (i = 0; i < 7; i++)
347 gaininfo[i++] = gain;
360 int *quant_index_table)
364 quant_index_table[0] =
get_bits(&q->
gb, 6) - 6;
380 quant_index_table[i] = quant_index_table[i - 1] + j - 12;
381 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
383 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
384 quant_index_table[i], i);
401 int *category,
int *category_index)
403 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits,
index, v, i, j;
404 int exp_index2[102] = { 0 };
405 int exp_index1[102] = { 0 };
407 int tmp_categorize_array[128 * 2] = { 0 };
420 for (i = 32; i > 0; i = i / 2) {
424 exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
428 if (num_bits >= bits_left - 32)
435 exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
437 exp_index1[i] = exp_idx;
438 exp_index2[i] = exp_idx;
440 tmpbias1 = tmpbias2 = num_bits;
443 if (tmpbias1 + tmpbias2 > 2 * bits_left) {
447 if (exp_index1[i] < 7) {
448 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
457 tmp_categorize_array[tmp_categorize_array1_idx++] =
index;
465 if (exp_index2[i] > 0) {
466 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
475 tmp_categorize_array[--tmp_categorize_array2_idx] =
index;
483 category[i] = exp_index2[i];
486 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
503 int idx = category_index[i];
520 int *subband_coef_index,
int *subband_coef_sign,
527 if (subband_coef_index[i]) {
529 if (subband_coef_sign[i])
549 int *subband_coef_index,
int *subband_coef_sign)
552 int vlc, vd, tmp, result;
556 for (i = 0; i <
vpr_tab[category]; i++) {
562 for (j = vd - 1; j >= 0; j--) {
564 subband_coef_index[vd * i + j] = vlc - tmp * (
kmax_tab[category] + 1);
567 for (j = 0; j < vd; j++) {
568 if (subband_coef_index[i * vd + j]) {
573 subband_coef_sign[i * vd + j] = 0;
576 subband_coef_sign[i * vd + j] = 0;
593 int *quant_index_table,
float *mlt_buffer)
605 index = category[band];
606 if (category[band] < 7) {
607 if (
unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
610 category[band + j] = 7;
614 memset(subband_coef_index, 0,
sizeof(subband_coef_index));
615 memset(subband_coef_sign, 0,
sizeof(subband_coef_sign));
618 subband_coef_index, subband_coef_sign,
630 int category_index[128] = { 0 };
631 int category[128] = { 0 };
632 int quant_index_table[102];
638 categorize(q, p, quant_index_table, category, category_index);
655 int gain_index,
int gain_index_next)
659 fc1 =
pow2tab[gain_index + 63];
661 if (gain_index == gain_index_next) {
665 fc2 = q->
gain_table[11 + (gain_index_next - gain_index)];
682 cook_gains *gains_ptr,
float *previous_buffer)
694 inbuffer[i] = inbuffer[i] * fc * q->
mlt_window[i] -
710 cook_gains *gains_ptr,
float *previous_buffer)
719 q->
imlt_window(q, buffer1, gains_ptr, previous_buffer);
722 for (i = 0; i < 8; i++)
723 if (gains_ptr->
now[i] || gains_ptr->
now[i + 1])
725 gains_ptr->
now[i], gains_ptr->
now[i + 1]);
728 memcpy(previous_buffer, buffer0,
745 int length = end - start + 1;
751 for (i = 0; i < length; i++)
756 for (i = 0; i < length; i++)
775 float *decode_buffer,
776 float *mlt_buffer1,
float *mlt_buffer2)
781 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
782 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
794 float *mlt_buffer_left,
float *mlt_buffer_right)
801 const float *cplscale;
806 memset(mlt_buffer_left, 0, 1024 *
sizeof(*mlt_buffer_left));
807 memset(mlt_buffer_right, 0, 1024 *
sizeof(*mlt_buffer_right));
815 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
816 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
825 idx -= decouple_tab[cpl_tmp];
827 f1 = cplscale[decouple_tab[cpl_tmp] + 1];
829 q->
decouple(q, p, i, f1, f2, decode_buffer,
830 mlt_buffer_left, mlt_buffer_right);
886 cook_gains *gains_ptr,
float *previous_buffer,
889 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
904 const uint8_t *inbuffer,
float **outbuffer)
906 int sub_packet_size = p->
size;
934 outbuffer ? outbuffer[p->
ch_idx + 1] : NULL);
938 outbuffer ? outbuffer[p->
ch_idx + 1] : NULL);
945 int *got_frame_ptr,
AVPacket *avpkt)
949 int buf_size = avpkt->
size;
951 float **samples =
NULL;
956 if (buf_size < avctx->block_align)
977 "frame subpacket size total > avctx->block_align!\n");
988 "subpacket[%i] size %i js %i %i block_align %i\n",
1016 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b); 1046 unsigned int channel_mask = 0;
1047 int samples_per_frame;
1075 samples_per_frame = bytestream2_get_be16(&gb);
1077 bytestream2_get_be32(&gb);
1245 dump_cook_context(q);
static void mlt_compensate_output(COOKContext *q, float *decode_buffer, cook_gains *gains_ptr, float *previous_buffer, float *out)
Final part of subpacket decoding: Apply modulated lapped transform, gain compensation, clip and convert to integer.
static av_cold void init_cplscales_table(COOKContext *q)
static const int cplband[51]
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
function for getting the jointstereo coupling information
This structure describes decoded (raw) audio or video data.
static void categorize(COOKContext *q, COOKSubpacket *p, int *quant_index_table, int *category, int *category_index)
Calculate the category and category_index vector.
static const uint16_t envelope_quant_index_huffcodes[13][24]
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
float decode_buffer_1[1024]
static const int kmax_tab[7]
static const int expbits_tab[8]
static const float *const cplscales[5]
#define DECLARE_ALIGNED(n, t, v)
av_cold void ff_audiodsp_init(AudioDSPContext *c)
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
static av_cold void init_pow2table(void)
#define FF_ARRAY_ELEMS(a)
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
#define AV_CH_LAYOUT_STEREO
VLC envelope_quant_index[13]
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
static const uint8_t *const ccpl_huffbits[5]
static const int vhsize_tab[7]
av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (%s)\, len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic ? ac->func_descr_generic :ac->func_descr)
static const float quant_centroid_tab[7][14]
static void imlt_gain(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer)
The modulated lapped transform, this takes transform coefficients and transforms them into timedomain...
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
static av_cold void init_gain_table(COOKContext *q)
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
void(* vector_clipf)(float *dst, const float *src, float min, float max, int len)
uint8_t * decoded_bytes_buffer
float mono_previous_buffer1[1024]
static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category, int *quant_index_table, float *mlt_buffer)
Fill the mlt_buffer with mlt coefficients.
static void expand_category(COOKContext *q, int *category, int *category_index)
Expand the category vector.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
static void interpolate(float *out, float v1, float v2, int size)
static int get_bits_count(const GetBitContext *s)
bitstream reader API header.
const float * cplscales[5]
static int decode_subpacket(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, float **outbuffer)
Cook subpacket decoding.
static int cook_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, cook_gains *gains_ptr)
First part of subpacket decoding: decode raw stream bytes and read gain info.
#define DECODE_BYTES_PAD1(bytes)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const int vd_tab[7]
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
static const float dither_tab[9]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
static av_always_inline unsigned int bytestream2_get_bytes_left(GetByteContext *g)
static const uint16_t *const ccpl_huffcodes[5]
void av_log(void *avcl, int level, const char *fmt,...)
float mono_previous_buffer2[1024]
const char * name
Name of the codec implementation.
static int decode_envelope(COOKContext *q, COOKSubpacket *p, int *quant_index_table)
Create the quant index table needed for the envelope.
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
uint64_t channel_layout
Audio channel layout.
static void saturate_output_float(COOKContext *q, float *out)
Saturate the output signal and interleave.
#define init_vlc(vlc, nb_bits, nb_codes, bits, bits_wrap, bits_size, codes, codes_wrap, codes_size, flags)
static const int vhvlcsize_tab[7]
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
void(* decouple)(struct cook *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2)
static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, int *subband_coef_index, int *subband_coef_sign)
Unpack the subband_coef_index and subband_coef_sign vectors.
int bit_rate
the average bitrate
static av_cold int init_cook_mlt(COOKContext *q)
audio channel layout utility functions
static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
static av_cold int cook_decode_init(AVCodecContext *avctx)
Cook initialization.
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
static const uint16_t *const cvh_huffcodes[7]
void ff_sine_window_init(float *window, int n)
Generate a sine window.
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
static void interpolate_float(COOKContext *q, float *buffer, int gain_index, int gain_index_next)
the actual requantization of the timedomain samples
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
static av_cold int init_cook_vlc_tables(COOKContext *q)
int sample_rate
samples per second
main external API structure.
static void(WINAPI *cond_broadcast)(pthread_cond_t *cond)
static void close(AVCodecParserContext *s)
float mono_mdct_output[2048]
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
static unsigned int get_bits1(GetBitContext *s)
static int joint_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer_left, float *mlt_buffer_right)
function for decoding joint stereo data
static av_cold int cook_decode_close(AVCodecContext *avctx)
static float pow2tab[127]
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
float decode_buffer_0[1060]
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
COOKSubpacket subpacket[MAX_SUBPACKETS]
float decode_buffer_2[1024]
static float rootpow2tab[127]
static const uint8_t envelope_quant_index_huffbits[13][24]
static const uint8_t *const cvh_huffbits[7]
static int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
void(* scalar_dequant)(struct cook *q, int index, int quant_index, int *subband_coef_index, int *subband_coef_sign, float *mlt_p)
static void scalar_dequant_float(COOKContext *q, int index, int quant_index, int *subband_coef_index, int *subband_coef_sign, float *mlt_p)
The real requantization of the mltcoefs.
common internal api header.
static av_cold int init(AVCodecParserContext *s)
static const int invradix_tab[7]
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
static const struct twinvq_data tab
void(* interpolate)(struct cook *q, float *buffer, int gain_index, int gain_index_next)
static void decode_gain_info(GetBitContext *gb, int *gaininfo)
Fill the gain array for the timedomain quantization.
#define FFSWAP(type, a, b)
void(* saturate_output)(struct cook *q, float *out)
static void imlt_window_float(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer)
Apply transform window, overlap buffers.
static void decouple_float(COOKContext *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2)
static const int vpr_tab[7]
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
void ff_free_vlc(VLC *vlc)
int nb_samples
number of audio samples (per channel) described by this frame
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
void(* imlt_window)(struct cook *q, float *buffer1, cook_gains *gains_ptr, float *previous_buffer)
unsigned int channel_mask
Cook AKA RealAudio G2 compatible decoderdata.