43 #define EXPMAX ((19 + EXPVLCBITS - 1) / EXPVLCBITS) 45 #define HGAINVLCBITS 9 46 #define HGAINMAX ((13 + HGAINVLCBITS - 1) / HGAINVLCBITS) 52 int prec,
const float *
tab,
int n)
57 for (i = 0; i < n; i++) {
86 flags2 =
AV_RL16(extradata + 2);
88 flags2 =
AV_RL16(extradata + 4);
138 t.v = ((
u.v <<
LSP_POW_BITS) & ((1 << 23) - 1)) | (127 << 23);
149 wdel = M_PI / frame_len;
150 for (i = 0; i < frame_len; i++)
154 for (i = 0; i < 256; i++) {
180 float p, q, w, v, val_max;
183 for (i = 0; i < n; i++) {
199 *val_max_ptr = val_max;
211 if (i == 0 || i >= 8)
224 1.7782794100389e-04, 2.0535250264571e-04,
225 2.3713737056617e-04, 2.7384196342644e-04,
226 3.1622776601684e-04, 3.6517412725484e-04,
227 4.2169650342858e-04, 4.8696752516586e-04,
228 5.6234132519035e-04, 6.4938163157621e-04,
229 7.4989420933246e-04, 8.6596432336006e-04,
230 1.0000000000000e-03, 1.1547819846895e-03,
231 1.3335214321633e-03, 1.5399265260595e-03,
232 1.7782794100389e-03, 2.0535250264571e-03,
233 2.3713737056617e-03, 2.7384196342644e-03,
234 3.1622776601684e-03, 3.6517412725484e-03,
235 4.2169650342858e-03, 4.8696752516586e-03,
236 5.6234132519035e-03, 6.4938163157621e-03,
237 7.4989420933246e-03, 8.6596432336006e-03,
238 1.0000000000000e-02, 1.1547819846895e-02,
239 1.3335214321633e-02, 1.5399265260595e-02,
240 1.7782794100389e-02, 2.0535250264571e-02,
241 2.3713737056617e-02, 2.7384196342644e-02,
242 3.1622776601684e-02, 3.6517412725484e-02,
243 4.2169650342858e-02, 4.8696752516586e-02,
244 5.6234132519035e-02, 6.4938163157621e-02,
245 7.4989420933246e-02, 8.6596432336007e-02,
246 1.0000000000000e-01, 1.1547819846895e-01,
247 1.3335214321633e-01, 1.5399265260595e-01,
248 1.7782794100389e-01, 2.0535250264571e-01,
249 2.3713737056617e-01, 2.7384196342644e-01,
250 3.1622776601684e-01, 3.6517412725484e-01,
251 4.2169650342858e-01, 4.8696752516586e-01,
252 5.6234132519035e-01, 6.4938163157621e-01,
253 7.4989420933246e-01, 8.6596432336007e-01,
254 1.0000000000000e+00, 1.1547819846895e+00,
255 1.3335214321633e+00, 1.5399265260595e+00,
256 1.7782794100389e+00, 2.0535250264571e+00,
257 2.3713737056617e+00, 2.7384196342644e+00,
258 3.1622776601684e+00, 3.6517412725484e+00,
259 4.2169650342858e+00, 4.8696752516586e+00,
260 5.6234132519035e+00, 6.4938163157621e+00,
261 7.4989420933246e+00, 8.6596432336007e+00,
262 1.0000000000000e+01, 1.1547819846895e+01,
263 1.3335214321633e+01, 1.5399265260595e+01,
264 1.7782794100389e+01, 2.0535250264571e+01,
265 2.3713737056617e+01, 2.7384196342644e+01,
266 3.1622776601684e+01, 3.6517412725484e+01,
267 4.2169650342858e+01, 4.8696752516586e+01,
268 5.6234132519035e+01, 6.4938163157621e+01,
269 7.4989420933246e+01, 8.6596432336007e+01,
270 1.0000000000000e+02, 1.1547819846895e+02,
271 1.3335214321633e+02, 1.5399265260595e+02,
272 1.7782794100389e+02, 2.0535250264571e+02,
273 2.3713737056617e+02, 2.7384196342644e+02,
274 3.1622776601684e+02, 3.6517412725484e+02,
275 4.2169650342858e+02, 4.8696752516586e+02,
276 5.6234132519035e+02, 6.4938163157621e+02,
277 7.4989420933246e+02, 8.6596432336007e+02,
278 1.0000000000000e+03, 1.1547819846895e+03,
279 1.3335214321633e+03, 1.5399265260595e+03,
280 1.7782794100389e+03, 2.0535250264571e+03,
281 2.3713737056617e+03, 2.7384196342644e+03,
282 3.1622776601684e+03, 3.6517412725484e+03,
283 4.2169650342858e+03, 4.8696752516586e+03,
284 5.6234132519035e+03, 6.4938163157621e+03,
285 7.4989420933246e+03, 8.6596432336007e+03,
286 1.0000000000000e+04, 1.1547819846895e+04,
287 1.3335214321633e+04, 1.5399265260595e+04,
288 1.7782794100389e+04, 2.0535250264571e+04,
289 2.3713737056617e+04, 2.7384196342644e+04,
290 3.1622776601684e+04, 3.6517412725484e+04,
291 4.2169650342858e+04, 4.8696752516586e+04,
292 5.6234132519035e+04, 6.4938163157621e+04,
293 7.4989420933246e+04, 8.6596432336007e+04,
294 1.0000000000000e+05, 1.1547819846895e+05,
295 1.3335214321633e+05, 1.5399265260595e+05,
296 1.7782794100389e+05, 2.0535250264571e+05,
297 2.3713737056617e+05, 2.7384196342644e+05,
298 3.1622776601684e+05, 3.6517412725484e+05,
299 4.2169650342858e+05, 4.8696752516586e+05,
300 5.6234132519035e+05, 6.4938163157621e+05,
301 7.4989420933246e+05, 8.6596432336007e+05,
309 int last_exp, n, code;
312 uint32_t *q, *q_end, iv;
313 const float *ptab = pow_tab + 60;
314 const uint32_t *iptab = (
const uint32_t *) ptab;
323 iv = iptab[last_exp];
331 }
while ((n -= 4) > 0);
342 last_exp += code - 60;
349 iv = iptab[last_exp];
358 }
while ((n -= 4) > 0);
373 int block_len, bsize, n;
390 memcpy(out + n + block_len, in + n + block_len, n *
sizeof(
float));
407 memcpy(out, in, n *
sizeof(
float));
412 memset(out + n + block_len, 0, n *
sizeof(
float));
422 int n, v,
a, ch, bsize;
423 int coef_nb_bits, total_gain;
442 "prev_block_len_bits %d out of range\n",
450 "block_len_bits %d out of range\n",
463 "next_block_len_bits %d out of range\n",
521 for (i = 0; i < n; i++) {
535 val = (int) 0x80000000;
536 for (i = 0; i < n; i++) {
538 if (val == (
int) 0x80000000) {
545 "hgain vlc invalid\n");
584 0, ptr, 0, nb_coefs[ch],
594 mdct_norm = 1.0 / (float) n4;
596 mdct_norm *= sqrt(n4);
603 float *coefs, *exponents,
mult, mult1,
noise;
604 int i, j, n, n1, last_high_band, esize;
610 mult = pow(10, total_gain * 0.05) / s->
max_exponent[ch];
612 coefs = s->
coefs[ch];
618 exponents[i << bsize >> esize] * mult1;
629 for (j = 0; j < n1; j++) {
635 for (i = 0; i < n; i++) {
636 v = exponents[i << bsize >> esize];
639 exp_power[j] = e2 / n;
641 ff_tlog(s->
avctx,
"%d: power=%f (%d)\n", j, exp_power[j], n);
643 exponents += n << bsize >> esize;
648 for (j = -1; j < n1; j++) {
656 mult1 = sqrt(exp_power[j] / exp_power[last_high_band]);
661 for (i = 0; i < n; i++) {
664 *coefs++ = noise * exponents[i << bsize >> esize] * mult1;
666 exponents += n << bsize >> esize;
669 for (i = 0; i < n; i++) {
672 *coefs++ = ((*coefs1++) + noise) *
673 exponents[i << bsize >> esize] *
mult;
675 exponents += n << bsize >> esize;
681 mult1 = mult * exponents[((-1 << bsize)) >> esize];
682 for (i = 0; i < n; i++) {
691 for (i = 0; i < n; i++)
692 *coefs++ = coefs1[i] * exponents[i << bsize >> esize] * mult;
694 for (i = 0; i < n; i++)
772 memcpy(samples[ch] + samples_offset, s->
frame_out[ch],
779 dump_floats(s,
"samples", 6, samples[ch] + samples_offset,
788 int *got_frame_ptr,
AVPacket *avpkt)
792 int buf_size = avpkt->
size;
794 int nb_frames, bit_offset, i, pos,
len, ret;
799 ff_tlog(avctx,
"***decode_superframe:\n");
805 if (buf_size < avctx->block_align) {
807 "Input packet size too small (%d < %d)\n",
835 "Invalid last frame bit offset %d > buf size %d (%d)\n",
879 for (i = 0; i < nb_frames; i++) {
890 len = buf_size - pos;
904 ff_dlog(s->
avctx,
"%d %d %d %d outbytes:%td eaten:%d\n",
906 (int8_t *) samples - (int8_t *) data, avctx->
block_align);
const struct AVCodec * codec
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static void wma_lsp_to_curve(WMACodecContext *s, float *out, float *val_max_ptr, int n, float *lsp)
NOTE: We use the same code as Vorbis here.
This structure describes decoded (raw) audio or video data.
static int noise(AVBSFContext *ctx, AVPacket *out)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
int next_block_len_bits
log2 of next block length
static const float pow_tab[]
pow(10, i / 16.0) for i in -60..95
av_log(ac->avr, AV_LOG_TRACE, "%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)
int ff_wma_run_level_decode(AVCodecContext *avctx, GetBitContext *gb, VLC *vlc, const float *level_table, const uint16_t *run_table, int version, WMACoef *ptr, int offset, int num_coefs, int block_len, int frame_len_bits, int coef_nb_bits)
Decode run level compressed coefficients.
int block_len
block length in samples
#define FF_ARRAY_ELEMS(a)
float exponents[MAX_CHANNELS][BLOCK_MAX_SIZE]
#define init_vlc(vlc, nb_bits, nb_codes, bits, bits_wrap, bits_size, codes, codes_wrap, codes_size, flags)
static void wma_window(WMACodecContext *s, float *out)
Apply MDCT window and add into output.
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
float lsp_pow_m_table2[(1<< LSP_POW_BITS)]
Macro definitions for various function/variable attributes.
static int wma_decode_block(WMACodecContext *s)
float lsp_cos_table[BLOCK_MAX_SIZE]
int high_band_start[BLOCK_NB_SIZES]
index of first coef in high band
enum AVSampleFormat sample_fmt
audio sample format
float WMACoef
type for decoded coefficients, int16_t would be enough for wma 1/2
const uint8_t ff_aac_scalefactor_bits[121]
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
int block_pos
current position in frame
static int decode_exp_vlc(WMACodecContext *s, int ch)
decode exponents coded with VLC codes
static int get_bits_count(const GetBitContext *s)
float lsp_pow_m_table1[(1<< LSP_POW_BITS)]
int nb_block_sizes
number of block sizes
int ff_wma_total_gain_to_bits(int total_gain)
static int get_bits_left(GetBitContext *gb)
static float pow_m1_4(WMACodecContext *s, float x)
compute x^-0.25 with an exponent and mantissa table.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
uint16_t exponent_bands[BLOCK_NB_SIZES][25]
uint8_t channel_coded[MAX_CHANNELS]
true if channel is coded
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
uint8_t last_superframe[MAX_CODED_SUPERFRAME_SIZE+AV_INPUT_BUFFER_PADDING_SIZE]
const char * name
Name of the codec implementation.
static int wma_decode_superframe(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
FFTSample output[BLOCK_MAX_SIZE *2]
const uint8_t ff_wma_hgain_huffbits[37]
static av_cold int wma_decode_init(AVCodecContext *avctx)
int exponent_high_bands[BLOCK_NB_SIZES][HIGH_BAND_MAX_SIZE]
int ff_wma_end(AVCodecContext *avctx)
int high_band_values[MAX_CHANNELS][HIGH_BAND_MAX_SIZE]
float * windows[BLOCK_NB_SIZES]
#define MAX_CODED_SUPERFRAME_SIZE
static int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt)
av_cold int ff_wma_init(AVCodecContext *avctx, int flags2)
const uint16_t ff_wma_hgain_huffcodes[37]
int version
1 = 0x160 (WMAV1), 2 = 0x161 (WMAV2)
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
int frame_len
frame length in samples
if(ac->has_optimized_func)
static av_cold void flush(AVCodecContext *avctx)
int frame_len_bits
frame_len = 1 << frame_len_bits
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
static int wma_decode_frame(WMACodecContext *s, float **samples, int samples_offset)
#define HIGH_BAND_MAX_SIZE
int use_exp_vlc
exponent coding: 0 = lsp, 1 = vlc + delta
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len)
float frame_out[MAX_CHANNELS][BLOCK_MAX_SIZE *2]
static int16_t mult(Float11 *f1, Float11 *f2)
int exponent_high_sizes[BLOCK_NB_SIZES]
static void decode_exp_lsp(WMACodecContext *s, int ch)
decode exponents coded with LSP coefficients (same idea as Vorbis)
static unsigned int get_bits1(GetBitContext *s)
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_log(ac->avr, AV_LOG_TRACE, "%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-> in
static void skip_bits(GetBitContext *s, int n)
int block_num
block number in current frame
int use_noise_coding
true if perceptual noise is added
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
int use_variable_block_len
uint8_t ms_stereo
true if mid/side stereo mode
FFTContext mdct_ctx[BLOCK_NB_SIZES]
const uint32_t ff_aac_scalefactor_code[121]
int exponents_bsize[MAX_CHANNELS]
log2 ratio frame/exp. length
float coefs[MAX_CHANNELS][BLOCK_MAX_SIZE]
int prev_block_len_bits
log2 of prev block length
int coefs_end[BLOCK_NB_SIZES]
max number of coded coefficients
float lsp_pow_e_table[256]
const float ff_wma_lsp_codebook[NB_LSP_COEFS][16]
common internal api header.
void(* vector_fmul_add)(float *dst, const float *src0, const float *src1, const float *src2, int len)
Calculate the product of two vectors of floats, add a third vector of floats and store the result in ...
static av_cold int init(AVCodecParserContext *s)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
WMACoef coefs1[MAX_CHANNELS][BLOCK_MAX_SIZE]
static const uint8_t * align_get_bits(GetBitContext *s)
static const struct twinvq_data tab
static enum AVSampleFormat sample_fmts[]
float max_exponent[MAX_CHANNELS]
void(* butterflies_float)(float *restrict v1, float *restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
int coefs_start
first coded coef
int block_len_bits
log2 of current block length
uint8_t ** extended_data
pointers to the data planes/channels.
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
int high_band_coded[MAX_CHANNELS][HIGH_BAND_MAX_SIZE]
float noise_table[NOISE_TAB_SIZE]
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the product of two vectors of floats, and store the result in a vector of floats...