34 int i, flags1, flags2, block_align;
42 "too many channels: got %i, need %i or fewer\n",
55 "bitrate too low: got %"PRId64
", need 24000 or higher\n",
113 float n = 2.0 * 32768.0 / window_len;
136 float v, *q, max_scale, *q_end;
144 v =
ff_exp10(*exp_param++ *(1.0 / 16.0));
145 max_scale =
FFMAX(max_scale, v);
164 last_exp = *exp_param++;
165 av_assert0(last_exp - 10 >= 0 && last_exp - 10 < 32);
171 int exp = *exp_param++;
172 int code = exp - last_exp + 60;
216 mdct_norm = 1.0 / (float) n4;
218 mdct_norm *= sqrt(n4);
234 float *coefs, *exponents,
mult;
241 coefs = src_coefs[
ch];
247 for (i = 0; i <
n; i++) {
248 double t = *coefs++ / (exponents[i] *
mult);
249 if (t < -32768 || t > 32767)
252 coefs1[i] =
lrint(t);
268 for (v = total_gain - 1; v >= 127; v -= 127)
279 for (i = 0; i <
n; i++) {
292 if (parse_exponents) {
312 eptr = ptr + nb_coefs[
ch];
315 for (; ptr < eptr; ptr++) {
318 int abs_level =
FFABS(level);
320 if (abs_level <= s->
coef_vlcs[tindex]->max_level)
321 if (run < s->
coef_vlcs[tindex]->levels[abs_level - 1])
322 code = run + s->
int_table[tindex][abs_level - 1];
329 if (1 << coef_nb_bits <= abs_level)
370 int i, total_gain, ret,
error;
385 a = s->
coefs[0][i] * 0.5;
386 b = s->
coefs[1][i] * 0.5;
396 for (i = 64; i; i >>= 1) {
403 while(total_gain <= 128 && error > 0)
406 av_log(avctx,
AV_LOG_ERROR,
"Invalid input data or requested bitrate too low, cannot encode\n");
427 #if CONFIG_WMAV1_ENCODER
441 #if CONFIG_WMAV2_ENCODER
const struct AVCodec * codec
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
This structure describes decoded (raw) audio or video data.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
int64_t bit_rate
the average bitrate
int next_block_len_bits
log2 of next block length
static float win(SuperEqualizerContext *s, float n, int N)
static av_cold int init(AVCodecContext *avctx)
void avpriv_align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
int block_len
block length in samples
const uint8_t * huffbits
VLC bit size.
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats, and store the result in a vector of floats...
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
float exponents[MAX_CHANNELS][BLOCK_MAX_SIZE]
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Macro definitions for various function/variable attributes.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
float WMACoef
type for decoded coefficients, int16_t would be enough for wma 1/2
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
const uint8_t ff_aac_scalefactor_bits[121]
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
const uint32_t * huffcodes
VLC bit values.
static int encode_frame(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], uint8_t *buf, int buf_size, int total_gain)
int nb_block_sizes
number of block sizes
int ff_wma_total_gain_to_bits(int total_gain)
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static uint8_t * put_bits_ptr(PutBitContext *s)
Return the pointer to the byte where the bitstream writer will put the next bit.
static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param)
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats...
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. ...
int initial_padding
Audio only.
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
FFTSample output[BLOCK_MAX_SIZE *2]
static int put_bits_count(PutBitContext *s)
int exponent_high_bands[BLOCK_NB_SIZES][HIGH_BAND_MAX_SIZE]
int ff_wma_end(AVCodecContext *avctx)
static int16_t mult(Float11 *f1, Float11 *f2)
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
#define MAX_CODED_SUPERFRAME_SIZE
av_cold int ff_wma_init(AVCodecContext *avctx, int flags2)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
int version
1 = 0x160 (WMAV1), 2 = 0x161 (WMAV2)
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
int frame_len
frame length in samples
static void error(const char *err)
static void init_exp(WMACodecContext *s, int ch, const int *exp_param)
int frame_size
Number of samples per channel in an audio frame.
int frame_len_bits
frame_len = 1 << frame_len_bits
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
static int apply_window_and_mdct(AVCodecContext *avctx, const AVFrame *frame)
int use_exp_vlc
exponent coding: 0 = lsp, 1 = vlc + delta
main external API structure.
float frame_out[MAX_CHANNELS][BLOCK_MAX_SIZE *2]
int exponent_high_sizes[BLOCK_NB_SIZES]
static int fixed_exp(int x)
int use_noise_coding
true if perceptual noise is added
int use_variable_block_len
uint8_t ms_stereo
true if mid/side stereo mode
static av_cold int encode_init(AVCodecContext *avctx)
FFTContext mdct_ctx[BLOCK_NB_SIZES]
const uint32_t ff_aac_scalefactor_code[121]
float coefs[MAX_CHANNELS][BLOCK_MAX_SIZE]
int prev_block_len_bits
log2 of prev block length
static int encode_block(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], int total_gain)
int coefs_end[BLOCK_NB_SIZES]
max number of coded coefficients
internal math functions header
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
int channels
number of audio channels
static int parse_exponents(DBEContext *s, DBEChannel *c)
WMACoef coefs1[MAX_CHANNELS][BLOCK_MAX_SIZE]
static const CoefVLCTable coef_vlcs[6]
static enum AVSampleFormat sample_fmts[]
float max_exponent[MAX_CHANNELS]
int coefs_start
first coded coef
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
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
int high_band_coded[MAX_CHANNELS][HIGH_BAND_MAX_SIZE]
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
const CoefVLCTable * coef_vlcs[2]
#define AV_NOPTS_VALUE
Undefined timestamp value.
const float * windows[BLOCK_NB_SIZES]