33 int i, flags1, flags2, block_align;
40 "too many channels: got %i, need %i or fewer\n",
53 "bitrate too low: got %i, need 24000 or higher\n",
108 const float *win = s->
windows[window_index];
110 float n = 2.0 * 32768.0 / window_len;
127 float v, *q, max_scale, *q_end;
135 v = pow(10, *exp_param++ *(1.0 / 16.0));
136 max_scale =
FFMAX(max_scale, v);
155 last_exp = *exp_param++;
156 av_assert0(last_exp - 10 >= 0 && last_exp - 10 < 32);
162 int exp = *exp_param++;
163 int code = exp - last_exp + 60;
176 int v, bsize, ch, coef_nb_bits, parse_exponents;
179 static const int fixed_exp[25] = {
207 mdct_norm = 1.0 / (float) n4;
209 mdct_norm *= sqrt(n4);
225 float *coefs, *exponents,
mult;
230 mult = pow(10, total_gain * 0.05) / s->
max_exponent[ch];
232 coefs = src_coefs[ch];
238 for (i = 0; i <
n; i++) {
239 double t = *coefs++ / (exponents[i] *
mult);
240 if (t < -32768 || t > 32767)
243 coefs1[i] =
lrint(t);
259 for (v = total_gain - 1; v >= 127; v -= 127)
270 for (i = 0; i <
n; i++) {
283 if (parse_exponents) {
303 eptr = ptr + nb_coefs[ch];
306 for (; ptr < eptr; ptr++) {
309 int abs_level =
FFABS(level);
311 if (abs_level <= s->
coef_vlcs[tindex]->max_level)
312 if (run < s->
coef_vlcs[tindex]->levels[abs_level - 1])
313 code = run + s->
int_table[tindex][abs_level - 1];
320 if (1 << coef_nb_bits <= abs_level)
361 int i, total_gain,
ret, error;
373 a = s->
coefs[0][i] * 0.5;
374 b = s->
coefs[1][i] * 0.5;
384 for (i = 64; i; i >>= 1) {
391 while(total_gain <= 128 && error > 0)
394 av_log(avctx,
AV_LOG_ERROR,
"Invalid input data or requested bitrate too low, cannot encode\n");
415 #if CONFIG_WMAV1_ENCODER
429 #if CONFIG_WMAV2_ENCODER
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and/or allocate data.
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
int next_block_len_bits
log2 of next block length
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...
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.
#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
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)
#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. ...
static void apply_window_and_mdct(AVCodecContext *avctx, const AVFrame *frame)
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]
Libavcodec external API header.
static int put_bits_count(PutBitContext *s)
int exponent_high_bands[BLOCK_NB_SIZES][HIGH_BAND_MAX_SIZE]
int ff_wma_end(AVCodecContext *avctx)
int bit_rate
the average bitrate
#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)
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 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
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
int use_exp_vlc
exponent coding: 0 = lsp, 1 = vlc + delta
main external API structure.
float frame_out[MAX_CHANNELS][BLOCK_MAX_SIZE *2]
static int16_t mult(Float11 *f1, Float11 *f2)
int exponent_high_sizes[BLOCK_NB_SIZES]
int use_noise_coding
true if perceptual noise is added
static av_always_inline av_const long int lrint(double x)
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
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
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
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
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]