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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",
81 s->use_exp_vlc = flags2 & 0x0001;
82 s->use_bit_reservoir = flags2 & 0x0002;
83 s->use_variable_block_len = flags2 & 0x0004;
91 for (
i = 0;
i <
s->nb_block_sizes;
i++)
94 block_align = avctx->
bit_rate * (int64_t)
s->frame_len /
106 float **audio = (
float **)
frame->extended_data;
108 int window_index =
s->frame_len_bits -
s->block_len_bits;
111 const float *
win =
s->windows[window_index];
112 int window_len = 1 <<
s->block_len_bits;
113 float n = 2.0 * 32768.0 / window_len;
116 memcpy(
s->output,
s->frame_out[ch], window_len *
sizeof(*
s->output));
117 s->fdsp->vector_fmul_scalar(
s->frame_out[ch], audio[ch], n,
len);
118 s->fdsp->vector_fmul_reverse(&
s->output[window_len],
s->frame_out[ch],
120 s->fdsp->vector_fmul(
s->frame_out[ch],
s->frame_out[ch],
win,
len);
121 mdct->mdct_calc(mdct,
s->coefs[ch],
s->output);
136 float v, *q, max_scale, *q_end;
138 ptr =
s->exponent_bands[
s->frame_len_bits -
s->block_len_bits];
139 q =
s->exponents[ch];
140 q_end = q +
s->block_len;
144 v =
ff_exp10(*exp_param++ *(1.0 / 16.0));
145 max_scale =
FFMAX(max_scale, v);
151 s->max_exponent[ch] = max_scale;
160 ptr =
s->exponent_bands[
s->frame_len_bits -
s->block_len_bits];
161 q =
s->exponents[ch];
162 q_end = q +
s->block_len;
163 if (
s->version == 1) {
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;
197 if (
s->use_variable_block_len) {
201 s->next_block_len_bits =
s->frame_len_bits;
202 s->prev_block_len_bits =
s->frame_len_bits;
203 s->block_len_bits =
s->frame_len_bits;
206 s->block_len = 1 <<
s->block_len_bits;
208 bsize =
s->frame_len_bits -
s->block_len_bits;
211 v =
s->coefs_end[bsize] -
s->coefs_start;
212 for (ch = 0; ch <
s->avctx->channels; ch++)
215 int n4 =
s->block_len / 2;
216 mdct_norm = 1.0 / (float) n4;
218 mdct_norm *= sqrt(n4);
221 if (
s->avctx->channels == 2)
224 for (ch = 0; ch <
s->avctx->channels; ch++) {
226 s->channel_coded[ch] = 1;
227 if (
s->channel_coded[ch])
231 for (ch = 0; ch <
s->avctx->channels; ch++) {
232 if (
s->channel_coded[ch]) {
234 float *coefs, *exponents,
mult;
237 coefs1 =
s->coefs1[ch];
238 exponents =
s->exponents[ch];
241 coefs = src_coefs[ch];
242 if (
s->use_noise_coding && 0) {
245 coefs +=
s->coefs_start;
247 for (
i = 0;
i < n;
i++) {
248 double t = *coefs++ / (exponents[
i] *
mult);
249 if (t < -32768 || t > 32767)
259 for (ch = 0; ch <
s->avctx->channels; ch++) {
260 int a =
s->channel_coded[ch];
268 for (v = total_gain - 1; v >= 127; v -= 127)
274 if (
s->use_noise_coding) {
275 for (ch = 0; ch <
s->avctx->channels; ch++) {
276 if (
s->channel_coded[ch]) {
278 n =
s->exponent_high_sizes[bsize];
279 for (
i = 0;
i < n;
i++) {
280 put_bits(&
s->pb, 1,
s->high_band_coded[ch][
i] = 0);
282 nb_coefs[ch] -=
s->exponent_high_bands[bsize][
i];
289 if (
s->block_len_bits !=
s->frame_len_bits)
293 for (ch = 0; ch <
s->avctx->channels; ch++) {
294 if (
s->channel_coded[ch]) {
295 if (
s->use_exp_vlc) {
306 for (ch = 0; ch <
s->avctx->channels; ch++) {
307 if (
s->channel_coded[ch]) {
310 tindex = (ch == 1 &&
s->ms_stereo);
311 ptr = &
s->coefs1[ch][0];
312 eptr = ptr + nb_coefs[ch];
315 for (; ptr < eptr; ptr++) {
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];
326 s->coef_vlcs[tindex]->huffcodes[
code]);
329 if (1 << coef_nb_bits <= abs_level)
332 put_bits(&
s->pb, coef_nb_bits, abs_level);
342 put_bits(&
s->pb,
s->coef_vlcs[tindex]->huffbits[1],
343 s->coef_vlcs[tindex]->huffcodes[1]);
345 if (
s->version == 1 &&
s->avctx->channels >= 2)
352 uint8_t *buf,
int buf_size,
int total_gain)
356 if (
s->use_bit_reservoir)
372 s->block_len_bits =
s->frame_len_bits;
373 s->block_len = 1 <<
s->block_len_bits;
384 for (
i = 0;
i <
s->block_len;
i++) {
385 a =
s->coefs[0][
i] * 0.5;
386 b =
s->coefs[1][
i] * 0.5;
387 s->coefs[0][
i] =
a +
b;
388 s->coefs[1][
i] =
a -
b;
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
static void error(const char *err)
int frame_size
Number of samples per channel in an audio frame.
@ AV_SAMPLE_FMT_FLTP
float, planar
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
static av_cold int init(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
static const CoefVLCTable coef_vlcs[6]
int sample_rate
samples per second
static enum AVSampleFormat sample_fmts[]
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
static int apply_window_and_mdct(AVCodecContext *avctx, const AVFrame *frame)
This structure describes decoded (raw) audio or video data.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
static void init_exp(WMACodecContext *s, int ch, const int *exp_param)
static float win(SuperEqualizerContext *s, float n, int N)
const struct AVCodec * codec
float WMACoef
type for decoded coefficients, int16_t would be enough for wma 1/2
static int encode_block(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], int total_gain)
int initial_padding
Audio only.
static av_cold int encode_init(AVCodecContext *avctx)
static int16_t mult(Float11 *f1, Float11 *f2)
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int ff_wma_total_gain_to_bits(int total_gain)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static int parse_exponents(DBEContext *s, DBEChannel *c)
int64_t bit_rate
the average bitrate
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
#define MAX_CODED_SUPERFRAME_SIZE
int ff_wma_end(AVCodecContext *avctx)
void avpriv_align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
const uint8_t ff_aac_scalefactor_bits[121]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
#define AV_NOPTS_VALUE
Undefined timestamp value.
av_cold int ff_wma_init(AVCodecContext *avctx, int flags2)
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
static int fixed_exp(int x)
int channels
number of audio channels
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
#define i(width, name, range_min, range_max)
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
static int put_bits_count(PutBitContext *s)
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
AVSampleFormat
Audio sample formats.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
const char * name
Name of the codec implementation.
static int encode_frame(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], uint8_t *buf, int buf_size, int total_gain)
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
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
main external API structure.
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)
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
This structure stores compressed data.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
const uint32_t ff_aac_scalefactor_code[121]