Go to the documentation of this file.
71 bytestream_put_byte (&bs, 0x1);
75 bytestream_put_le16 (&bs, 0x0);
76 bytestream_put_byte (&bs, 0x0);
81 int tmp = 0x0, extended_toc = 0;
85 { { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 17, 0, 21, 25, 29 } },
86 { { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 18, 0, 22, 26, 30 } },
87 { { 1, 5, 9, 0, 0 }, { 0, 0, 0, 13, 15 }, { 19, 0, 23, 27, 31 } },
88 { { 2, 6, 10, 0, 0 }, { 0, 0, 0, 14, 16 }, { 20, 0, 24, 28, 32 } },
89 { { 3, 7, 11, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } },
90 { { 4, 8, 12, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } },
92 int cfg = toc_cfg[
s->packet.framesize][
s->packet.mode][
s->packet.bandwidth];
96 if (
s->packet.frames == 2) {
97 if (
s->frame[0].framebits ==
s->frame[1].framebits) {
103 }
else if (
s->packet.frames > 2) {
107 tmp |= (
s->channels > 1) << 2;
108 tmp |= (cfg - 1) << 3;
111 for (
int i = 0;
i < (
s->packet.frames - 1);
i++)
112 *fsize_needed |= (
s->frame[
i].framebits !=
s->frame[
i + 1].framebits);
113 tmp = (*fsize_needed) << 7;
115 tmp |=
s->packet.frames;
118 *
size = 1 + extended_toc;
125 const int subframesize =
s->avctx->frame_size;
130 for (
int ch = 0; ch <
f->channels; ch++) {
139 for (
int sf = 0; sf < subframes; sf++) {
140 if (sf != (subframes - 1))
145 for (
int ch = 0; ch <
f->channels; ch++) {
151 memcpy(&
b->samples[sf*subframesize],
input,
len);
156 if (sf != (subframes - 1))
164 const int subframesize =
s->avctx->frame_size;
168 for (
int ch = 0; ch <
f->channels; ch++) {
170 float m =
b->emph_coeff;
180 for (
int sf = 0; sf < subframes; sf++) {
181 for (
int ch = 0; ch <
f->channels; ch++) {
183 float m =
b->emph_coeff;
184 for (
int i = 0;
i < subframesize;
i++) {
185 float sample =
b->samples[sf*subframesize +
i];
186 b->samples[sf*subframesize +
i] =
sample - m;
189 if (sf != (subframes - 1))
198 float *
win =
s->scratch, *
temp =
s->scratch + 1920;
201 for (
int ch = 0; ch <
f->channels; ch++) {
203 float *
src1 =
b->overlap;
204 for (
int t = 0; t <
f->blocks; t++) {
210 s->tx_fn[0](
s->tx[0],
b->coeffs + t,
win,
sizeof(
float)*
f->blocks);
216 memset(
win, 0, wlen*
sizeof(
float));
217 for (
int ch = 0; ch <
f->channels; ch++) {
228 s->dsp->vector_fmul_reverse(
temp,
b->samples + rwin,
232 s->tx_fn[
f->size](
s->tx[
f->size],
b->coeffs,
win,
sizeof(
float));
236 for (
int ch = 0; ch <
f->channels; ch++) {
242 float *coeffs = &
block->coeffs[band_offset];
244 for (
int j = 0; j < band_size; j++)
245 ener += coeffs[j]*coeffs[j];
248 ener = 1.0f/
block->lin_energy[
i];
250 for (
int j = 0; j < band_size; j++)
263 int tf_select = 0,
diff = 0, tf_changed = 0, tf_select_needed;
264 int bits =
f->transient ? 2 : 4;
268 for (
int i =
f->start_band; i < f->end_band;
i++) {
270 const int tbit = (
diff ^ 1) ==
f->tf_change[
i];
275 bits =
f->transient ? 4 : 5;
281 tf_select =
f->tf_select;
284 for (
int i =
f->start_band; i < f->end_band;
i++)
290 float gain =
f->pf_gain;
291 int txval, octave =
f->pf_octave,
period =
f->pf_period, tapset =
f->pf_tapset;
298 txval =
FFMIN(octave, 6);
302 txval =
av_clip(
period - (16 << octave) + 1, 0, (1 << (4 + octave)) - 1);
304 period = txval + (16 << octave) - 1;
306 txval =
FFMIN(((
int)(gain / 0.09375
f)) - 1, 7);
308 gain = 0.09375f * (txval + 1);
315 for (
int i = 0;
i < 2;
i++) {
328 float alpha, beta, prev[2] = { 0, 0 };
339 beta = 1.0f - (4915.0f/32768.0f);
345 for (
int i =
f->start_band; i < f->end_band;
i++) {
346 for (
int ch = 0; ch <
f->channels; ch++) {
349 const float last =
FFMAX(-9.0
f, last_energy[ch][
i]);
354 }
else if (
left >= 2) {
357 }
else if (
left >= 1) {
363 prev[ch] += beta * q_en;
371 uint32_t inter, intra;
390 for (
int i =
f->start_band; i < f->end_band;
i++) {
391 if (!
f->fine_bits[
i])
393 for (
int ch = 0; ch <
f->channels; ch++) {
395 int quant, lim = (1 <<
f->fine_bits[
i]);
399 offset = 0.5f - ((
quant + 0.5f) * (1 << (14 -
f->fine_bits[
i])) / 16384.0
f);
407 for (
int priority = 0; priority < 2; priority++) {
408 for (
int i =
f->start_band; i < f->end_band && (
f->framebits -
opus_rc_tell(rc)) >=
f->channels;
i++) {
411 for (
int ch = 0; ch <
f->channels; ch++) {
413 const float err =
block->error_energy[
i];
414 const float offset = 0.5f * (1 << (14 -
f->fine_bits[
i] - 1)) / 16384.0f;
433 if (
f->framebits >= 16)
435 for (
int ch = 0; ch <
s->channels; ch++)
436 memset(
s->last_quantized_energy[ch], 0.0f,
sizeof(
float)*
CELT_MAX_BANDS);
475 if (
f->anticollapse_needed)
481 for (
int ch = 0; ch <
f->channels; ch++) {
484 s->last_quantized_energy[ch][
i] =
block->energy[
i] +
block->error_energy[
i];
491 dst[1] = v - dst[0] >> 2;
492 return 1 + (v >= 252);
504 for (
int i = 0;
i <
s->packet.frames - 1;
i++) {
506 s->frame[
i].framebits >> 3);
511 for (
int i = 0;
i <
s->packet.frames;
i++) {
513 s->frame[
i].framebits >> 3);
514 offset +=
s->frame[
i].framebits >> 3;
527 f->format =
s->avctx->sample_fmt;
528 f->nb_samples =
s->avctx->frame_size;
538 for (
int i = 0;
i <
s->channels;
i++) {
540 memset(
f->extended_data[
i], 0,
bps*
f->nb_samples);
557 if (!
s->afq.remaining_samples || !avctx->
frame_num)
569 int pad_empty =
s->packet.frames*(
frame_size/
s->avctx->frame_size) -
s->bufqueue.available + 1;
574 for (
int i = 0;
i < pad_empty;
i++) {
582 for (
int i = 0;
i <
s->packet.frames;
i++) {
584 alloc_size +=
s->frame[
i].framebits >> 3;
588 alloc_size += 2 +
s->packet.frames*2;
650 avctx->
bit_rate = coupled*(96000) + (
s->channels - coupled*2)*(48000);
652 int64_t clipped_rate =
av_clip(avctx->
bit_rate, 6000, 255000 *
s->channels);
653 av_log(avctx,
AV_LOG_ERROR,
"Unsupported bitrate %"PRId64
" kbps, clipping to %"PRId64
" kbps\n",
654 avctx->
bit_rate/1000, clipped_rate/1000);
681 for (
int ch = 0; ch <
s->channels; ch++)
682 memset(
s->last_quantized_energy[ch], 0.0f,
sizeof(
float)*
CELT_MAX_BANDS);
693 max_frames =
ceilf(
FFMIN(
s->options.max_delay_ms, 120.0f)/2.5f);
701 for (
int i = 0;
i < max_frames;
i++) {
702 s->frame[
i].dsp =
s->dsp;
703 s->frame[
i].avctx =
s->avctx;
704 s->frame[
i].seed = 0;
705 s->frame[
i].pvq =
s->pvq;
706 s->frame[
i].apply_phase_inv =
s->options.apply_phase_inv;
707 s->frame[
i].block[0].emph_coeff =
s->frame[
i].block[1].emph_coeff = 0.0f;
713 #define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
715 {
"opus_delay",
"Maximum delay in milliseconds", offsetof(
OpusEncContext,
options.max_delay_ms),
AV_OPT_TYPE_FLOAT, { .dbl =
OPUS_MAX_LOOKAHEAD }, 2.5f,
OPUS_MAX_LOOKAHEAD,
OPUSENC_FLAGS,
"max_delay_ms" },
716 {
"apply_phase_inv",
"Apply intensity stereo phase inversion", offsetof(
OpusEncContext,
options.apply_phase_inv),
AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1,
OPUSENC_FLAGS,
"apply_phase_inv" },
729 {
"compression_level",
"10" },
747 .p.supported_samplerates = (
const int []){ 48000, 0 },
int frame_size
Number of samples per channel in an audio frame.
@ AV_SAMPLE_FMT_FLTP
float, planar
const float ff_celt_postfilter_taps[3][3]
void ff_opus_rc_enc_cdf(OpusRangeCoder *rc, int val, const uint16_t *cdf)
int ff_opus_psy_process(OpusPsyContext *s, OpusPacketInfo *p)
static AVFrame * spawn_empty_frame(OpusEncContext *s)
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
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
const uint8_t ff_celt_freq_bands[]
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
#define AV_CHANNEL_LAYOUT_STEREO
int sample_rate
samples per second
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
void ff_opus_psy_celt_frame_init(OpusPsyContext *s, CeltFrame *f, int index)
#define AV_CH_LAYOUT_MONO
void ff_opus_rc_enc_uint(OpusRangeCoder *rc, uint32_t val, uint32_t size)
CELT: write a uniformly distributed integer.
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
const uint16_t ff_celt_model_tapset[]
static const AVOption opusenc_options[]
This structure describes decoded (raw) audio or video data.
#define OPUS_RC_CHECKPOINT_SPAWN(rc)
static void opus_packet_assembler(OpusEncContext *s, AVPacket *avpkt)
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
static av_cold int opus_encode_end(AVCodecContext *avctx)
int nb_channels
Number of channels in this layout.
int av_cold ff_celt_pvq_init(CeltPVQ **pvq, int encode)
static void exp_quant_coarse(OpusRangeCoder *rc, CeltFrame *f, float last_energy[][CELT_MAX_BANDS], int intra)
static __device__ float ceilf(float a)
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
static AVFrame * ff_bufqueue_get(struct FFBufQueue *queue)
Get the first buffer from the queue and remove it.
static av_always_inline uint32_t opus_rc_tell(const OpusRangeCoder *rc)
CELT: estimate bits of entropy that have thus far been consumed for the current CELT frame,...
const uint8_t ff_celt_coarse_energy_dist[4][2][42]
static float win(SuperEqualizerContext *s, float n, int N)
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
int initial_padding
Audio only.
static int opus_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
int flags
AV_CODEC_FLAG_*.
static av_always_inline float scale(float x, float s)
av_cold int ff_opus_psy_end(OpusPsyContext *s)
#define FF_CODEC_ENCODE_CB(func)
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
#define AV_CH_LAYOUT_STEREO
static int quant(float coef, const float Q, const float rounding)
Quantize one coefficient.
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define OPUS_MAX_LOOKAHEAD
#define AV_CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
#define OPUS_BLOCK_SIZE(x)
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
@ AV_TX_FLOAT_MDCT
Standard MDCT with a sample data type of float, double or int32_t, respecively.
static __device__ float floor(float a)
int(* init)(AVBSFContext *ctx)
int ff_opus_psy_celt_frame_process(OpusPsyContext *s, CeltFrame *f, int index)
#define CELT_MAX_FINE_BITS
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
#define CODEC_LONG_NAME(str)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
void av_cold ff_celt_pvq_uninit(CeltPVQ **pvq)
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default minimum maximum flags name is the option keep it simple and lowercase description are in without period
int64_t bit_rate
the average bitrate
const char * av_default_item_name(void *ptr)
Return the context name.
static void ff_bufqueue_discard_all(struct FFBufQueue *queue)
Unref and remove all buffers from the queue.
static void celt_frame_mdct(OpusEncContext *s, CeltFrame *f)
static void celt_quant_fine(CeltFrame *f, OpusRangeCoder *rc)
static const float *const ff_celt_window
static void celt_frame_setup_input(OpusEncContext *s, CeltFrame *f)
static __device__ float sqrtf(float a)
static int opus_gen_toc(OpusEncContext *s, uint8_t *toc, int *size, int *fsize_needed)
const uint8_t ff_celt_freq_range[]
void ff_opus_psy_postencode_update(OpusPsyContext *s, CeltFrame *f)
#define CELT_ENERGY_SILENCE
const OptionDef options[]
void ff_opus_rc_enc_init(OpusRangeCoder *rc)
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
An AVChannelLayout holds information about the channel layout of audio data.
#define DECLARE_ALIGNED(n, t, v)
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
static av_always_inline int diff(const struct color_info *a, const struct color_info *b, const int trans_thresh)
const int8_t ff_celt_tf_select[4][2][2][2]
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
static void ff_bufqueue_add(void *log, struct FFBufQueue *queue, AVFrame *buf)
Add a buffer to the queue.
static int write_opuslacing(uint8_t *dst, int v)
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 input
static void opus_write_extradata(AVCodecContext *avctx)
const float ff_celt_beta_coef[]
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
float last_quantized_energy[OPUS_MAX_CHANNELS][CELT_MAX_BANDS]
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
int nb_samples
number of audio samples (per channel) described by this frame
static AVFrame * ff_bufqueue_peek(struct FFBufQueue *queue, unsigned index)
Get a buffer from the queue without altering it.
const float ff_celt_window_padded[136]
#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...
void ff_opus_rc_enc_laplace(OpusRangeCoder *rc, int *value, uint32_t symbol, int decay)
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Structure holding the queue.
uint8_t ** extended_data
pointers to the data planes/channels.
void ff_opus_rc_put_raw(OpusRangeCoder *rc, uint32_t val, uint32_t count)
CELT: write 0 - 31 bits to the rawbits buffer.
static void celt_encode_frame(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f, int index)
AVSampleFormat
Audio sample formats.
#define OPUS_MAX_CHANNELS
void ff_opus_psy_signal_eof(OpusPsyContext *s)
const char * name
Name of the codec implementation.
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
#define CELT_POSTFILTER_MINPERIOD
void ff_celt_quant_bands(CeltFrame *f, OpusRangeCoder *rc)
const uint8_t ff_opus_default_coupled_streams[]
static const AVClass opusenc_class
void ff_celt_bitalloc(CeltFrame *f, OpusRangeCoder *rc, int encode)
AVTXContext * tx[CELT_BLOCK_NB]
int64_t frame_num
Frame counter, set by libavcodec.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
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
static const FFCodecDefault opusenc_defaults[]
#define AV_INPUT_BUFFER_PADDING_SIZE
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
main external API structure.
const uint16_t ff_celt_model_energy_small[]
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, size_t size)
Allocate new information of a packet.
static void celt_quant_coarse(CeltFrame *f, OpusRangeCoder *rc, float last_energy[][CELT_MAX_BANDS])
void ff_opus_rc_enc_log(OpusRangeCoder *rc, int val, uint32_t bits)
av_tx_fn tx_fn[CELT_BLOCK_NB]
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
const float ff_celt_alpha_coef[]
static void celt_enc_quant_pfilter(OpusRangeCoder *rc, CeltFrame *f)
int av_channel_layout_copy(AVChannelLayout *dst, const AVChannelLayout *src)
Make a copy of a channel layout.
static void celt_quant_final(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
const FFCodec ff_opus_encoder
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
av_cold int ff_opus_psy_init(OpusPsyContext *s, AVCodecContext *avctx, struct FFBufQueue *bufqueue, OpusEncOptions *options)
#define AV_CHANNEL_LAYOUT_MONO
static const int16_t alpha[]
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
The exact code depends on how similar the blocks are and how related they are to the block
static void celt_apply_preemph_filter(OpusEncContext *s, CeltFrame *f)
const float ff_celt_mean_energy[]
struct FFBufQueue bufqueue
#define OPUS_RC_CHECKPOINT_BITS(rc)
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
#define OPUS_RC_CHECKPOINT_ROLLBACK(rc)
int ff_alloc_packet(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and allocate data.
static void celt_enc_tf(CeltFrame *f, OpusRangeCoder *rc)
static av_cold int opus_encode_init(AVCodecContext *avctx)
void ff_opus_rc_enc_end(OpusRangeCoder *rc, uint8_t *dst, int size)