33 #define FF_BUFQUEUE_SIZE 145
36 #define OPUS_MAX_LOOKAHEAD ((FF_BUFQUEUE_SIZE - 1)*2.5f)
38 #define OPUS_MAX_CHANNELS 2
41 #define OPUS_MAX_FRAMES_PER_PACKET 48
43 #define OPUS_BLOCK_SIZE(x) (2 * 15 * (1 << ((x) + 2)))
45 #define OPUS_SAMPLES_TO_BLOCK_SIZE(x) (ff_log2((x) / (2 * 15)) - 2)
81 bytestream_put_byte (&bs, 0x1);
82 bytestream_put_byte (&bs, avctx->
channels);
85 bytestream_put_le16 (&bs, 0x0);
86 bytestream_put_byte (&bs, 0x0);
91 int i,
tmp = 0x0, extended_toc = 0;
95 { { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 17, 0, 21, 25, 29 } },
96 { { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 18, 0, 22, 26, 30 } },
97 { { 1, 5, 9, 0, 0 }, { 0, 0, 0, 13, 15 }, { 19, 0, 23, 27, 31 } },
98 { { 2, 6, 10, 0, 0 }, { 0, 0, 0, 14, 16 }, { 20, 0, 24, 28, 32 } },
99 { { 3, 7, 11, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } },
100 { { 4, 8, 12, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } },
118 tmp |= (cfg - 1) << 3;
123 tmp = (*fsize_needed) << 7;
127 *size = 1 + extended_toc;
140 for (ch = 0; ch < f->
channels; ch++) {
149 for (sf = 0; sf < subframes; sf++) {
150 if (sf != (subframes - 1))
155 for (ch = 0; ch < f->
channels; ch++) {
159 const size_t left = (subframesize - cur->
nb_samples)*bps;
161 memcpy(&b->
samples[sf*subframesize], input, len);
166 if (sf != (subframes - 1))
179 for (ch = 0; ch < f->
channels; ch++) {
191 for (sf = 0; sf < subframes; sf++) {
192 for (ch = 0; ch < f->
channels; ch++) {
195 for (i = 0; i < subframesize; i++) {
197 b->
samples[sf*subframesize + i] = sample - m;
200 if (sf != (subframes - 1))
214 for (ch = 0; ch < f->
channels; ch++) {
217 for (t = 0; t < f->
blocks; t++) {
221 win[CELT_OVERLAP + i] = src2[i]*ff_celt_window[CELT_OVERLAP - i - 1];
230 for (ch = 0; ch < f->
channels; ch++) {
233 memset(win, 0, wlen*
sizeof(
float));
235 memcpy(&win[lap_dst + CELT_OVERLAP], b->
samples, rwin*
sizeof(
float));
256 for (ch = 0; ch < f->
channels; ch++) {
268 for (j = 0; j < block->
band_bins[i]; j++)
271 block->
lin_energy[i] = sqrtf(ener) + FLT_EPSILON;
274 for (j = 0; j < block->
band_bins[i]; j++)
289 int i, tf_select = 0,
diff = 0, tf_changed = 0, tf_select_needed;
294 for (i = f->
start_band; i < f->end_band; i++) {
316 int i, j, low, high, total, done, bandbits, remaining, tbits_8ths;
319 int intensitystereo_bit = 0;
320 int dualstereo_bit = 0;
343 for (i = f->
start_band; i < f->end_band; i++) {
344 int quanta, b_dynalloc, boost_amount = f->
alloc_boost[i];
349 quanta =
FFMIN(quanta << 3,
FFMAX(6 << 3, quanta));
350 b_dynalloc = dynalloc;
352 while (
opus_rc_tell_frac(rc) + (b_dynalloc << 3) < tbits_8ths && boost[i] < cap[i]) {
353 int is_boost = boost_amount--;
360 tbits_8ths -= quanta;
366 dynalloc =
FFMAX(2, dynalloc - 1);
381 if (tbits_8ths >= 1 << 3)
383 tbits_8ths -= skip_bit;
388 if (intensitystereo_bit <= tbits_8ths) {
389 tbits_8ths -= intensitystereo_bit;
390 if (tbits_8ths >= 1 << 3) {
391 dualstereo_bit = 1 << 3;
392 tbits_8ths -= 1 << 3;
395 intensitystereo_bit = 0;
400 for (i = f->
start_band; i < f->end_band; i++) {
404 int scale = duration + f->
channels - 1;
411 trim_offset[i] = trim * (band << scale) >> 6;
420 while (low <= high) {
421 int center = (low + high) >> 1;
429 bandbits =
FFMAX(0, bandbits + trim_offset[i]);
430 bandbits += boost[i];
432 if (bandbits >= threshold[i] || done) {
434 total +=
FFMIN(bandbits, cap[i]);
435 }
else if (bandbits >= f->
channels << 3)
439 if (total > tbits_8ths)
447 for (i = f->
start_band; i < f->end_band; i++) {
455 bits1[i] =
FFMAX(0, bits1[i] + trim_offset[i]);
457 bits2[i] =
FFMAX(0, bits2[i] + trim_offset[i]);
459 bits1[i] += boost[i];
460 bits2[i] += boost[i];
464 bits2[i] =
FFMAX(0, bits2[i] - bits1[i]);
471 int center = (low + high) >> 1;
477 if (bandbits >= threshold[j] || done) {
479 total +=
FFMIN(bandbits, cap[j]);
480 }
else if (bandbits >= f->
channels << 3)
483 if (total > tbits_8ths)
494 if (bandbits >= threshold[i] || done)
497 bandbits = (bandbits >= f->
channels << 3) ?
500 bandbits =
FFMIN(bandbits, cap[i]);
510 if (j == skip_startband) {
512 tbits_8ths += skip_bit;
517 remaining = tbits_8ths - total;
525 if (allocation >=
FFMAX(threshold[j], (f->
channels + 1) << 3)) {
532 allocation -= 1 << 3;
537 if (intensitystereo_bit) {
538 total -= intensitystereo_bit;
540 total += intensitystereo_bit;
547 if (intensitystereo_bit) {
552 tbits_8ths += dualstereo_bit;
553 else if (dualstereo_bit)
557 remaining = tbits_8ths - total;
560 for (i = f->
start_band; i < f->coded_bands; i++) {
568 for (i = f->
start_band; i < f->coded_bands; i++) {
570 int prev_extra = extrabits;
571 f->
pulses[i] += extrabits;
579 int fine_bits, max_bits;
582 f->
pulses[i] -= extrabits;
592 if (f->
pulses[i] + offset < 2 * (dof << 3))
594 else if (f->
pulses[i] + offset < 3 * (dof << 3))
597 fine_bits = (f->
pulses[i] + offset + (dof << 2)) / (dof << 3);
600 max_bits =
FFMAX(max_bits, 0);
602 f->
fine_bits[i] = av_clip(fine_bits, 0, max_bits);
613 f->
pulses[i] -= extrabits;
626 extrabits -= fineextra;
642 float alpha, beta, prev[2] = { 0, 0 };
653 beta = 1.0f - 4915.0f/32768.0f;
659 for (i = f->
start_band; i < f->end_band; i++) {
660 for (ch = 0; ch < f->
channels; ch++) {
668 }
else if (left >= 2) {
669 q_en = av_clip(q_en, -1, 1);
671 }
else if (left >= 1) {
672 q_en = av_clip(q_en, -1, 0);
677 prev[
ch] += beta * q_en;
685 for (i = f->
start_band; i < f->end_band; i++) {
688 for (ch = 0; ch < f->
channels; ch++) {
692 quant = av_clip(floor(diff*lim), 0, lim - 1);
694 offset = 0.5f - ((quant + 0.5f) * (1 << (14 - f->
fine_bits[i])) / 16384.0f);
703 for (priority = 0; priority < 2; priority++) {
707 for (ch = 0; ch < f->
channels; ch++) {
710 const float offset = 0.5f * (1 << (14 - f->
fine_bits[i] - 1)) / 16384.0f;
711 const int sign =
FFABS(err + offset) <
FFABS(err - offset);
721 float lowband_scratch[8 * 22];
722 float norm[2 * 8 * 100];
726 int update_lowband = 1;
727 int lowband_offset = 0;
738 float *norm2 = norm + 8 * 100;
739 int effective_lowband = -1;
747 if (i <= f->coded_bands - 1) {
754 (update_lowband || lowband_offset == 0))
761 int foldstart, foldend;
766 foldstart = lowband_offset;
768 foldend = lowband_offset - 1;
772 for (j = foldstart; j < foldend; j++) {
779 cm[0] = cm[1] = (1 << f->
blocks) - 1;
785 norm[j] = (norm[j] + norm2[j]) / 2;
790 effective_lowband != -1 ? norm + (effective_lowband << f->
size) :
NULL, f->
size,
791 norm + band_offset, 0, 1.0f, lowband_scratch, cm[0]);
794 effective_lowband != -1 ? norm2 + (effective_lowband << f->
size) :
NULL, f->
size,
795 norm2 + band_offset, 0, 1.0f, lowband_scratch, cm[1]);
798 effective_lowband != -1 ? norm + (effective_lowband << f->
size) :
NULL, f->
size,
799 norm + band_offset, 0, 1.0f, lowband_scratch, cm[0]|cm[1]);
808 update_lowband = (b > band_size << 3);
851 for (ch = 0; ch < f->
channels; ch++) {
870 *need_more = !end && *need_more;
897 memset(f->
alloc_boost, 0,
sizeof(
int)*CELT_MAX_BANDS);
905 int i,
offset, fsize_needed;
943 int i, ret,
frame_size, need_more, alloc_size = 0;
970 for (i = 0; i < pad_empty; i++) {
1003 *got_packet_ptr = 1;
1049 int64_t clipped_rate = av_clip(avctx->
bit_rate, 6000, 255000 * s->
channels);
1050 av_log(avctx,
AV_LOG_ERROR,
"Unsupported bitrate %"PRId64
" kbps, clipping to %"PRId64
" kbps\n",
1051 avctx->
bit_rate/1000, clipped_rate/1000);
1096 #define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1111 {
"compression_level",
"10" },
1128 .supported_samplerates = (
const int []){ 48000, 0 },
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static AVFrame * ff_bufqueue_get(struct FFBufQueue *queue)
Get the first buffer from the queue and remove it.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
static av_cold int opus_encode_init(AVCodecContext *avctx)
This structure describes decoded (raw) audio or video data.
enum OpusBandwidth bandwidth
static const AVCodecDefault opusenc_defaults[]
static int noise(AVBSFContext *ctx, AVPacket *out)
static const AVOption opusenc_options[]
#define OPUS_SAMPLES_TO_BLOCK_SIZE(x)
const uint8_t ff_celt_coarse_energy_dist[4][2][42]
static AVFrame * spawn_empty_frame(OpusEncContext *s)
const uint8_t ff_celt_log_freq_range[]
void ff_opus_rc_enc_init(OpusRangeCoder *rc)
float coeffs[CELT_MAX_FRAME_SIZE]
int64_t bit_rate
the average bitrate
#define LIBAVUTIL_VERSION_INT
static void opus_packet_assembler(OpusEncContext *s, AVPacket *avpkt)
static av_cold int init(AVCodecContext *avctx)
const uint8_t ff_celt_freq_bands[]
#define OPUS_MAX_LOOKAHEAD
static av_cold int opus_encode_end(AVCodecContext *avctx)
#define AV_CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
void ff_opus_rc_enc_log(OpusRangeCoder *rc, int val, uint32_t bits)
#define AV_CH_LAYOUT_STEREO
void ff_opus_rc_enc_uint(OpusRangeCoder *rc, uint32_t val, uint32_t size)
CELT: write a uniformly distributed integer.
Structure holding the queue.
const uint8_t ff_celt_band_end[]
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
int fine_priority[CELT_MAX_BANDS]
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
enum AVSampleFormat sample_fmt
audio sample format
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
#define OPUS_BLOCK_SIZE(x)
static av_cold int end(AVCodecContext *avctx)
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
av_cold int ff_mdct15_init(MDCT15Context **ps, int inverse, int N, double scale)
Init an (i)MDCT of the length 2 * 15 * (2^N)
static void celt_bitalloc(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
static const uint8_t bits2[81]
static int celt_frame_map_norm_bands(OpusEncContext *s, CeltFrame *f)
float lin_energy[CELT_MAX_BANDS]
const OptionDef options[]
static int opus_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
static void celt_enc_tf(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
const uint8_t ff_celt_log2_frac[]
static double alpha(void *priv, double x, double y)
int alloc_boost[CELT_MAX_BANDS]
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void ff_opus_psy_process(OpusEncContext *s, int end, int *need_more)
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define OPUS_MAX_FRAMES_PER_PACKET
static void celt_quant_bands(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
int initial_padding
Audio only.
int flags
AV_CODEC_FLAG_*.
int tf_change[CELT_MAX_BANDS]
const char * name
Name of the codec implementation.
int pulses[CELT_MAX_BANDS]
static int opus_gen_toc(OpusEncContext *s, uint8_t *toc, int *size, int *fsize_needed)
float samples[CELT_MAX_FRAME_SIZE]
static const uint8_t offset[127][2]
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
static void celt_frame_mdct(OpusEncContext *s, CeltFrame *f)
int fine_bits[CELT_MAX_BANDS]
uint64_t channel_layout
Audio channel layout.
float * band_coeffs[CELT_MAX_BANDS]
uint64_t channel_layout
Channel layout of the audio data.
const int8_t ff_celt_tf_select[4][2][2][2]
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
MDCT15Context * mdct[CELT_BLOCK_NB]
static void celt_frame_setup_input(OpusEncContext *s, CeltFrame *f)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
float error_energy[CELT_MAX_BANDS]
static void ff_bufqueue_discard_all(struct FFBufQueue *queue)
Unref and remove all buffers from the queue.
const uint8_t ff_celt_freq_range[]
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
const uint8_t ff_opus_default_coupled_streams[]
const float ff_celt_window[120]
void ff_opus_rc_enc_cdf(OpusRangeCoder *rc, int val, const uint16_t *cdf)
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
static void celt_quant_fine(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
int frame_size
Number of samples per channel in an audio frame.
const uint8_t ff_celt_static_caps[4][2][21]
struct FFBufQueue bufqueue
AVSampleFormat
Audio sample formats.
unsigned short available
number of available buffers
int sample_rate
samples per second
const uint16_t ff_celt_model_spread[]
main external API structure.
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...
#define CELT_MAX_FINE_BITS
static void celt_quant_coarse(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
static void celt_encode_frame(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
void ff_opus_rc_enc_laplace(OpusRangeCoder *rc, int *value, uint32_t symbol, int decay)
Describe the class of an AVClass context structure.
int band_bins[CELT_MAX_BANDS]
const uint8_t ff_celt_static_alloc[11][21]
#define CELT_ENERGY_SILENCE
static void opus_write_extradata(AVCodecContext *avctx)
Recommmends skipping the specified number of samples.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
static void ff_opus_psy_celt_frame_setup(OpusEncContext *s, CeltFrame *f, int index)
uint32_t ff_celt_encode_band(CeltFrame *f, OpusRangeCoder *rc, const int band, float *X, float *Y, int N, int b, uint32_t blocks, float *lowband, int duration, float *lowband_out, int level, float gain, float *lowband_scratch, int fill)
const uint16_t ff_celt_model_energy_small[]
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
const uint16_t ff_celt_model_alloc_trim[]
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
float energy[CELT_MAX_BANDS]
static const AVClass opusenc_class
static void celt_quant_final(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
const float ff_celt_beta_coef[]
void ff_opus_rc_put_raw(OpusRangeCoder *rc, uint32_t val, uint32_t count)
CELT: write 0 - 31 bits to the rawbits buffer.
common internal api header.
#define OPUS_MAX_CHANNELS
void ff_opus_rc_enc_end(OpusRangeCoder *rc, uint8_t *dst, int size)
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_YASM &&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
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
av_cold void ff_mdct15_uninit(MDCT15Context **ps)
Frees a context.
float last_quantized_energy[OPUS_MAX_CHANNELS][CELT_MAX_BANDS]
int cutoff
Audio cutoff bandwidth (0 means "automatic")
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
const float ff_celt_alpha_coef[]
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
const float ff_celt_mean_energy[]
static void ff_bufqueue_add(void *log, struct FFBufQueue *queue, AVFrame *buf)
Add a buffer to the queue.
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
void(* mdct)(struct MDCT15Context *s, float *dst, const float *src, ptrdiff_t stride)
Calculate a full 2N -> N MDCT.
uint8_t collapse_masks[CELT_MAX_BANDS]
This structure stores compressed data.
static void celt_apply_preemph_filter(OpusEncContext *s, CeltFrame *f)
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
static const uint8_t bits1[81]
static av_always_inline uint32_t opus_rc_tell_frac(const OpusRangeCoder *rc)
static AVFrame * ff_bufqueue_peek(struct FFBufQueue *queue, unsigned index)
Get a buffer from the queue without altering it.