29 #define WS_MAX_CHANNELS 32 30 #define INF_TS 0x7FFFFFFFFFFFFFFF 106 #define LCG_A 1284865837 107 #define LCG_C 4150755663 108 #define LCG_AI 849225893 118 uint32_t
a,
c, t = *
s;
146 for (j = 0; j < 7; j++) {
166 if (b < (uint64_t)1 << 32) {
168 return ((a / b) << 32) | ((a %
b) << 32) /
b;
170 if (b < (uint64_t)1 << 48) {
171 for (i = 0; i < 4; i++) {
173 r = (r << 16) | (a / b);
178 for (i = 63; i >= 0; i--) {
179 if (a >= (uint64_t)1 << 63 || a << 1 >=
b) {
180 r |= (uint64_t)1 << i;
191 uint64_t dt = ts - (uint64_t)in->
ts_start;
192 uint64_t dt2 = dt & 1 ?
193 dt * ((dt - 1) >> 1) : (dt >> 1) * (dt - 1);
203 for (i = 0; i < ws->
nb_inter; i++) {
221 uint64_t pink_ts_next = ts & ~(
PINK_UNIT - 1);
241 int64_t dphi1, dphi2, dt, cur_ts = -0x8000000000000000;
255 for (i = 0; i < ws->
nb_inter; i++) {
257 if (edata_end - edata < 24)
273 if (edata_end - edata < 20 || avc->
sample_rate <= 0)
284 in->
ddphi = (int64_t)(dphi2 - (uint64_t)dphi1) / dt;
285 if (phi & 0x80000000) {
291 in->
phi0 = (uint64_t)phi << 33;
295 if (edata_end - edata < 8)
304 in->
amp0 = (uint64_t)a1 << 32;
305 in->
damp = (int64_t)(((uint64_t)a2 << 32) - ((uint64_t)a1 << 32)) / dt;
307 if (edata != edata_end)
319 "This implementation is limited to %d channels.\n",
350 uint32_t
c, all_ch = 0;
374 val = amp * (unsigned)pink;
380 for (c = in->
channels, cv = channels; c; c >>= 1, cv++)
382 *cv += (unsigned)val;
385 for (c = all_ch, cv = channels;
c; c >>= 1, cv++)
398 for (i = ws->
next_inter; i < ws->nb_inter; i++) {
427 if (packet->
size != 12)
439 pcm = (int16_t *)frame->
data[0];
441 memset(channels, 0, avc->
channels *
sizeof(*channels));
446 *(pcm++) = channels[
c] >> 16;
448 ws->
cur_ts += (uint64_t)duration;
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
struct ws_interval * inter
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
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
static __device__ float floor(float a)
#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
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
enum ws_interval_type type
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void wavesynth_synth_sample(struct wavesynth_context *ws, int64_t ts, int32_t *channels)
static void pink_fill(struct wavesynth_context *ws)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
const char * name
Name of the codec implementation.
static uint32_t lcg_next(uint32_t *s)
static uint64_t frac64(uint64_t a, uint64_t b)
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
int32_t pink_pool[PINK_UNIT]
static void wavesynth_enter_intervals(struct wavesynth_context *ws, int64_t ts)
Libavcodec external API header.
int sample_rate
samples per second
main external API structure.
static av_cold int wavesynth_init(AVCodecContext *avc)
static void lcg_seek(uint32_t *s, uint32_t dt)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static uint64_t phi_at(struct ws_interval *in, int64_t ts)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
AVCodec ff_ffwavesynth_decoder
common internal api header.
static void wavesynth_seek(struct wavesynth_context *ws, int64_t ts)
static int wavesynth_decode(AVCodecContext *avc, void *rframe, int *rgot_frame, AVPacket *packet)
static av_cold int wavesynth_close(AVCodecContext *avc)
int channels
number of audio channels
static int wavesynth_parse_extradata(AVCodecContext *avc)
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
#define MKTAG(a, b, c, d)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
static double val(void *priv, double ch)
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators...