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72 if (shift < 0x2 || shift > 0xd) {
74 }
else if (
shift < 0x8) {
90 for (
c = 0;
c < 10;
c++) {
93 if (
c&1) offs = (80 * 6 + 80 * 16 * 0 + 3 +
c*12000);
94 else offs = (80 * 6 + 80 * 16 * 3 + 3 +
c*12000);
97 if (
c&1) offs = (80 * 6 + 80 * 16 * 1 + 3 +
c*12000);
98 else offs = (80 * 6 + 80 * 16 * 4 + 3 +
c*12000);
101 if (
c&1) offs = (80 * 3 + 8 +
c*12000);
102 else offs = (80 * 5 + 48 + 5 +
c*12000);
105 offs = (80*1 + 3 + 3);
110 if (
frame[offs] == t)
132 int size, chan,
i, j,
d, of, smpls, freq,
quant, half_ch;
134 const uint8_t *as_pack;
141 smpls = as_pack[1] & 0x3f;
142 freq = as_pack[4] >> 3 & 0x07;
143 quant = as_pack[4] & 0x07;
156 ipcm = (sys->
height == 720 && !(
frame[1] & 0x0C)) ? 2 : 0;
164 for (chan = 0; chan < sys->
n_difchan; chan++) {
173 if (
quant == 1 &&
i == half_ch) {
182 for (j = 0; j < 9; j++) {
183 for (
d = 8;
d < 80;
d += 2) {
192 pcm[of * 2] =
frame[
d + 1];
193 pcm[of * 2 + 1] =
frame[
d];
195 if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)
198 lc = ((uint16_t)
frame[
d] << 4) |
199 ((uint16_t)
frame[
d + 2] >> 4);
200 rc = ((uint16_t)
frame[
d + 1] << 4) |
201 ((uint16_t)
frame[
d + 2] & 0x0f);
212 pcm[of * 2] = lc & 0xff;
213 pcm[of * 2 + 1] = lc >> 8;
218 pcm[of * 2] = rc & 0xff;
219 pcm[of * 2 + 1] = rc >> 8;
234 const uint8_t *as_pack;
235 int freq, stype, smpls,
quant,
i, ach;
238 if (!as_pack || !
c->sys) {
243 smpls = as_pack[1] & 0x3f;
244 freq = as_pack[4] >> 3 & 0x07;
245 stype = as_pack[3] & 0x1f;
246 quant = as_pack[4] & 0x07;
250 "Unrecognized audio sample rate index (%d)\n", freq);
261 ach = ((
int[4]) { 1, 0, 2, 4 })[stype];
262 if (ach == 1 &&
quant && freq == 2)
266 for (
i = 0;
i < ach;
i++) {
275 c->audio_pkt[
i].size = 0;
276 c->audio_pkt[
i].data =
c->audio_buf[
i];
277 c->audio_pkt[
i].stream_index =
c->ast[
i]->index;
280 c->audio_pkt[
i].duration = 0;
281 c->audio_pkt[
i].pos = -1;
286 c->ast[
i]->start_time = 0;
290 return (
c->sys->audio_min_samples[freq] + smpls) * 4;
295 const uint8_t *vsc_pack;
299 par =
c->vst->codecpar;
302 c->sys->time_base.den);
303 c->vst->avg_frame_rate =
av_inv_q(
c->vst->time_base);
307 apt =
frame[4] & 0x07;
308 is16_9 = (vsc_pack && ((vsc_pack[2] & 0x07) == 0x02 ||
309 (!apt && (vsc_pack[2] & 0x07) == 0x07)));
310 c->vst->sample_aspect_ratio =
c->sys->sar[is16_9];
314 return c->sys->frame_size;
319 const uint8_t *tc_pack;
324 int prevent_df =
c->sys->ltc_divisor == 25 ||
c->sys->ltc_divisor == 50;
344 c->vst->codecpar->bit_rate = 25000000;
345 c->vst->start_time = 0;
374 for (
i = 0;
i <
c->ach;
i++) {
375 if (
c->ast[
i] &&
c->audio_pkt[
i].size) {
384 c->audio_pkt[
i].size = 0;
394 uint8_t *
buf,
int buf_size, int64_t
pos)
397 uint8_t *ppcm[5] = { 0 };
401 buf_size < c->sys->frame_size) {
408 for (
i = 0;
i <
c->ach;
i++) {
409 c->audio_pkt[
i].pos =
pos;
410 c->audio_pkt[
i].size =
size;
411 c->audio_pkt[
i].pts = (
c->sys->height == 720) ? (
c->frames & ~1) :
c->frames;
412 c->audio_pkt[
i].duration = 1;
413 ppcm[
i] =
c->audio_buf[
i];
420 if (
c->sys->height == 720) {
422 c->audio_pkt[2].size =
c->audio_pkt[3].size = 0;
424 c->audio_pkt[0].size =
c->audio_pkt[1].size = 0;
443 int64_t timestamp,
int flags)
465 c->audio_pkt[0].size =
c->audio_pkt[1].size = 0;
466 c->audio_pkt[2].size =
c->audio_pkt[3].size = 0;
484 #define PARTIAL_FRAME_SIZE (3 * 80)
510 unsigned state, marker_pos = 0;
518 while ((
state & 0xffffff7f) != 0x1f07003f) {
523 if (
state == 0x003f0700 ||
state == 0xff3f0700)
542 if (!
c->dv_demux.sys) {
544 "Can't determine profile of DV input stream.\n");
550 c->dv_demux.sys->time_base);
568 if (!
c->dv_demux.sys)
570 size =
c->dv_demux.sys->frame_size;
574 }
else if (
ret == 0) {
585 int64_t timestamp,
int flags)
600 unsigned marker_pos = 0;
604 int secondary_matches = 0;
611 if ((
state & 0x0007f840) == 0x00070000) {
614 if ((
state & 0xff07ff7f) == 0x1f07003f) {
616 if ((
state & 0xffffff7f) == 0x1f07003f) {
622 if (
state == 0x003f0700 ||
state == 0xff3f0700)
624 if (
state == 0xff3f0701 &&
i - marker_pos == 80)
629 if (matches && p->
buf_size / matches < 1024 * 1024) {
630 if (matches > 4 || firstmatch ||
631 (secondary_matches >= 10 &&
632 p->
buf_size / secondary_matches < 24000))
648 .extensions =
"dv,dif",
#define AV_TIMECODE_STR_SIZE
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
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
This struct describes the properties of an encoded stream.
For DV, one packet corresponds exactly to one frame.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
#define AV_CHANNEL_LAYOUT_STEREO
int buf_size
Size of buf except extra allocated bytes.
int64_t avio_size(AVIOContext *s)
Get the filesize.
#define AV_PKT_FLAG_KEY
The packet contains a keyframe.
static void frame_offset(AVFrame *frame, int dir, int is_pal)
static int read_seek(AVFormatContext *ctx, int stream_index, int64_t timestamp, int flags)
static av_always_inline int64_t avio_tell(AVIOContext *s)
ftell() equivalent for AVIOContext.
static int quant(float coef, const float Q, const float rounding)
Quantize one coefficient.
unsigned int avio_rb32(AVIOContext *s)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define FF_ARRAY_ELEMS(a)
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int64_t data_offset
offset of the first packet
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
static int read_header(FFV1Context *f)
and forward the result(frame or status change) to the corresponding input. If nothing is possible
Rational number (pair of numerator and denominator).
This structure contains the data a format has to probe a file.
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
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
An AVChannelLayout holds information about the channel layout of audio data.
const AVDVProfile * av_dv_frame_profile(const AVDVProfile *sys, const uint8_t *frame, unsigned buf_size)
Get a DV profile for the provided compressed frame.
#define AV_NOPTS_VALUE
Undefined timestamp value.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
uint8_t audio_buf[4][8192]
#define DV_MAX_FRAME_SIZE
largest possible DV frame, in bytes (1080i50)
int avio_r8(AVIOContext *s)
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
int flags
A combination of AV_PKT_FLAG values.
struct DVPacket audio_pkt[4]
#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...
char * av_timecode_make_smpte_tc_string2(char *buf, AVRational rate, uint32_t tcsmpte, int prevent_df, int skip_field)
Get the timecode string from the SMPTE timecode format.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
static av_always_inline AVRational av_inv_q(AVRational q)
Invert a rational.
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
fseek() equivalent for AVIOContext.
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
uint8_t * buf
actual buffer data
#define AVIO_SEEKABLE_NORMAL
Seeking works like for a local file.
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
static int shift(int a, int b)
const uint8_t(* audio_shuffle)[9]
This structure stores compressed data.
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
int64_t pos
byte position in stream, -1 if unknown
#define flags(name, subs,...)
int64_t bit_rate
The average bitrate of the encoded data (in bits per second).
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
uint8_t buf[DV_MAX_FRAME_SIZE]
int avio_feof(AVIOContext *s)
Similar to feof() but also returns nonzero on read errors.