Go to the documentation of this file.
88 #define SPDIF_FLAG_BIGENDIAN 0x01
97 {
"spdif_flags",
"IEC 61937 encapsulation flags", offsetof(
IEC61937Context, spdif_flags),
AV_OPT_TYPE_FLAGS, {.i64 = 0}, 0, INT_MAX,
AV_OPT_FLAG_ENCODING_PARAM, .unit =
"spdif_flags" },
99 {
"dtshd_rate",
"mux complete DTS frames in HD mode at the specified IEC958 rate (in Hz, default 0=disabled)", offsetof(
IEC61937Context, dtshd_rate),
AV_OPT_TYPE_INT, {.i64 = 0}, 0, 768000,
AV_OPT_FLAG_ENCODING_PARAM },
100 {
"dtshd_fallback_time",
"min secs to strip HD for after an overflow (-1: till the end, default 60)", offsetof(
IEC61937Context, dtshd_fallback),
AV_OPT_TYPE_INT, {.i64 = 60}, -1, INT_MAX,
AV_OPT_FLAG_ENCODING_PARAM },
114 int bitstream_mode =
pkt->
data[5] & 0x7;
124 static const uint8_t eac3_repeat[4] = {6, 3, 2, 1};
129 if (bsid > 10 && (
pkt->
data[4] & 0xc0) != 0xc0)
130 repeat = eac3_repeat[(
pkt->
data[4] & 0x30) >> 4];
140 if (++
ctx->hd_buf_count < repeat){
145 ctx->pkt_offset = 24576;
146 ctx->out_buf =
ctx->hd_buf[0];
147 ctx->out_bytes =
ctx->hd_buf_filled;
148 ctx->length_code =
ctx->hd_buf_filled;
150 ctx->hd_buf_count = 0;
151 ctx->hd_buf_filled = 0;
165 case 512:
return 0x0;
166 case 1024:
return 0x1;
167 case 2048:
return 0x2;
168 case 4096:
return 0x3;
169 case 8192:
return 0x4;
170 case 16384:
return 0x5;
179 static const char dtshd_start_code[10] = { 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xfe, 0xfe };
199 "impossible repetition period of %d for the current DTS stream"
200 " (blocks = %d, sample rate = %d)\n",
ctx->dtshd_rate,
period,
215 if (
sizeof(dtshd_start_code) + 2 + pkt_size
217 if (!
ctx->dtshd_skip)
219 "temporarily sending core only\n");
220 if (
ctx->dtshd_fallback > 0)
227 if (
ctx->dtshd_skip && core_size) {
228 pkt_size = core_size;
229 if (
ctx->dtshd_fallback >= 0)
233 ctx->out_bytes =
sizeof(dtshd_start_code) + 2 + pkt_size;
237 ctx->length_code =
FFALIGN(
ctx->out_bytes + 0x8, 0x10) - 0x8;
243 ctx->out_buf =
ctx->hd_buf[0];
245 memcpy(
ctx->hd_buf[0], dtshd_start_code,
sizeof(dtshd_start_code));
246 AV_WB16(
ctx->hd_buf[0] +
sizeof(dtshd_start_code), pkt_size);
247 memcpy(
ctx->hd_buf[0] +
sizeof(dtshd_start_code) + 2,
pkt->
data, pkt_size);
263 switch (syncword_dts) {
271 ctx->extra_bswap = 1;
280 ctx->extra_bswap = 1;
309 if (core_size && core_size < pkt->
size) {
310 ctx->out_bytes = core_size;
311 ctx->length_code = core_size << 3;
314 ctx->pkt_offset = blocks << 7;
316 if (
ctx->out_bytes ==
ctx->pkt_offset) {
320 ctx->use_preamble = 0;
339 int layer = 3 - ((
pkt->
data[1] >> 1) & 3);
340 int extension =
pkt->
data[2] & 1;
342 if (layer == 3 ||
version == 1) {
347 if (
version == 2 && extension) {
349 ctx->pkt_offset = 4608;
384 "%"PRIu32
" samples in AAC frame not supported\n",
samples);
396 #define MAT_PKT_OFFSET 61440
397 #define MAT_FRAME_SIZE 61424
400 0x07, 0x9E, 0x00, 0x03, 0x84, 0x01, 0x01, 0x01, 0x80, 0x00, 0x56, 0xA5, 0x3B, 0xF4, 0x81, 0x83,
401 0x49, 0x80, 0x77, 0xE0,
404 0xC3, 0xC1, 0x42, 0x49, 0x3B, 0xFA, 0x82, 0x83, 0x49, 0x80, 0x77, 0xE0,
407 0xC3, 0xC2, 0xC0, 0xC4, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x97, 0x11,
410 #define MAT_CODE(position, data) { .pos = position, .code = data, .len = sizeof(data) }
412 static const struct {
425 uint8_t *hd_buf =
ctx->hd_buf[
ctx->hd_buf_idx];
427 int padding_remaining = 0;
428 uint16_t input_timing;
429 int total_frame_size =
pkt->
size;
430 const uint8_t *dataptr =
pkt->
data;
431 int data_remaining =
pkt->
size;
447 ctx->truehd_samples_per_frame = 40 << (ratebits & 3);
449 ctx->truehd_samples_per_frame);
452 if (!
ctx->truehd_samples_per_frame)
456 if (
ctx->truehd_prev_size) {
457 uint16_t delta_samples = input_timing -
ctx->truehd_prev_time;
467 int delta_bytes = delta_samples * 2560 /
ctx->truehd_samples_per_frame;
470 padding_remaining = delta_bytes -
ctx->truehd_prev_size;
473 delta_samples, delta_bytes);
476 if (padding_remaining < 0 || padding_remaining >=
MAT_FRAME_SIZE / 2) {
478 ctx->truehd_prev_time, input_timing,
ctx->truehd_samples_per_frame);
479 padding_remaining = 0;
490 while (padding_remaining || data_remaining ||
495 int code_len =
mat_codes[next_code_idx].len;
496 int code_len_remaining = code_len;
499 ctx->hd_buf_filled += code_len;
507 ctx->out_buf = hd_buf;
508 ctx->hd_buf_idx ^= 1;
509 hd_buf =
ctx->hd_buf[
ctx->hd_buf_idx];
510 ctx->hd_buf_filled = 0;
516 if (padding_remaining) {
518 int counted_as_padding =
FFMIN(padding_remaining,
520 padding_remaining -= counted_as_padding;
521 code_len_remaining -= counted_as_padding;
524 if (code_len_remaining)
525 total_frame_size += code_len_remaining;
528 if (padding_remaining) {
532 memset(hd_buf +
ctx->hd_buf_filled, 0, padding_to_insert);
533 ctx->hd_buf_filled += padding_to_insert;
534 padding_remaining -= padding_to_insert;
536 if (padding_remaining)
540 if (data_remaining) {
544 memcpy(hd_buf +
ctx->hd_buf_filled, dataptr, data_to_insert);
545 ctx->hd_buf_filled += data_to_insert;
546 dataptr += data_to_insert;
547 data_remaining -= data_to_insert;
551 ctx->truehd_prev_size = total_frame_size;
552 ctx->truehd_prev_time = input_timing;
555 total_frame_size,
ctx->hd_buf_filled);
573 switch (
s->streams[0]->codecpar->codec_id) {
602 s->streams[0]->codecpar->codec_id);
633 ctx->use_preamble = 1;
634 ctx->extra_bswap = 0;
639 if (!
ctx->pkt_offset)
648 if (
ctx->use_preamble) {
666 if (
ctx->out_bytes & 1)
672 ctx->data_type,
ctx->out_bytes,
ctx->pkt_offset);
680 .p.extensions =
"spdif",
int truehd_samples_per_frame
samples per frame for padding calculation
static enum IEC61937DataType mpeg_data_type[2][3]
int extra_bswap
extra bswap for payload (for LE DTS => standard BE DTS)
#define AV_LOG_WARNING
Something somehow does not look correct.
@ IEC61937_MPEG2_AAC_LSF_4096
MPEG-2 AAC ADTS quarter-rate low sampling frequency.
static int spdif_header_mpeg(AVFormatContext *s, AVPacket *pkt)
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 int spdif_header_aac(AVFormatContext *s, AVPacket *pkt)
static int spdif_write_packet(struct AVFormatContext *s, AVPacket *pkt)
uint16_t truehd_prev_time
input_timing from the last frame
int truehd_prev_size
previous frame size in bytes, including any MAT codes
#define DCA_SYNCWORD_CORE_14B_BE
int pkt_offset
data burst repetition period in bytes
static const uint8_t mat_start_code[20]
int buffer_size
size of allocated buffer
static int spdif_header_truehd(AVFormatContext *s, AVPacket *pkt)
@ IEC61937_MPEG2_LAYER1_LSF
MPEG-2, layer-1 low sampling frequency.
static av_always_inline void spdif_put_16(IEC61937Context *ctx, AVIOContext *pb, unsigned int val)
static const struct @348 mat_codes[]
int av_adts_header_parse(const uint8_t *buf, uint32_t *samples, uint8_t *frames)
Extract the number of samples and frames from AAC data.
void avio_wl16(AVIOContext *s, unsigned int val)
static const AVClass spdif_class
if it could not because there are no more frames
static double val(void *priv, double ch)
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
static int spdif_header_dts4(AVFormatContext *s, AVPacket *pkt, int core_size, int sample_rate, int blocks)
#define SPDIF_FLAG_BIGENDIAN
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define FF_ARRAY_ELEMS(a)
void ff_spdif_bswap_buf16(uint16_t *dst, const uint16_t *src, int w)
void * av_fast_realloc(void *ptr, unsigned int *size, size_t min_size)
Reallocate the given buffer if it is not large enough, otherwise do nothing.
@ IEC61937_MPEG2_EXT
MPEG-2 data with extension.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
const uint32_t ff_dca_sample_rates[16]
enum IEC61937DataType data_type
burst info - reference to type of payload of the data-burst
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
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
static int spdif_header_dts(AVFormatContext *s, AVPacket *pkt)
const char * av_default_item_name(void *ptr)
Return the context name.
int hd_buf_count
number of frames in the hd audio buffer (eac3)
@ IEC61937_DTSHD
DTS HD data.
void ffio_fill(AVIOContext *s, int b, int64_t count)
#define AV_OPT_FLAG_ENCODING_PARAM
A generic parameter which can be set by the user for muxing or encoding.
@ IEC61937_DTS3
DTS type III (2048 samples)
@ IEC61937_MPEG1_LAYER23
MPEG-1 layer 2 or 3 data or MPEG-2 without extension.
static int spdif_header_eac3(AVFormatContext *s, AVPacket *pkt)
#define DCA_SYNCWORD_CORE_BE
const uint8_t * out_buf
pointer to the outgoing data before byte-swapping
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
int(* header_info)(AVFormatContext *s, AVPacket *pkt)
function, which generates codec dependent header information.
@ IEC61937_DTS2
DTS type II (1024 samples)
static const uint8_t mat_end_code[16]
int out_bytes
amount of outgoing bytes
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
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
#define DCA_SYNCWORD_CORE_14B_LE
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
int length_code
length code in bits or bytes, depending on data type
@ IEC61937_DTS1
DTS type I (512 samples)
static int spdif_write_header(AVFormatContext *s)
uint8_t * buffer
allocated buffer, used for swap bytes
#define i(width, name, range_min, range_max)
#define FF_OFMT_FLAG_MAX_ONE_OF_EACH
If this flag is set, it indicates that for each codec type whose corresponding default codec (i....
int dtshd_skip
counter used for skipping DTS-HD frames
int use_preamble
preamble enabled (disabled for exactly pre-padded DTS)
@ IEC61937_MPEG1_LAYER1
MPEG-1 layer 1.
@ IEC61937_EAC3
E-AC-3 data.
#define DCA_SYNCWORD_SUBSTREAM
@ IEC61937_MPEG2_AAC_LSF_2048
MPEG-2 AAC ADTS half-rate low sampling frequency.
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_INPUT_BUFFER_PADDING_SIZE
int hd_buf_size
size of the hd audio buffer (eac3, dts4)
#define BURST_HEADER_SIZE
static int spdif_dts4_subtype(int period)
@ IEC61937_TRUEHD
TrueHD data.
Filter the word “frame” indicates either a video frame or a group of audio samples
int hd_buf_filled
amount of bytes in the hd audio buffer (eac3, truehd)
#define MAT_CODE(position, data)
uint8_t * hd_buf[2]
allocated buffers to concatenate hd audio frames
#define DCA_SYNCWORD_CORE_LE
@ IEC61937_MPEG2_LAYER2_LSF
MPEG-2, layer-2 low sampling frequency.
#define avpriv_request_sample(...)
This structure stores compressed data.
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
static int spdif_header_ac3(AVFormatContext *s, AVPacket *pkt)
static const uint8_t mat_middle_code[12]
void avio_wb16(AVIOContext *s, unsigned int val)
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
const FFOutputFormat ff_spdif_muxer
static const AVOption options[]
@ IEC61937_MPEG2_AAC
MPEG-2 AAC ADTS.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static void spdif_deinit(AVFormatContext *s)
int hd_buf_idx
active hd buffer index (truehd)
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_WB32 unsigned int_TMPL AV_RB24
static const uint16_t spdif_mpeg_pkt_offset[2][3]
@ IEC61937_MPEG2_LAYER3_LSF
MPEG-2, layer-3 low sampling frequency.
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_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16