82 #define SPDIF_FLAG_BIGENDIAN 0x01 91 {
"spdif_flags",
"IEC 61937 encapsulation flags", offsetof(
IEC61937Context,
spdif_flags),
AV_OPT_TYPE_FLAGS, {.i64 = 0}, 0, INT_MAX,
AV_OPT_FLAG_ENCODING_PARAM,
"spdif_flags" },
93 {
"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 },
94 {
"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 },
108 int bitstream_mode = pkt->
data[5] & 0x7;
118 static const uint8_t eac3_repeat[4] = {6, 3, 2, 1};
121 int bsid = pkt->
data[5] >> 3;
122 if (bsid > 10 && (pkt->
data[4] & 0xc0) != 0xc0)
123 repeat = eac3_repeat[(pkt->
data[4] & 0x30) >> 4];
157 case 512:
return 0x0;
158 case 1024:
return 0x1;
159 case 2048:
return 0x2;
160 case 4096:
return 0x3;
161 case 8192:
return 0x4;
162 case 16384:
return 0x5;
171 static const char dtshd_start_code[10] = { 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xfe, 0xfe };
172 int pkt_size = pkt->
size;
186 period = ctx->
dtshd_rate * (blocks << 5) / sample_rate;
191 "impossible repetition period of %d for the current DTS stream" 192 " (blocks = %d, sample rate = %d)\n", ctx->
dtshd_rate, period,
193 blocks << 5, sample_rate);
207 if (
sizeof(dtshd_start_code) + 2 + pkt_size
211 "temporarily sending core only\n");
220 pkt_size = core_size;
225 ctx->
out_bytes =
sizeof(dtshd_start_code) + 2 + pkt_size;
237 memcpy(ctx->
hd_buf, dtshd_start_code,
sizeof(dtshd_start_code));
239 memcpy(ctx->
hd_buf +
sizeof(dtshd_start_code) + 2, pkt->
data, pkt_size);
255 switch (syncword_dts) {
258 core_size = ((
AV_RB24(pkt->
data + 5) >> 4) & 0x3fff) + 1;
267 (((pkt->
data[5] & 0x07) << 4) | ((pkt->
data[6] & 0x3f) >> 2));
271 (((pkt->
data[4] & 0x07) << 4) | ((pkt->
data[7] & 0x3f) >> 2));
301 if (core_size && core_size < pkt->
size) {
331 int layer = 3 - ((pkt->
data[1] >> 1) & 3);
332 int extension = pkt->
data[2] & 1;
334 if (layer == 3 || version == 1) {
338 av_log(s,
AV_LOG_DEBUG,
"version: %i layer: %i extension: %i\n", version, layer, extension);
339 if (version == 2 && extension) {
376 "%"PRIu32
" samples in AAC frame not supported\n", samples);
393 #define MAT_FRAME_SIZE 61424 394 #define TRUEHD_FRAME_OFFSET 2560 395 #define MAT_MIDDLE_CODE_OFFSET -4 400 int mat_code_length = 0;
401 static const char mat_end_code[16] = { 0xC3, 0xC2, 0xC0, 0xC4, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x97, 0x11 };
404 static const char mat_start_code[20] = { 0x07, 0x9E, 0x00, 0x03, 0x84, 0x01, 0x01, 0x01, 0x80, 0x00, 0x56, 0xA5, 0x3B, 0xF4, 0x81, 0x83, 0x49, 0x80, 0x77, 0xE0 };
406 memcpy(ctx->
hd_buf, mat_start_code,
sizeof(mat_start_code));
409 static const char mat_middle_code[12] = { 0xC3, 0xC1, 0x42, 0x49, 0x3B, 0xFA, 0x82, 0x83, 0x49, 0x80, 0x77, 0xE0 };
412 mat_middle_code,
sizeof(mat_middle_code));
432 memcpy(&ctx->
hd_buf[
MAT_FRAME_SIZE -
sizeof(mat_end_code)], mat_end_code,
sizeof(mat_end_code));
552 .extensions =
"spdif",
560 .priv_class = &spdif_class,
MPEG-2 AAC ADTS half-rate low sampling frequency.
uint8_t * out_buf
pointer to the outgoing data before byte-swapping
static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int unqueue)
const char const char void * val
void avio_wl16(AVIOContext *s, unsigned int val)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
int pkt_offset
data burst repetition period in bytes
static int spdif_header_mpeg(AVFormatContext *s, AVPacket *pkt)
#define AV_LOG_WARNING
Something somehow does not look correct.
#define LIBAVUTIL_VERSION_INT
#define BURST_HEADER_SIZE
MPEG-2, layer-1 low sampling frequency.
static int spdif_dts4_subtype(int period)
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
const char * av_default_item_name(void *ptr)
Return the context name.
static int spdif_write_packet(struct AVFormatContext *s, AVPacket *pkt)
static int spdif_header_dts(AVFormatContext *s, AVPacket *pkt)
void ff_spdif_bswap_buf16(uint16_t *dst, const uint16_t *src, int w)
static int spdif_header_eac3(AVFormatContext *s, AVPacket *pkt)
int buffer_size
size of allocated buffer
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
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.
static const AVClass spdif_class
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
#define DCA_SYNCWORD_CORE_14B_BE
AVStream ** streams
A list of all streams in the file.
AVOutputFormat ff_spdif_muxer
#define MAT_MIDDLE_CODE_OFFSET
#define SPDIF_FLAG_BIGENDIAN
DTS type II (1024 samples)
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
static int spdif_write_trailer(AVFormatContext *s)
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
#define DCA_SYNCWORD_CORE_BE
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
DTS type III (2048 samples)
#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.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
preferred ID for decoding MPEG audio layer 1, 2 or 3
MPEG-1 layer 2 or 3 data or MPEG-2 without extension.
MPEG-2, layer-3 low sampling frequency.
int hd_buf_count
number of frames in the hd audio buffer
#define DCA_SYNCWORD_CORE_14B_LE
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
void ffio_fill(AVIOContext *s, int b, int count)
uint8_t * buffer
allocated buffer, used for swap bytes
int out_bytes
amount of outgoing bytes
static int write_trailer(AVFormatContext *s1)
#define TRUEHD_FRAME_OFFSET
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.
const uint32_t avpriv_dca_sample_rates[16]
static av_always_inline void spdif_put_16(IEC61937Context *ctx, AVIOContext *pb, unsigned int val)
enum IEC61937DataType data_type
burst info - reference to type of payload of the data-burst
static enum IEC61937DataType mpeg_data_type[2][3]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static const AVOption options[]
static int spdif_write_header(AVFormatContext *s)
AVIOContext * pb
I/O context.
int use_preamble
preamble enabled (disabled for exactly pre-padded DTS)
#define DCA_SYNCWORD_CORE_LE
int hd_buf_filled
amount of bytes in the hd audio buffer
Describe the class of an AVClass context structure.
int length_code
length code in bits or bytes, depending on data type
static int spdif_header_truehd(AVFormatContext *s, AVPacket *pkt)
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
void avio_wb16(AVIOContext *s, unsigned int val)
MPEG-2, layer-2 low sampling frequency.
static int spdif_header_aac(AVFormatContext *s, AVPacket *pkt)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
MPEG-2 data with extension.
int dtshd_skip
counter used for skipping DTS-HD frames
int hd_buf_size
size of the hd audio buffer
MPEG-2 AAC ADTS quarter-rate low sampling frequency.
static const uint16_t spdif_mpeg_pkt_offset[2][3]
void * priv_data
Format private data.
#define DCA_SYNCWORD_SUBSTREAM
static void write_header(FFV1Context *f)
static int spdif_header_ac3(AVFormatContext *s, AVPacket *pkt)
static int spdif_header_dts4(AVFormatContext *s, AVPacket *pkt, int core_size, int sample_rate, int blocks)
AVCodecParameters * codecpar
Codec parameters associated with this stream.
int extra_bswap
extra bswap for payload (for LE DTS => standard BE DTS)
This structure stores compressed data.
uint8_t * hd_buf
allocated buffer to concatenate hd audio frames
Common code between the AC-3 encoder and decoder.