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38 #define CHECK_LOOP(type) \
39 if (ast->loop ## type > 0) { \
40 ast->loop ## type = av_rescale_rnd(ast->loop ## type, par->sample_rate, 1000, AV_ROUND_DOWN); \
41 if (ast->loop ## type < 0 || ast->loop ## type > UINT_MAX) { \
42 av_log(s, AV_LOG_ERROR, "Invalid loop" #type " value\n"); \
43 return AVERROR(EINVAL); \
52 unsigned int codec_tag;
54 if (
s->nb_streams == 1) {
55 par =
s->streams[0]->codecpar;
114 if (
s->streams[0]->nb_frames == 0)
185 #define OFFSET(obj) offsetof(ASTMuxContext, obj)
#define AV_LOG_WARNING
Something somehow does not look correct.
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 av_always_inline void ffio_wfourcc(AVIOContext *pb, const uint8_t *s)
static int ast_write_trailer(AVFormatContext *s)
This struct describes the properties of an encoded stream.
static av_cold int end(AVCodecContext *avctx)
@ AV_CODEC_ID_PCM_S16BE_PLANAR
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
static av_always_inline int64_t avio_tell(AVIOContext *s)
ftell() equivalent for AVIOContext.
AVOutputFormat ff_ast_muxer
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const AVClass ast_muxer_class
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
const AVCodecTag ff_codec_ast_tags[]
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static const AVOption options[]
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static int write_trailer(AVFormatContext *s1)
const char * av_default_item_name(void *ptr)
Return the context name.
int sample_rate
Audio only.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
void avio_wb32(AVIOContext *s, unsigned int val)
void avio_wl32(AVIOContext *s, unsigned int val)
static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int unqueue)
int block_align
Audio only.
int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
fseek() equivalent for AVIOContext.
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 AVIO_SEEKABLE_NORMAL
Seeking works like for a local file.
int64_t avio_skip(AVIOContext *s, int64_t offset)
Skip given number of bytes forward.
void avio_wb64(AVIOContext *s, uint64_t val)
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
This structure stores compressed data.
void avio_wb16(AVIOContext *s, unsigned int val)
static int ast_write_header(AVFormatContext *s)
static void write_header(FFV1Context *f)
static int ast_write_packet(AVFormatContext *s, AVPacket *pkt)