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36 #define VMD_HEADER_SIZE 0x0330
37 #define BYTES_PER_FRAME_RECORD 16
77 if ((!
w ||
w > 2048 || !
h ||
h > 2048) &&
90 unsigned int toc_offset;
91 unsigned char *raw_frame_table;
92 int raw_frame_table_size;
93 int64_t current_offset;
96 unsigned int total_frames;
97 int64_t current_audio_pts = 0;
123 vst->codecpar->codec_tag = 0;
124 vst->codecpar->width =
width;
125 vst->codecpar->height =
height;
126 if(vmd->
is_indeo3 && vst->codecpar->width > 320){
127 vst->codecpar->width >>= 1;
128 vst->codecpar->height >>= 1;
172 av_reduce(&num, &den, num, den, (1UL<<31)-1);
183 raw_frame_table =
NULL;
191 raw_frame_table =
av_malloc(raw_frame_table_size);
197 if (
avio_read(pb, raw_frame_table, raw_frame_table_size) !=
198 raw_frame_table_size) {
206 current_offset =
AV_RL32(&raw_frame_table[6 *
i + 2]);
221 if (
size > INT_MAX / 2) {
238 if(!current_audio_pts)
239 current_audio_pts += sound_buffers - 1;
252 current_offset +=
size;
303 (
frame->frame_record[0] == 0x02) ?
"video" :
"audio",
static void error(const char *err)
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.
enum AVMediaType codec_type
General type of the encoded data.
#define AVERROR_EOF
End of file.
#define AV_CH_LAYOUT_MONO
static int vmd_read_close(AVFormatContext *s)
unsigned int frames_per_block
uint32_t codec_tag
Additional information about the codec (corresponds to the AVI FOURCC).
int buf_size
Size of buf except extra allocated bytes.
static av_cold int read_close(AVFormatContext *ctx)
static av_always_inline int64_t avio_tell(AVIOContext *s)
ftell() equivalent for AVIOContext.
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 type
int av_reduce(int *dst_num, int *dst_den, int64_t num, int64_t den, int64_t max)
Reduce a fraction.
#define AV_CH_LAYOUT_STEREO
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
unsigned int current_frame
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static int vmd_probe(const AVProbeData *p)
unsigned char vmd_header[VMD_HEADER_SIZE]
AVCodecParameters * codecpar
Codec parameters associated with this stream.
AVInputFormat ff_vmd_demuxer
static int read_header(FFV1Context *f)
This structure contains the data a format has to probe a file.
int sample_rate
Audio only.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static int vmd_read_header(AVFormatContext *s)
int ffio_limit(AVIOContext *s, int size)
#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...
int block_align
Audio only.
#define av_malloc_array(a, b)
static int vmd_read_packet(AVFormatContext *s, AVPacket *pkt)
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
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 please make sure they are as small as space on each network bandwidth and so on benefit from smaller test cases Also keep in mind older checkouts use existing sample that means in practice generally do not remove or overwrite files as it likely would break older checkouts or releases Also all needed samples for a commit should be ideally before the push If you need an account for frequently uploading samples or you wish to help others by doing that send a mail to ffmpeg devel rsync vauL Duo ug o o w
int index
stream index in AVFormatContext
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
#define BYTES_PER_FRAME_RECORD
int bits_per_coded_sample
The number of bits per sample in the codedwords.
int64_t audio_sample_counter
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
This structure stores compressed data.
int64_t pos
byte position in stream, -1 if unknown
uint64_t channel_layout
Audio only.
int64_t bit_rate
The average bitrate of the encoded data (in bits per second).
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
unsigned char frame_record[BYTES_PER_FRAME_RECORD]