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21 #include <vorbis/vorbisenc.h>
40 int i, hsizes[3],
ret;
48 vorbis_info_init(&
context->vi) ;
49 vorbis_comment_init(&
context->vc) ;
51 if(p[0] == 0 && p[1] == 30) {
53 for(
i = 0;
i < 3;
i++){
54 hsizes[
i] = bytestream_get_be16((
const uint8_t **)&p);
55 sizesum += 2 + hsizes[
i];
67 unsigned int sizesum = 1;
71 while((*p == 0xFF) && (sizesum < avccontext->extradata_size)) {
82 "vorbis header sizes damaged\n");
91 "vorbis header sizes: %d, %d, %d, / extradata_len is %d \n",
99 "vorbis initial header len is wrong: %d\n", *p);
133 ogg_int16_t *ptr, *
data = (ogg_int16_t*)buf ;
140 for(j = 0 ; j <
samples ; j++) {
141 *ptr = av_clip_int16(mono[j] * 32767.
f);
150 int *got_frame_ptr,
AVPacket *avpkt)
156 int samples, total_samples, total_bytes;
165 frame->nb_samples = 8192*4;
180 if(vorbis_synthesis(&
context->vb,
op) == 0)
186 while((
samples = vorbis_synthesis_pcmout(&
context->vd, &pcm)) > 0) {
193 frame->nb_samples = total_samples;
194 *got_frame_ptr = total_samples > 0;
202 vorbis_block_clear(&
context->vb);
203 vorbis_dsp_clear(&
context->vd);
204 vorbis_info_clear(&
context->vi) ;
205 vorbis_comment_clear(&
context->vc) ;
static void error(const char *err)
vorbis_dsp_state vd
DSP state used for analysis
static av_cold int init(AVCodecContext *avctx)
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
int sample_rate
samples per second
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce output
static int oggvorbis_decode_init(AVCodecContext *avccontext)
This structure describes decoded (raw) audio or video data.
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
vorbis_info vi
vorbis_info used during init
vorbis_block vb
vorbis_block used for analysis
static int ogg_packet(AVFormatContext *s, int *sid, int *dstart, int *dsize, int64_t *fpos)
find the next Ogg packet
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static int op(uint8_t **dst, const uint8_t *dst_end, GetByteContext *gb, int pixel, int count, int *x, int width, int linesize)
Perform decode operation.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static int conv(int samples, float **pcm, char *buf, int channels)
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 and describe what they for example set the foo of the bar offset is the offset of the field in your context
Rational number (pair of numerator and denominator).
AVCodec ff_libvorbis_decoder
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum AVSampleFormat sample_fmt
audio sample format
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 channels
number of audio channels
vorbis_comment vc
VorbisComment info
#define i(width, name, range_min, range_max)
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
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
main external API structure.
FFmpeg currently uses a custom build this text attempts to document some of its obscure features and options Makefile the full command issued by make and its output will be shown on the screen DBG Preprocess x86 external assembler files to a dbg asm file in the object which then gets compiled Helps in developing those assembler files DESTDIR Destination directory for the install useful to prepare packages or install FFmpeg in cross environments GEN Set to ‘1’ to generate the missing or mismatched references Makefile builds all the libraries and the executables fate Run the fate test note that you must have installed it fate list List all fate regression test targets install Install headers
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
static int oggvorbis_decode_close(AVCodecContext *avccontext)
This structure stores compressed data.
static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.