Go to the documentation of this file.
21 #include <vorbis/vorbisenc.h>
38 #define LIBVORBIS_FRAME_SIZE 64
40 #define BUFFER_SIZE (1024 * 64)
77 case OV_EINVAL:
return AVERROR(EINVAL);
78 case OV_EIMPL:
return AVERROR(EINVAL);
98 if ((
ret = vorbis_encode_setup_vbr(vi, avctx->
channels,
107 if ((
ret = vorbis_encode_setup_managed(vi, avctx->
channels,
113 if (minrate == -1 && maxrate == -1)
114 if ((
ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET,
NULL)))
120 cfreq = avctx->
cutoff / 1000.0;
121 if ((
ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
127 if ((
ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &
s->iblock)))
151 "output stream will have incorrect "
152 "channel layout.\n",
name);
155 "will use Vorbis channel layout for "
160 if ((
ret = vorbis_encode_setup_init(vi)))
171 return 1 + l / 255 + l;
179 if (
s->dsp_initialized)
180 vorbis_analysis_wrote(&
s->vd, 0);
182 vorbis_block_clear(&
s->vb);
183 vorbis_dsp_clear(&
s->vd);
184 vorbis_info_clear(&
s->vi);
203 vorbis_info_init(&
s->vi);
208 if ((
ret = vorbis_analysis_init(&
s->vd, &
s->vi))) {
213 s->dsp_initialized = 1;
214 if ((
ret = vorbis_block_init(&
s->vd, &
s->vb))) {
220 vorbis_comment_init(&
s->vc);
224 if ((
ret = vorbis_analysis_headerout(&
s->vd, &
s->vc, &
header, &header_comm,
245 memcpy(&p[
offset], header_comm.packet, header_comm.bytes);
246 offset += header_comm.bytes;
247 memcpy(&p[
offset], header_code.packet, header_code.bytes);
248 offset += header_code.bytes;
257 vorbis_comment_clear(&
s->vc);
294 if ((
ret = vorbis_analysis_wrote(&
s->vd,
samples)) < 0) {
301 if (!
s->eof &&
s->afq.frame_alloc)
302 if ((
ret = vorbis_analysis_wrote(&
s->vd, 0)) < 0) {
310 while ((
ret = vorbis_analysis_blockout(&
s->vd, &
s->vb)) == 1) {
311 if ((
ret = vorbis_analysis(&
s->vb,
NULL)) < 0)
313 if ((
ret = vorbis_bitrate_addblock(&
s->vb)) < 0)
317 while ((
ret = vorbis_bitrate_flushpacket(&
s->vd, &
op)) == 1) {
380 .wrapper_name =
"libvorbis",
static void error(const char *err)
int frame_size
Number of samples per channel in an audio frame.
@ AV_SAMPLE_FMT_FLTP
float, planar
#define AV_LOG_WARNING
Something somehow does not look correct.
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 name
vorbis_comment vc
VorbisComment info
#define AV_CH_LAYOUT_5POINT0_BACK
static av_cold int init(AVCodecContext *avctx)
void av_vorbis_parse_free(AVVorbisParseContext **s)
Free the parser and everything associated with it.
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
unsigned int av_xiphlacing(unsigned char *s, unsigned int v)
Encode extradata length to a buffer.
uint64_t channel_layout
Audio channel layout.
int av_fifo_generic_write(AVFifoBuffer *f, void *src, int size, int(*func)(void *, void *, int))
Feed data from a user-supplied callback to an AVFifoBuffer.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
int sample_rate
samples per second
int64_t rc_min_rate
minimum bitrate
static const AVCodecDefault defaults[]
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
static const AVClass vorbis_class
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
This structure describes decoded (raw) audio or video data.
static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
#define LIBVORBIS_FRAME_SIZE
int av_vorbis_parse_frame(AVVorbisParseContext *s, const uint8_t *buf, int buf_size)
Get the duration for a Vorbis packet.
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
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
int av_fifo_generic_read(AVFifoBuffer *f, void *dest, int buf_size, void(*func)(void *, void *, int))
Feed data from an AVFifoBuffer to a user-supplied callback.
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
int initial_padding
Audio only.
static const AVOption options[]
const uint8_t ff_vorbis_encoding_channel_layout_offsets[8][8]
int flags
AV_CODEC_FLAG_*.
AVCodec ff_libvorbis_encoder
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
static int ogg_packet(AVFormatContext *s, int *sid, int *dstart, int *dsize, int64_t *fpos)
find the next Ogg packet
#define AV_CH_LAYOUT_STEREO
AVClass * av_class
class for AVOptions
vorbis_info vi
vorbis_info used during init
#define AV_CH_LAYOUT_QUAD
static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int av_fifo_space(const AVFifoBuffer *f)
Return the amount of space in bytes in the AVFifoBuffer, that is the amount of data you can write int...
int global_quality
Global quality for codecs which cannot change it per frame.
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
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_assert0(cond)
assert() equivalent, that is always enabled.
int64_t rc_max_rate
maximum bitrate
#define AV_OPT_FLAG_AUDIO_PARAM
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
int64_t bit_rate
the average bitrate
const char * av_default_item_name(void *ptr)
Return the context name.
#define AV_CH_LAYOUT_5POINT1
#define AV_CH_FRONT_CENTER
AudioFrameQueue afq
frame queue for timestamps
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
#define AV_NOPTS_VALUE
Undefined timestamp value.
vorbis_block vb
vorbis_block used for analysis
AVVorbisParseContext * vp
parse context to get durations
static const uint8_t header[24]
static int xiph_len(int l)
#define AV_CH_LAYOUT_5POINT1_BACK
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
static int vorbis_error_to_averror(int ov_err)
int channels
number of audio channels
#define AV_CH_LAYOUT_5POINT0
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
int cutoff
Audio cutoff bandwidth (0 means "automatic")
AVSampleFormat
Audio sample formats.
#define AV_CH_LAYOUT_7POINT1
#define AV_CH_BACK_CENTER
const char * name
Name of the codec implementation.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
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
AVVorbisParseContext * av_vorbis_parse_init(const uint8_t *extradata, int extradata_size)
Allocate and initialize the Vorbis parser using headers in the extradata.
static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
#define AV_INPUT_BUFFER_PADDING_SIZE
main external API structure.
double iblock
impulse block bias option
the frame and frame reference mechanism is intended to as much as expensive copies of that data while still allowing the filters to produce correct results The data is stored in buffers represented by AVFrame structures Several references can point to the same frame buffer
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
int dsp_initialized
vd has been initialized
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
int av_fifo_size(const AVFifoBuffer *f)
Return the amount of data in bytes in the AVFifoBuffer, that is the amount of data you can read from ...
void av_fifo_freep(AVFifoBuffer **f)
Free an AVFifoBuffer and reset pointer to NULL.
This structure stores compressed data.
AVFifoBuffer * av_fifo_alloc(unsigned int size)
Initialize an AVFifoBuffer.
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
AVFifoBuffer * pkt_fifo
output packet buffer
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
vorbis_dsp_state vd
DSP state used for analysis