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;
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))) {
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,
243 memcpy(&p[offset], header.packet, header.bytes);
244 offset += header.bytes;
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);
287 buffer = vorbis_analysis_buffer(&s->
vd, samples);
289 int co = (channels > 8) ? c :
292 samples *
sizeof(*buffer[c]));
294 if ((ret = vorbis_analysis_wrote(&s->
vd, samples)) < 0) {
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",
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
#define AV_CH_LAYOUT_7POINT1
This structure describes decoded (raw) audio or video data.
AudioFrameQueue afq
frame queue for timestamps
#define AV_LOG_WARNING
Something somehow does not look correct.
int64_t bit_rate
the average bitrate
#define LIBAVUTIL_VERSION_INT
static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
static av_cold int init(AVCodecContext *avctx)
static const AVOption options[]
#define AV_OPT_FLAG_AUDIO_PARAM
AVVorbisParseContext * av_vorbis_parse_init(const uint8_t *extradata, int extradata_size)
Allocate and initialize the Vorbis parser using headers in the extradata.
const char * av_default_item_name(void *ptr)
Return the context name.
AVFifoBuffer * pkt_fifo
output packet buffer
#define AV_CH_LAYOUT_STEREO
static void error(const char *err)
#define AV_CH_LAYOUT_5POINT0
static int xiph_len(int l)
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.
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 AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
double iblock
impulse block bias option
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
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
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
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...
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
AVClass * av_class
class for AVOptions
A public API for Vorbis parsing.
static const AVCodecDefault defaults[]
static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
AVVorbisParseContext * vp
parse context to get durations
static const uint8_t header[24]
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
static const AVClass vorbis_class
#define AV_CH_LAYOUT_5POINT1
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void av_vorbis_parse_free(AVVorbisParseContext **s)
Free the parser and everything associated with it.
vorbis_comment vc
VorbisComment info.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int initial_padding
Audio only.
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.
int flags
AV_CODEC_FLAG_*.
simple assert() macros that are a bit more flexible than ISO C assert().
#define AV_CH_LAYOUT_QUAD
const char * name
Name of the codec implementation.
AVCodec ff_libvorbis_encoder
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
uint64_t channel_layout
Audio channel layout.
static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
int64_t rc_min_rate
minimum bitrate
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
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
#define AV_CH_FRONT_CENTER
#define AV_CH_LAYOUT_5POINT1_BACK
vorbis_info vi
vorbis_info used during init
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.
int frame_size
Number of samples per channel in an audio frame.
const uint8_t ff_vorbis_encoding_channel_layout_offsets[8][8]
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
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 ...
int sample_rate
samples per second
static int ogg_packet(AVFormatContext *s, int *sid, int *dstart, int *dsize, int64_t *fpos)
find the next Ogg packet
main external API structure.
int dsp_initialized
vd has been initialized
a very simple circular buffer FIFO implementation
static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
unsigned int av_xiphlacing(unsigned char *s, unsigned int v)
Encode extradata length to a buffer.
Describe the class of an AVClass context structure.
#define LIBVORBIS_FRAME_SIZE
#define AV_CH_LAYOUT_5POINT0_BACK
int global_quality
Global quality for codecs which cannot change it per frame.
#define AV_CH_BACK_CENTER
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.
static int vorbis_error_to_averror(int ov_err)
common internal api header.
int av_vorbis_parse_frame(AVVorbisParseContext *s, const uint8_t *buf, int buf_size)
Get the duration for a Vorbis packet.
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
int cutoff
Audio cutoff bandwidth (0 means "automatic")
AVFifoBuffer * av_fifo_alloc(unsigned int size)
Initialize an AVFifoBuffer.
int channels
number of audio channels
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
Filter the word “frame” indicates either a video frame or a group of audio samples
void av_fifo_freep(AVFifoBuffer **f)
Free an AVFifoBuffer and reset pointer to NULL.
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
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
uint8_t ** extended_data
pointers to the data planes/channels.
vorbis_dsp_state vd
DSP state used for analysis.
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
vorbis_block vb
vorbis_block used for analysis
#define AV_NOPTS_VALUE
Undefined timestamp value.
int64_t rc_max_rate
maximum bitrate