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",
94 headers[0] = extradata +
offset;
95 headers[1] = extradata + offset + hsizes[0];
96 headers[2] = extradata + offset + hsizes[0] + hsizes[1];
99 "vorbis initial header len is wrong: %d\n", *p);
105 context->
op.b_o_s= i==0;
106 context->
op.bytes = hsizes[
i];
107 context->
op.packet = headers[
i];
108 if(vorbis_synthesis_headerin(&context->
vi, &context->
vc, &context->
op)<0){
120 vorbis_synthesis_init(&context->
vd, &context->
vi);
121 vorbis_block_init(&context->
vd, &context->
vb);
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;
168 output = (int16_t *)frame->
data[0];
171 op->packet = avpkt->
data;
172 op->bytes = avpkt->
size;
180 if(vorbis_synthesis(&context->
vb, op) == 0)
181 vorbis_synthesis_blockin(&context->
vd, &context->
vb) ;
186 while((samples = vorbis_synthesis_pcmout(&context->
vd, &pcm)) > 0) {
187 conv(samples, pcm, (
char*)output + total_bytes, context->
vi.channels) ;
188 total_bytes += samples * 2 * context->
vi.channels ;
190 vorbis_synthesis_read(&context->
vd, 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) ;
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
static int conv(int samples, float **pcm, char *buf, int channels)
static av_cold int init(AVCodecContext *avctx)
vorbis_dsp_state vd
DSP state used for analysis.
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
static void error(const char *err)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
vorbis_block vb
vorbis_block used for analysis
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
static int oggvorbis_decode_init(AVCodecContext *avccontext)
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
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
vorbis_comment vc
VorbisComment info.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
const char * name
Name of the codec implementation.
vorbis_info vi
vorbis_info used during init
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
Libavcodec external API header.
static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data, int *got_frame_ptr, AVPacket *avpkt)
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 ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static int oggvorbis_decode_close(AVCodecContext *avccontext)
Rational number (pair of numerator and denominator).
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
common internal api header.
AVCodec ff_libvorbis_decoder
int channels
number of audio channels
Filter the word “frame” indicates either a video frame or a group of audio samples
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
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 local context
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
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