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   54     switch (
frame->subbands) {
 
   56         for (ch = 0; ch < 
frame->channels; ch++) {
 
   57             x = &
s->X[ch][
s->position - 4 *
 
   58                     s->increment + 
frame->blocks * 4];
 
   60                         blk += 
s->increment) {
 
   66                 x -= 4 * 
s->increment;
 
   69         return frame->blocks * 4;
 
   72         for (ch = 0; ch < 
frame->channels; ch++) {
 
   73             x = &
s->X[ch][
s->position - 8 *
 
   74                     s->increment + 
frame->blocks * 8];
 
   76                         blk += 
s->increment) {
 
   82                 x -= 8 * 
s->increment;
 
   85         return frame->blocks * 8;
 
  102     uint8_t crc_header[11] = { 0 };
 
  105     uint32_t audio_sample;
 
  109     uint32_t levels[2][8];  
 
  110     uint32_t sb_sample_delta[2][8];
 
  119         avpkt->
data[1]  = (
frame->frequency           & 0x03) << 6;
 
  120         avpkt->
data[1] |= (((
frame->blocks >> 2) - 1) & 0x03) << 4;
 
  121         avpkt->
data[1] |= (
frame->mode                & 0x03) << 2;
 
  122         avpkt->
data[1] |= (
frame->allocation          & 0x01) << 1;
 
  123         avpkt->
data[1] |= ((
frame->subbands == 8)     & 0x01) << 0;
 
  133     crc_header[0] = avpkt->
data[1];
 
  134     crc_header[1] = avpkt->
data[2];
 
  141         crc_header[crc_pos >> 3] = joint;
 
  142         crc_pos += 
frame->subbands;
 
  145     for (ch = 0; ch < 
frame->channels; ch++) {
 
  146         for (sb = 0; sb < 
frame->subbands; sb++) {
 
  148             crc_header[crc_pos >> 3] <<= 4;
 
  149             crc_header[crc_pos >> 3] |= 
frame->scale_factor[ch][sb] & 0x0F;
 
  156         crc_header[crc_pos >> 3] <<= 8 - (crc_pos % 8);
 
  162     for (ch = 0; ch < 
frame->channels; ch++) {
 
  163         for (sb = 0; sb < 
frame->subbands; sb++) {
 
  164             levels[ch][sb] = ((1 << 
bits[ch][sb]) - 1) <<
 
  165                 (32 - (
frame->scale_factor[ch][sb] +
 
  167             sb_sample_delta[ch][sb] = (uint32_t) 1 <<
 
  168                 (
frame->scale_factor[ch][sb] +
 
  174         for (ch = 0; ch < 
frame->channels; ch++) {
 
  175             for (sb = 0; sb < 
frame->subbands; sb++) {
 
  177                 if (
bits[ch][sb] == 0)
 
  180                 audio_sample = ((uint64_t) levels[ch][sb] *
 
  181                     (sb_sample_delta[ch][sb] +
 
  182                     frame->sb_sample_f[
blk][ch][sb])) >> 32;
 
  246                                / (1000000 * 
frame->subbands)) - 10, 4, 16) & ~3;
 
  268     memset(&sbc->
dsp.X, 0, 
sizeof(sbc->
dsp.X));
 
  270     sbc->
dsp.increment = sbc->
msbc ? 1 : 4;
 
  277                             const AVFrame *av_frame, 
int *got_packet_ptr)
 
  285     int frame_length = 4 + (4 * 
frame->subbands * 
frame->channels) / 8
 
  286                      + ((
frame->blocks * 
frame->bitpool * (1 + dual)
 
  297     if (
frame->subbands == 8)
 
  298         sbc->
dsp.position = sbc->
dsp.sbc_enc_process_input_8s(
 
  299                 sbc->
dsp.position, av_frame->
data[0], sbc->
dsp.X,
 
  302         sbc->
dsp.position = sbc->
dsp.sbc_enc_process_input_4s(
 
  303                 sbc->
dsp.position, av_frame->
data[0], sbc->
dsp.X,
 
  309         j = sbc->
dsp.sbc_calc_scalefactors_j(
frame->sb_sample_f,
 
  314         sbc->
dsp.sbc_calc_scalefactors(
frame->sb_sample_f,
 
  326 #define OFFSET(x) offsetof(SBCEncContext, x) 
  327 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM 
  329     { 
"sbc_delay", 
"set maximum algorithmic latency",
 
  331     { 
"msbc",      
"use mSBC mode (wideband speech mono SBC)",
 
  358     .supported_samplerates = (
const int[]) { 16000, 32000, 44100, 48000, 0 },
 
  
int frame_size
Number of samples per channel in an audio frame.
 
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
 
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
 
#define AV_CH_LAYOUT_MONO
 
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
 
This structure describes decoded (raw) audio or video data.
 
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
 
static int sbc_analyze_audio(SBCDSPContext *s, struct sbc_frame *frame)
 
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
 
static const AVOption options[]
 
const struct AVCodec * codec
 
#define AV_CH_LAYOUT_STEREO
 
const int * supported_samplerates
array of supported audio samplerates, or NULL if unknown, array is terminated by 0
 
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
 
int global_quality
Global quality for codecs which cannot change it per frame.
 
#define FF_AVCTX_PROFILE_OPTION(name, description, type, value)
 
av_cold void ff_sbcdsp_init(SBCDSPContext *s)
 
#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 NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
 
static int sbc_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *av_frame, int *got_packet_ptr)
 
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
 
static int sbc_encode_init(AVCodecContext *avctx)
 
int channels
number of audio channels
 
#define DECLARE_ALIGNED(n, t, v)
 
#define SBC_X_BUFFER_SIZE
 
int nb_samples
number of audio samples (per channel) described by this frame
 
static int put_bits_count(PutBitContext *s)
 
AVSampleFormat
Audio sample formats.
 
@ AV_SAMPLE_FMT_S16
signed 16 bits
 
const char * name
Name of the codec implementation.
 
const AVProfile ff_sbc_profiles[]
 
uint8_t ff_sbc_crc8(const AVCRC *ctx, const uint8_t *data, size_t len)
 
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
 
static const AVClass sbc_class
 
main external API structure.
 
#define FF_PROFILE_SBC_MSBC
 
void ff_sbc_calculate_bits(const struct sbc_frame *frame, int(*bits)[8])
 
#define SBC_MODE_DUAL_CHANNEL
 
static size_t sbc_pack_frame(AVPacket *avpkt, struct sbc_frame *frame, int joint, int msbc)
 
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
 
This structure stores compressed data.
 
static const uint16_t channel_layouts[7]
 
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
 
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
 
#define SBC_MODE_JOINT_STEREO
 
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.