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56 switch (
frame->subbands) {
58 for (ch = 0; ch <
frame->channels; ch++) {
59 x = &
s->X[ch][
s->position - 4 *
60 s->increment +
frame->blocks * 4];
62 blk +=
s->increment) {
68 x -= 4 *
s->increment;
71 return frame->blocks * 4;
74 for (ch = 0; ch <
frame->channels; ch++) {
75 x = &
s->X[ch][
s->position - 8 *
76 s->increment +
frame->blocks * 8];
78 blk +=
s->increment) {
84 x -= 8 *
s->increment;
87 return frame->blocks * 8;
104 uint8_t crc_header[11] = { 0 };
107 uint32_t audio_sample;
111 uint32_t levels[2][8];
112 uint32_t sb_sample_delta[2][8];
121 avpkt->
data[1] = (
frame->frequency & 0x03) << 6;
122 avpkt->
data[1] |= (((
frame->blocks >> 2) - 1) & 0x03) << 4;
123 avpkt->
data[1] |= (
frame->mode & 0x03) << 2;
124 avpkt->
data[1] |= (
frame->allocation & 0x01) << 1;
125 avpkt->
data[1] |= ((
frame->subbands == 8) & 0x01) << 0;
135 crc_header[0] = avpkt->
data[1];
136 crc_header[1] = avpkt->
data[2];
143 crc_header[crc_pos >> 3] = joint;
144 crc_pos +=
frame->subbands;
147 for (ch = 0; ch <
frame->channels; ch++) {
148 for (sb = 0; sb <
frame->subbands; sb++) {
150 crc_header[crc_pos >> 3] <<= 4;
151 crc_header[crc_pos >> 3] |=
frame->scale_factor[ch][sb] & 0x0F;
158 crc_header[crc_pos >> 3] <<= 8 - (crc_pos % 8);
164 for (ch = 0; ch <
frame->channels; ch++) {
165 for (sb = 0; sb <
frame->subbands; sb++) {
166 levels[ch][sb] = ((1 <<
bits[ch][sb]) - 1) <<
167 (32 - (
frame->scale_factor[ch][sb] +
169 sb_sample_delta[ch][sb] = (uint32_t) 1 <<
170 (
frame->scale_factor[ch][sb] +
176 for (ch = 0; ch <
frame->channels; ch++) {
177 for (sb = 0; sb <
frame->subbands; sb++) {
179 if (
bits[ch][sb] == 0)
182 audio_sample = ((uint64_t) levels[ch][sb] *
183 (sb_sample_delta[ch][sb] +
184 frame->sb_sample_f[
blk][ch][sb])) >> 32;
248 / (1000000 *
frame->subbands)) - 10, 4, 16) & ~3;
270 memset(&sbc->
dsp.X, 0,
sizeof(sbc->
dsp.X));
272 sbc->
dsp.increment = sbc->
msbc ? 1 : 4;
279 const AVFrame *av_frame,
int *got_packet_ptr)
287 int frame_length = 4 + (4 *
frame->subbands *
frame->channels) / 8
288 + ((
frame->blocks *
frame->bitpool * (1 + dual)
299 if (
frame->subbands == 8)
300 sbc->
dsp.position = sbc->
dsp.sbc_enc_process_input_8s(
301 sbc->
dsp.position, av_frame->
data[0], sbc->
dsp.X,
304 sbc->
dsp.position = sbc->
dsp.sbc_enc_process_input_4s(
305 sbc->
dsp.position, av_frame->
data[0], sbc->
dsp.X,
311 j = sbc->
dsp.sbc_calc_scalefactors_j(
frame->sb_sample_f,
316 sbc->
dsp.sbc_calc_scalefactors(
frame->sb_sample_f,
328 #define OFFSET(x) offsetof(SBCEncContext, x)
329 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
331 {
"sbc_delay",
"set maximum algorithmic latency",
333 {
"msbc",
"use mSBC mode (wideband speech mono SBC)",
360 .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...
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
static int put_bytes_output(const PutBitContext *s)
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 AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#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
#define i(width, name, range_min, range_max)
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 AVCodec ff_sbc_encoder
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.
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
#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