51 f->
mant = i? (i<<6) >> f->
exp : 1<<5;
60 res = (((f1->
mant * f2->
mant) + 0x30) >> 4);
61 res = exp > 19 ? res << (exp - 19) : res >> (19 - exp);
62 return (f1->
sign ^ f2->
sign) ? -res : res;
67 return (value < 0) ? -1 : 1;
104 { 116, 365, 365, 116 };
106 { -22, 439, 439, -22 };
111 { 7, 217, 330, INT_MAX };
113 { INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
115 { -4, 30, 137, 582, 582, 137, 30, -4 };
117 { 0, 1, 2, 7, 7, 2, 1, 0 };
120 { -125, 79, 177, 245, 299, 348, 399, INT_MAX };
122 { INT16_MIN, 4, 135, 213, 273, 323, 373, 425,
123 425, 373, 323, 273, 213, 135, 4, INT16_MIN };
125 { -12, 18, 41, 64, 112, 198, 355, 1122,
126 1122, 355, 198, 112, 64, 41, 18, -12};
128 { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
131 { -122, -16, 67, 138, 197, 249, 297, 338,
132 377, 412, 444, 474, 501, 527, 552, INT_MAX };
134 { INT16_MIN, -66, 28, 104, 169, 224, 274, 318,
135 358, 395, 429, 459, 488, 514, 539, 566,
136 566, 539, 514, 488, 459, 429, 395, 358,
137 318, 274, 224, 169, 104, 28, -66, INT16_MIN };
139 { 14, 14, 24, 39, 40, 41, 58, 100,
140 141, 179, 219, 280, 358, 440, 529, 696,
141 696, 529, 440, 358, 280, 219, 179, 141,
142 100, 58, 41, 40, 39, 24, 14, 14 };
144 { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
145 6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
167 dln = ((exp<<7) + (((d<<7)>>
exp)&0x7f)) - (c->
y>>2);
188 dex = (dql>>7) & 0
xf;
189 dqt = (1<<7) + (dql & 0x7f);
190 return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
195 int dq, re_signal, pk0, fa1,
i, tr, ylint, ylfrac, thr2, al, dq0;
202 ylint = (c->
yl >> 15);
203 ylfrac = (c->
yl >> 10) & 0x1f;
204 thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
205 tr= (c->
td == 1 && dq > ((3*thr2)>>2));
209 re_signal = (int16_t)(c->
se + dq);
212 pk0 = (c->
sez + dq) ?
sgn(c->
sez + dq) : 0;
213 dq0 = dq ?
sgn(dq) : 0;
223 c->
a[1] += 128*pk0*c->
pk[1] + fa1 - (c->
a[1]>>7);
224 c->
a[1] =
av_clip(c->
a[1], -12288, 12288);
225 c->
a[0] += 64*3*pk0*c->
pk[0] - (c->
a[0] >> 8);
226 c->
a[0] =
av_clip(c->
a[0], -(15360 - c->
a[1]), 15360 - c->
a[1]);
229 c->
b[i] += 128*dq0*sgn(-c->
dq[i].
sign) - (c->
b[
i]>>8);
234 c->
pk[0] = pk0 ? pk0 : 1;
236 i2f(re_signal, &c->
sr[0]);
238 c->
dq[i] = c->
dq[i-1];
242 c->
td = c->
a[1] < -11776;
250 c->
ap += (-c->
ap) >> 4;
251 if (c->
y <= 1535 || c->
td ||
abs((c->
dms << 2) - c->
dml) >= (c->
dml >> 3))
257 c->
yl += c->
yu + ((-c->
yl)>>6);
260 al = (c->
ap >= 256) ? 1<<6 : c->
ap >> 2;
261 c->
y = (c->
yl + (c->
yu - (c->
yl>>6))*al) >> 6;
272 return av_clip(re_signal * 4, -0xffff, 0xffff);
280 for (i=0; i<2; i++) {
284 for (i=0; i<6; i++) {
295 #if CONFIG_ADPCM_G726_ENCODER || CONFIG_ADPCM_G726LE_ENCODER 316 "allowed when the compliance level is higher than unofficial. " 317 "Resample or reduce the compliance level.\n");
351 const int16_t *
samples = (
const int16_t *)frame->
data[0];
378 #define OFFSET(x) offsetof(G726Context, x) 379 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM 391 #if CONFIG_ADPCM_G726_ENCODER 392 static const AVClass g726_class = {
405 .
init = g726_encode_init,
406 .encode2 = g726_encode_frame,
410 .priv_class = &g726_class,
415 #if CONFIG_ADPCM_G726LE_ENCODER 416 static const AVClass g726le_class = {
429 .
init = g726_encode_init,
430 .encode2 = g726_encode_frame,
434 .priv_class = &g726le_class,
439 #if CONFIG_ADPCM_G726_DECODER || CONFIG_ADPCM_G726LE_DECODER 466 int *got_frame_ptr,
AVPacket *avpkt)
470 int buf_size = avpkt->
size;
474 int out_samples,
ret;
476 out_samples = buf_size * 8 / c->
code_size;
482 samples = (int16_t *)frame->
data[0];
486 while (out_samples--)
506 #if CONFIG_ADPCM_G726_DECODER 513 .
init = g726_decode_init,
514 .
decode = g726_decode_frame,
515 .
flush = g726_decode_flush,
520 #if CONFIG_ADPCM_G726LE_DECODER 526 .
init = g726_decode_init,
527 .
decode = g726_decode_frame,
528 .
flush = g726_decode_flush,
const struct AVCodec * codec
static const int16_t iquant_tbl40[]
static const int quant_tbl32[]
32kbit/s 4 bits per sample
static av_cold int g726_reset(G726Context *c)
This structure describes decoded (raw) audio or video data.
const int16_t * W
special table #1 ;-)
ptrdiff_t const GLvoid * data
static void flush(AVCodecContext *avctx)
static int FUNC() dqt(CodedBitstreamContext *ctx, RWContext *rw, JPEGRawQuantisationTableSpecification *current)
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static int16_t inverse_quant(G726Context *c, int i)
Paragraph 4.2.3 page 22: Inverse adaptive quantizer.
int64_t bit_rate
the average bitrate
#define LIBAVUTIL_VERSION_INT
static const int quant_tbl16[]
16kbit/s 2 bits per sample
static av_cold int init(AVCodecContext *avctx)
uint8_t exp
4 bits exponent
#define avpriv_request_sample(...)
const char * av_default_item_name(void *ptr)
Return the context name.
const uint8_t * F
special table #2
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
static const uint8_t F_tbl24[]
static const int16_t W_tbl24[]
int b[6]
sixth order predictor coeffs
static int sgn(int value)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
int av_log2_16bit(unsigned v)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
const int * quant
quantization table
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
enum AVSampleFormat sample_fmt
audio sample format
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
static const int16_t iquant_tbl16[]
GLsizei GLboolean const GLfloat * value
static const int16_t W_tbl16[]
static const int quant_tbl24[]
24kbit/s 3 bits per sample
static uint8_t quant(G726Context *c, int d)
Paragraph 4.2.2 page 18: Adaptive quantizer.
bitstream reader API header.
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
int ap
scale factor control
G726Tables tbls
static tables needed for computation
#define FF_COMPLIANCE_UNOFFICIAL
Allow unofficial extensions.
static int get_bits_left(GetBitContext *gb)
AVCodec ff_adpcm_g726_decoder
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static Float11 * i2f(int i, Float11 *f)
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 see the OFFSET() macro
const char * name
Name of the codec implementation.
static const AVCodecDefault defaults[]
AVCodec ff_adpcm_g726_encoder
static const int16_t W_tbl32[]
uint64_t channel_layout
Audio channel layout.
static void flush_put_bits_le(PutBitContext *s)
AVCodec ff_adpcm_g726le_decoder
static const int16_t W_tbl40[]
static void put_bits_le(PutBitContext *s, int n, BitBuf value)
audio channel layout utility functions
static int16_t mult(Float11 *f1, Float11 *f2)
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
static const uint8_t F_tbl32[]
static const uint8_t F_tbl40[]
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
int se
estimated signal for the next iteration
static const int16_t iquant_tbl24[]
static const G726Tables G726Tables_pool[]
static const int quant_tbl40[]
40kbit/s 5 bits per sample
int a[2]
second order predictor coeffs
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
int frame_size
Number of samples per channel in an audio frame.
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
uint8_t mant
6 bits mantissa
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
int little_endian
little-endian bitstream as used in aiff and Sun AU
Describe the class of an AVClass context structure.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static int16_t g726_decode(G726Context *c, int I)
static unsigned int get_bits_le(GetBitContext *s, int n)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
int dml
long average magnitude of F[i]
static const uint8_t F_tbl16[]
static const int16_t iquant_tbl32[]
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
AVCodec ff_adpcm_g726le_encoder
int y
quantizer scaling factor for the next iteration
#define xf(width, name, var, range_min, range_max, subs,...)
const int16_t * iquant
inverse quantization table
int channels
number of audio channels
int sez
estimated second order prediction
static enum AVSampleFormat sample_fmts[]
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
int dms
short average magnitude of F[i]
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).