Go to the documentation of this file.
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 };
159 int sign,
exp,
i, dln;
167 dln = ((
exp<<7) + (((d<<7)>>
exp)&0x7f)) - (
c->y>>2);
169 while (
c->tbls.quant[
i] < INT_MAX &&
c->tbls.quant[
i] < dln)
174 if (
c->code_size != 2 &&
i == 0)
187 dql =
c->tbls.iquant[
i] + (
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;
197 int I_sig= I >> (
c->code_size - 1);
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;
221 fa1 = av_clip_intp2((-
c->a[0]*
c->pk[0]*pk0)>>5, 8);
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];
240 c->dq[0].sign = I_sig;
242 c->td =
c->a[1] < -11776;
245 c->dms += (
c->tbls.F[I]<<4) + ((-
c->dms) >> 5);
246 c->dml += (
c->tbls.F[I]<<4) + ((-
c->dml) >> 7);
250 c->ap += (-
c->ap) >> 4;
251 if (
c->y <= 1535 ||
c->td ||
abs((
c->dms << 2) -
c->dml) >= (
c->dml >> 3))
256 c->yu = av_clip(
c->y +
c->tbls.W[I] + ((-
c->y)>>5), 544, 5120);
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++) {
281 c->sr[
i].mant = 1<<5;
284 for (
i=0;
i<6;
i++) {
285 c->dq[
i].mant = 1<<5;
295 #if CONFIG_ADPCM_G726_ENCODER || CONFIG_ADPCM_G726LE_ENCODER
300 i = av_mod_uintp2(
quant(
c, sig/4 -
c->se),
c->code_size);
311 c->little_endian = !strcmp(avctx->
codec->
name,
"g726le");
316 "allowed when the compliance level is higher than unofficial. "
317 "Resample or reduce the compliance level.\n");
334 c->code_size = av_clip(
c->code_size, 2, 5);
342 avctx->
frame_size = ((
int[]){ 4096, 2736, 2048, 1640 })[
c->code_size - 2];
351 const int16_t *
samples = (
const int16_t *)
frame->data[0];
360 for (
i = 0;
i <
frame->nb_samples;
i++)
361 if (
c->little_endian) {
367 if (
c->little_endian) {
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
451 c->little_endian = !strcmp(avctx->
codec->
name,
"g726le");
454 if (
c->code_size < 2 ||
c->code_size > 5) {
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;
479 frame->nb_samples = out_samples;
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,
static int sgn(int value)
int frame_size
Number of samples per channel in an audio frame.
G726Tables tbls
static tables needed for computation
static const int16_t W_tbl16[]
static av_cold int init(AVCodecContext *avctx)
static const int16_t iquant_tbl32[]
static int get_bits_left(GetBitContext *gb)
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 void put_bits_le(PutBitContext *s, int n, unsigned int value)
uint64_t channel_layout
Audio channel layout.
int sample_rate
samples per second
static enum AVSampleFormat sample_fmts[]
uint8_t mant
6 bits mantissa
const uint8_t * F
special table #2
#define AV_CH_LAYOUT_MONO
static const int16_t W_tbl32[]
int dml
long average magnitude of F[i]
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
AVCodec ff_adpcm_g726_encoder
static const int16_t iquant_tbl40[]
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
static const int16_t iquant_tbl16[]
const int * quant
quantization table
static const int quant_tbl24[]
24kbit/s 3 bits per sample
static const int quant_tbl16[]
16kbit/s 2 bits per sample
static const uint8_t F_tbl40[]
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
#define FF_COMPLIANCE_UNOFFICIAL
Allow unofficial extensions.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
const struct AVCodec * codec
static av_cold int g726_reset(G726Context *c)
static const AVCodecDefault defaults[]
static int16_t g726_decode(G726Context *c, int I)
int a[2]
second order predictor coeffs
static int16_t mult(Float11 *f1, Float11 *f2)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int little_endian
little-endian bitstream as used in aiff and Sun AU
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static const int quant_tbl40[]
40kbit/s 5 bits per sample
static uint8_t quant(G726Context *c, int d)
Paragraph 4.2.2 page 18: Adaptive quantizer.
static unsigned int get_bits_le(GetBitContext *s, int n)
static const int quant_tbl32[]
32kbit/s 4 bits per sample
static const int16_t W_tbl24[]
static const uint8_t F_tbl16[]
int dms
short average magnitude of F[i]
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
static void flush(AVCodecContext *avctx)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
int64_t bit_rate
the average bitrate
const char * av_default_item_name(void *ptr)
Return the context name.
static const G726Tables G726Tables_pool[]
int sez
estimated second order prediction
static void flush_put_bits_le(PutBitContext *s)
static const int16_t W_tbl40[]
static const int16_t iquant_tbl24[]
static const uint8_t F_tbl32[]
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
const OptionDef options[]
int y
quantizer scaling factor for the next iteration
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_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 Float11 * i2f(int i, Float11 *f)
enum AVSampleFormat sample_fmt
audio sample format
int channels
number of audio channels
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
int ap
scale factor control
#define i(width, name, range_min, range_max)
AVSampleFormat
Audio sample formats.
#define xf(width, name, var, range_min, range_max, subs,...)
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 value
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
static const uint8_t F_tbl24[]
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
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
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 see the OFFSET() macro
main external API structure.
@ AV_CODEC_ID_ADPCM_G726LE
const int16_t * iquant
inverse quantization table
AVCodec ff_adpcm_g726_decoder
static int FUNC() dqt(CodedBitstreamContext *ctx, RWContext *rw, JPEGRawQuantisationTableSpecification *current)
uint8_t exp
4 bits exponent
#define avpriv_request_sample(...)
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
This structure stores compressed data.
AVCodec ff_adpcm_g726le_encoder
const int16_t * W
special table #1 ;-)
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
int b[6]
sixth order predictor coeffs
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
static int16_t inverse_quant(G726Context *c, int i)
Paragraph 4.2.3 page 22: Inverse adaptive quantizer.
int se
estimated signal for the next iteration
AVCodec ff_adpcm_g726le_decoder