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67 int buf_size = avpkt->
size;
82 "Frame too small (%d bytes). Truncated file?\n", buf_size);
104 refl_rms[1] =
ff_interp(ractx, block_coefs[1], 2,
105 energy <= ractx->old_energy,
107 refl_rms[2] =
ff_interp(ractx, block_coefs[2], 3, 0, energy);
static void do_output_subblock(RA144Context *ractx, const int16_t *lpc_coefs, int gval, GetBitContext *gb)
static av_cold int init(AVCodecContext *avctx)
#define NBLOCKS
number of subblocks within a block
uint64_t channel_layout
Audio channel layout.
#define FFSWAP(type, a, b)
#define AV_CH_LAYOUT_MONO
const int16_t ff_energy_tab[32]
This structure describes decoded (raw) audio or video data.
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
av_cold void ff_audiodsp_init(AudioDSPContext *c)
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
AVCodec ff_ra_144_decoder
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
unsigned int * lpc_coef[2]
LPC coefficients: lpc_coef[0] is the coefficients of the current frame and lpc_coef[1] of the previou...
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
const int16_t *const ff_lpc_refl_cb[10]
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
static const int sizes[][2]
static int ra144_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Uncompress one block (20 bytes -> 160*2 bytes).
unsigned int lpc_tables[2][10]
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
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.
enum AVSampleFormat sample_fmt
audio sample format
unsigned int ff_rms(const int *data)
int channels
number of audio channels
void ff_int_to_int16(int16_t *out, const int *inp)
#define i(width, name, range_min, range_max)
unsigned int old_energy
previous frame energy
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
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
main external API structure.
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
This structure stores compressed data.
static av_cold int ra144_decode_init(AVCodecContext *avctx)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
unsigned int lpc_refl_rms[2]
#define BLOCKSIZE
subblock size in 16-bit words