Go to the documentation of this file.
29 #define MIN_PACKET_SIZE 16
30 #define MAX_PACKET_SIZE 0x104000
34 static const uint8_t dca2wav_norm[28] = {
35 2, 0, 1, 9, 10, 3, 8, 4, 5, 9, 10, 6, 7, 12,
36 13, 14, 3, 6, 7, 11, 12, 14, 16, 15, 17, 8, 4, 5,
39 static const uint8_t dca2wav_wide[28] = {
40 2, 0, 1, 4, 5, 3, 8, 4, 5, 9, 10, 6, 7, 12,
41 13, 14, 3, 9, 10, 11, 12, 14, 16, 15, 17, 8, 4, 5,
44 int dca_ch, wav_ch, nchannels = 0;
48 if (dca_mask & (1
U << dca_ch))
49 ch_remap[nchannels++] = dca_ch;
57 dca2wav = dca2wav_wide;
59 dca2wav = dca2wav_norm;
60 for (dca_ch = 0; dca_ch < 28; dca_ch++) {
61 if (dca_mask & (1 << dca_ch)) {
62 wav_ch = dca2wav[dca_ch];
63 if (!(wav_mask & (1 << wav_ch))) {
64 wav_map[wav_ch] = dca_ch;
65 wav_mask |= 1 << wav_ch;
69 for (wav_ch = 0; wav_ch < 18; wav_ch++)
70 if (wav_mask & (1 << wav_ch))
71 ch_remap[nchannels++] = wav_map[wav_ch];
80 int *coeff_l,
int nsamples,
int ch_mask)
83 int *coeff_r = coeff_l + av_popcount(ch_mask);
93 for (spkr = 0; spkr <= max_spkr; spkr++) {
94 if (!(ch_mask & (1
U << spkr)))
111 int *coeff_l,
int nsamples,
int ch_mask)
114 int *coeff_r = coeff_l + av_popcount(ch_mask);
115 const float scale = 1.0f / (1 << 15);
122 coeff_l[
pos ] * scale, nsamples);
124 coeff_r[
pos + 1] * scale, nsamples);
127 for (spkr = 0; spkr <= max_spkr; spkr++) {
128 if (!(ch_mask & (1
U << spkr)))
133 *coeff_l * scale, nsamples);
137 *coeff_r * scale, nsamples);
145 int *got_frame_ptr,
AVPacket *avpkt)
150 int input_size = avpkt->
size;
151 int i,
ret, prev_packet =
s->packet;
207 asset = &
s->exss.assets[0];
251 if (
s->xll.chset[0].freq == 96000 &&
s->core.sample_rate == 48000)
261 &&
s->xll.nchsets > 1) {
326 s->core.avctx = avctx;
327 s->exss.avctx = avctx;
328 s->xll.avctx = avctx;
329 s->lbr.avctx = avctx;
340 s->core.dcadsp = &
s->dcadsp;
341 s->xll.dcadsp = &
s->dcadsp;
342 s->lbr.dcadsp = &
s->dcadsp;
348 s->request_channel_layout = 0;
368 #define OFFSET(x) offsetof(DCAContext, x)
369 #define PARAM AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
@ AV_SAMPLE_FMT_FLTP
float, planar
static int dcadec_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define AV_LOG_WARNING
Something somehow does not look correct.
static av_cold int dcadec_close(AVCodecContext *avctx)
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
@ AV_CLASS_CATEGORY_DECODER
uint64_t channel_layout
Audio channel layout.
int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset)
#define DCA_SPEAKER_LAYOUT_5POINT0
static enum AVSampleFormat sample_fmts[]
av_cold void ff_dca_core_close(DCACoreDecoder *s)
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
int ff_dca_xll_parse(DCAXllDecoder *s, uint8_t *data, DCAExssAsset *asset)
This structure describes decoded (raw) audio or video data.
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
#define DCA_SPEAKER_LAYOUT_STEREO
#define AV_LOG_VERBOSE
Detailed information.
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
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
#define DCA_FILTER_MODE_FIXED
static av_cold int dcadec_init(AVCodecContext *avctx)
#define DCA_PACKET_RECOVERY
Sync error recovery flag.
int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame)
int avpriv_dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size)
Convert bitstream to one representation based on sync marker.
const AVProfile ff_dca_profiles[]
static av_cold void dcadec_flush(AVCodecContext *avctx)
#define DCA_SPEAKER_LAYOUT_7POINT1_WIDE
void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples, int *coeff_l, int nsamples, int ch_mask)
#define AV_CH_LAYOUT_STEREO
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples, int *coeff_l, int nsamples, int ch_mask)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static const AVOption dcadec_options[]
#define DCA_SPEAKER_LAYOUT_7POINT0_WIDE
#define DCA_SPEAKER_LAYOUT_5POINT1
#define AV_CH_LAYOUT_STEREO_DOWNMIX
#define av_assert0(cond)
assert() equivalent, that is always enabled.
void(* dmix_scale)(int32_t *dst, int scale, ptrdiff_t len)
int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask)
int ff_dca_lbr_filter_frame(DCALbrDecoder *s, AVFrame *frame)
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
#define DCA_HAS_STEREO(mask)
static void flush(AVCodecContext *avctx)
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
const char * av_default_item_name(void *ptr)
Return the context name.
#define AV_CH_LAYOUT_5POINT1
av_cold void ff_dca_init_vlcs(void)
#define AV_EF_EXPLODE
abort decoding on minor error detection
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
#define DCA_SYNCWORD_CORE_BE
#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.
av_cold void ff_dcadsp_init(DCADSPContext *s)
int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size)
#define DCA_PACKET_RESIDUAL
Core valid for residual decoding.
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some input
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
int channels
number of audio channels
av_cold int ff_dca_core_init(DCACoreDecoder *s)
#define AV_CH_LAYOUT_5POINT0
void(* dmix_add)(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)
int ff_dca_exss_parse(DCAExssParser *s, const uint8_t *data, int size)
#define i(width, name, range_min, range_max)
void(* vector_fmac_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float and add to destination vector.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
AVSampleFormat
Audio sample formats.
void av_fast_padded_malloc(void *ptr, unsigned int *size, size_t min_size)
Same behaviour av_fast_malloc but the buffer has additional AV_INPUT_BUFFER_PADDING_SIZE at the end w...
const char * name
Name of the codec implementation.
av_cold int ff_dca_lbr_init(DCALbrDecoder *s)
#define DCA_SYNCWORD_SUBSTREAM
int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth)
int ff_dca_xll_filter_frame(DCAXllDecoder *s, AVFrame *frame)
av_cold void ff_dca_lbr_flush(DCALbrDecoder *s)
av_cold void ff_dca_lbr_close(DCALbrDecoder *s)
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 dcadec_class
main external API structure.
int ff_dca_lbr_parse(DCALbrDecoder *s, uint8_t *data, DCAExssAsset *asset)
int extension_mask
Coding components used in asset.
This structure stores compressed data.
av_cold void ff_dca_xll_flush(DCAXllDecoder *s)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
av_cold void ff_dca_xll_close(DCAXllDecoder *s)
av_cold void ff_dca_core_flush(DCACoreDecoder *s)