Go to the documentation of this file.
88 if (sd->
size <
sizeof(*di)) {
104 "surround %f (%f ltrt) - lfe %f",
113 if (gain == INT32_MIN)
135 if (sd->
size <
sizeof(*rg)) {
152 if (sd->
size <
sizeof(*ast)) {
181 #if FF_API_OLD_CHANNEL_LAYOUT
184 char chlayout_str[128];
185 uint32_t checksum = 0;
189 int data_size = buf->
nb_samples * block_align;
196 s->plane_checksums = tmp_ptr;
203 s->plane_checksums[0];
206 #if FF_API_OLD_CHANNEL_LAYOUT
216 "n:%"PRId64
" pts:%s pts_time:%s pos:%"PRId64
" "
217 "fmt:%s channels:%d chlayout:%s rate:%d nb_samples:%d "
218 "checksum:%08"PRIX32
" ",
#define FF_ENABLE_DEPRECATION_WARNINGS
#define AV_LOG_WARNING
Something somehow does not look correct.
static const AVFilterPad outputs[]
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 ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static void dump_unknown(AVFilterContext *ctx, AVFrameSideData *sd)
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
This structure describes decoded (raw) audio or video data.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
int32_t album_gain
Same as track_gain, but for the whole album.
@ AV_AUDIO_SERVICE_TYPE_VOICE_OVER
const char * name
Filter name.
int nb_channels
Number of channels in this layout.
A link between two filters.
double surround_mix_level_ltrt
Absolute scale factor representing the nominal level of the surround channels during an Lt/Rt compati...
const AVFilter ff_af_ashowinfo
@ AV_FRAME_DATA_MATRIXENCODING
The data is the AVMatrixEncoding enum defined in libavutil/channel_layout.h.
AVChannelLayout ch_layout
Channel layout of the audio data.
This structure describes optional metadata relevant to a downmix procedure.
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1<< 16)) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out->ch+ch,(const uint8_t **) in->ch+ch, off *(out-> planar
A filter pad used for either input or output.
attribute_deprecated uint64_t channel_layout
Channel layout of the audio data.
@ AV_MATRIX_ENCODING_DOLBY
uint32_t * plane_checksums
Scratch space for individual plane checksums for planar audio.
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
void * av_realloc_array(void *ptr, size_t nmemb, size_t size)
static void dump_downmix(AVFilterContext *ctx, AVFrameSideData *sd)
int64_t pkt_pos
reordered pos from the last AVPacket that has been input into the decoder
@ AV_FRAME_DATA_AUDIO_SERVICE_TYPE
This side data must be associated with an audio frame and corresponds to enum AVAudioServiceType defi...
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
@ AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED
@ AV_MATRIX_ENCODING_DPLIIX
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
uint32_t track_peak
Peak track amplitude, with 100000 representing full scale (but values may overflow).
#define FILTER_INPUTS(array)
@ AV_MATRIX_ENCODING_DOLBYHEADPHONE
static av_cold void uninit(AVFilterContext *ctx)
static void dump_matrixenc(AVFilterContext *ctx, AVFrameSideData *sd)
double surround_mix_level
Absolute scale factor representing the nominal level of the surround channels during a regular downmi...
@ AV_AUDIO_SERVICE_TYPE_EMERGENCY
static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
AVAdler av_adler32_update(AVAdler adler, const uint8_t *buf, size_t len)
Calculate the Adler32 checksum of a buffer.
static const struct @328 planes[]
#define av_ts2timestr(ts, tb)
Convenience macro, the return value should be used only directly in function arguments but never stan...
@ AV_FRAME_DATA_REPLAYGAIN
ReplayGain information in the form of the AVReplayGain struct.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
An AVChannelLayout holds information about the channel layout of audio data.
int sample_rate
Sample rate of the audio data.
@ AV_MATRIX_ENCODING_NONE
static void dump_audio_service_type(AVFilterContext *ctx, AVFrameSideData *sd)
static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
double center_mix_level_ltrt
Absolute scale factor representing the nominal level of the center channel during an Lt/Rt compatible...
int32_t track_gain
Track replay gain in microbels (divide by 100000 to get the value in dB).
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
double lfe_mix_level
Absolute scale factor representing the level at which the LFE data is mixed into L/R channels during ...
#define AV_LOG_INFO
Standard information.
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 layout
int nb_samples
number of audio samples (per channel) described by this frame
double center_mix_level
Absolute scale factor representing the nominal level of the center channel during a regular downmix.
#define i(width, name, range_min, range_max)
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
uint8_t ** extended_data
pointers to the data planes/channels.
enum AVDownmixType preferred_downmix_type
Type of downmix preferred by the mastering engineer.
@ AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED
@ AV_DOWNMIX_TYPE_LORO
Lo/Ro 2-channel downmix (Stereo).
AVFrameSideData ** side_data
const char * name
Pad name.
static const AVFilterPad inputs[]
FF_ENABLE_DEPRECATION_WARNINGS int av_channel_layout_from_mask(AVChannelLayout *channel_layout, uint64_t mask)
Initialize a native channel layout from a bitmask indicating which channels are present.
@ AV_AUDIO_SERVICE_TYPE_KARAOKE
@ AV_MATRIX_ENCODING_DOLBYEX
@ AV_AUDIO_SERVICE_TYPE_COMMENTARY
@ AV_DOWNMIX_TYPE_DPLII
Lt/Rt 2-channel downmix, Dolby Pro Logic II compatible.
uint32_t album_peak
Same as track_peak, but for the whole album,.
#define AVFILTER_FLAG_METADATA_ONLY
The filter is a "metadata" filter - it does not modify the frame data in any way.
enum AVFrameSideDataType type
int av_channel_layout_describe(const AVChannelLayout *channel_layout, char *buf, size_t buf_size)
Get a human-readable string describing the channel layout properties.
#define FF_DISABLE_DEPRECATION_WARNINGS
ReplayGain information (see http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1....
Structure to hold side data for an AVFrame.
@ AV_AUDIO_SERVICE_TYPE_EFFECTS
#define FILTER_OUTPUTS(array)
@ AV_MATRIX_ENCODING_DPLIIZ
#define flags(name, subs,...)
#define av_ts2str(ts)
Convenience macro, the return value should be used only directly in function arguments but never stan...
@ AV_AUDIO_SERVICE_TYPE_DIALOGUE
static void dump_replaygain(AVFilterContext *ctx, AVFrameSideData *sd)
@ AV_DOWNMIX_TYPE_LTRT
Lt/Rt 2-channel downmix, Dolby Surround compatible.
@ AV_FRAME_DATA_DOWNMIX_INFO
Metadata relevant to a downmix procedure.
@ AV_AUDIO_SERVICE_TYPE_MAIN
@ AV_MATRIX_ENCODING_DPLII