32 #define FF_INTERNAL_FIELDS 1
63 #define NB_ITEMS(list) (list ## _size / sizeof(*list))
73 for (
i = 0;
i < nb_counts;
i++)
76 for (
i = lc = 0;
i < nb_layouts;
i++) {
78 if (n < 64 && (counts & ((uint64_t)1 << n)))
80 "Removing channel layout 0x%"PRIx64
", redundant with %d channels\n",
151 #if FF_API_BUFFERSINK_ALLOC
188 "%d buffers queued in %s, something may be wrong.\n",
206 #define MAKE_AVFILTERLINK_ACCESSOR(type, field) \
207 type av_buffersink_get_##field(const AVFilterContext *ctx) { \
208 av_assert0(ctx->filter->activate == activate); \
209 return ctx->inputs[0]->field; \
227 #define CHECK_LIST_SIZE(field) \
228 if (buf->field ## _size % sizeof(*buf->field)) { \
229 av_log(ctx, AV_LOG_ERROR, "Invalid size for " #field ": %d, " \
230 "should be multiple of %d\n", \
231 buf->field ## _size, (int)sizeof(*buf->field)); \
232 return AVERROR(EINVAL); \
289 "Conflicting all_channel_counts and list in options\n");
309 #define OFFSET(x) offsetof(BufferSinkContext, x)
310 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
316 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
339 .
name =
"buffersink",
340 .description =
NULL_IF_CONFIG_SMALL(
"Buffer video frames, and make them available to the end of the filter graph."),
342 .priv_class = &buffersink_class,
359 .
name =
"abuffersink",
360 .description =
NULL_IF_CONFIG_SMALL(
"Buffer audio frames, and make them available to the end of the filter graph."),
361 .priv_class = &abuffersink_class,
static enum AVSampleFormat sample_fmts[]
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static const char *const format[]
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Main libavfilter public API header.
int ff_filter_graph_run_once(AVFilterGraph *graph)
Run one round of processing on a filter graph.
static av_cold int init(AVCodecContext *avctx)
static const AVFilterPad avfilter_vsink_buffer_inputs[]
static const AVOption abuffersink_options[]
static const AVFilterPad avfilter_asink_abuffer_inputs[]
static const AVOption buffersink_options[]
#define MAKE_AVFILTERLINK_ACCESSOR(type, field)
static av_cold int common_init(AVFilterContext *ctx)
static int vsink_query_formats(AVFilterContext *ctx)
static int get_frame_internal(AVFilterContext *ctx, AVFrame *frame, int flags, int samples)
static int return_or_keep_frame(BufferSinkContext *buf, AVFrame *out, AVFrame *in, int flags)
AVFILTER_DEFINE_CLASS(buffersink)
AVFilter ff_asink_abuffer
static int activate(AVFilterContext *ctx)
static int asink_query_formats(AVFilterContext *ctx)
static void cleanup_redundant_layouts(AVFilterContext *ctx)
#define CHECK_LIST_SIZE(field)
memory buffer sink API for audio and video
#define flags(name, subs,...)
audio channel layout utility functions
common internal and external API header
static const uint16_t channel_layouts[7]
static const uint8_t channel_counts[7]
static size_t ff_framequeue_queued_frames(const FFFrameQueue *fq)
Get the number of queued frames.
@ AV_OPT_TYPE_BINARY
offset must point to a pointer immediately followed by an int for the length
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
void av_buffersink_set_frame_size(AVFilterContext *ctx, unsigned frame_size)
Set the frame size for an audio buffer sink.
int attribute_align_arg av_buffersink_get_frame(AVFilterContext *ctx, AVFrame *frame)
Get a frame with filtered data from sink and put it in frame.
int attribute_align_arg av_buffersink_get_frame_flags(AVFilterContext *ctx, AVFrame *frame, int flags)
Get a frame with filtered data from sink and put it in frame.
#define AV_BUFFERSINK_FLAG_PEEK
Tell av_buffersink_get_buffer_ref() to read video/samples buffer reference, but not remove it from th...
int attribute_align_arg av_buffersink_get_samples(AVFilterContext *ctx, AVFrame *frame, int nb_samples)
Same as av_buffersink_get_frame(), but with the ability to specify the number of samples read.
AVABufferSinkParams * av_abuffersink_params_alloc(void)
Create an AVABufferSinkParams structure.
#define AV_BUFFERSINK_FLAG_NO_REQUEST
Tell av_buffersink_get_buffer_ref() not to request a frame from its input.
AVBufferSinkParams * av_buffersink_params_alloc(void)
Create an AVBufferSinkParams structure.
void av_frame_move_ref(AVFrame *dst, AVFrame *src)
Move everything contained in src to dst and reset src.
int av_frame_ref(AVFrame *dst, const AVFrame *src)
Set up a new reference to the data described by the source frame.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define AV_LOG_WARNING
Something somehow does not look correct.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
static void * av_x_if_null(const void *p, const void *x)
Return x default pointer in case p is NULL.
AVSampleFormat
Audio sample formats.
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
#define attribute_align_arg
enum MovChannelLayoutTag * layouts
AVPixelFormat
Pixel format.
Deprecated and unused struct to use for initializing an abuffersink context.
A reference to a data buffer.
Deprecated and unused struct to use for initializing a buffersink context.
enum AVPixelFormat * pixel_fmts
list of allowed pixel formats, terminated by AV_PIX_FMT_NONE
Describe the class of an AVClass context structure.
A list of supported channel layouts.
A link between two filters.
int max_samples
Maximum number of samples to filter at once.
int partial_buf_size
Size of the partial buffer to allocate.
int frame_wanted_out
True if a frame is currently wanted on the output of this filter.
int min_samples
Minimum number of samples to filter at once.
A filter pad used for either input or output.
const char * name
Pad name.
const char * name
Filter name.
This structure describes decoded (raw) audio or video data.
Rational number (pair of numerator and denominator).
int64_t * channel_layouts
list of accepted channel layouts, terminated by -1
enum AVPixelFormat * pixel_fmts
list of accepted pixel formats, must be terminated with -1
enum AVSampleFormat * sample_fmts
list of accepted sample formats, terminated by AV_SAMPLE_FMT_NONE
int * channel_counts
list of accepted channel counts, terminated by -1
int * sample_rates
list of accepted sample rates, terminated by -1