48 #define OFFSET(x) offsetof(AudioEchoContext, x) 49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 66 for (p = item_str; *p; p++) {
73 static void fill_items(
char *item_str,
int *nb_items,
float *items)
75 char *p, *saveptr =
NULL;
76 int i, new_nb_items = 0;
79 for (i = 0; i < *nb_items; i++) {
83 new_nb_items +=
av_sscanf(tstr,
"%f", &items[new_nb_items]) == 1;
86 *nb_items = new_nb_items;
105 int nb_delays, nb_decays,
i;
123 if (nb_delays != nb_decays) {
124 av_log(ctx,
AV_LOG_ERROR,
"Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
138 for (i = 0; i < nb_delays; i++) {
139 if (s->
delay[i] <= 0 || s->
delay[i] > 90000) {
186 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) 188 #define ECHO(name, type, min, max) \ 189 static void echo_samples_## name ##p(AudioEchoContext *ctx, \ 190 uint8_t **delayptrs, \ 191 uint8_t * const *src, uint8_t **dst, \ 192 int nb_samples, int channels) \ 194 const double out_gain = ctx->out_gain; \ 195 const double in_gain = ctx->in_gain; \ 196 const int nb_echoes = ctx->nb_echoes; \ 197 const int max_samples = ctx->max_samples; \ 198 int i, j, chan, av_uninit(index); \ 200 av_assert1(channels > 0); \ 202 for (chan = 0; chan < channels; chan++) { \ 203 const type *s = (type *)src[chan]; \ 204 type *d = (type *)dst[chan]; \ 205 type *dbuf = (type *)delayptrs[chan]; \ 207 index = ctx->delay_index; \ 208 for (i = 0; i < nb_samples; i++, s++, d++) { \ 212 out = in * in_gain; \ 213 for (j = 0; j < nb_echoes; j++) { \ 214 int ix = index + max_samples - ctx->samples[j]; \ 215 ix = MOD(ix, max_samples); \ 216 out += dbuf[ix] * ctx->decay[j]; \ 220 *d = av_clipd(out, min, max); \ 223 index = MOD(index + 1, max_samples); \ 226 ctx->delay_index = index; \ 229 ECHO(dbl,
double, -1.0, 1.0 )
230 ECHO(flt,
float, -1.0, 1.0 )
231 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
255 "out_gain %f can cause saturation of output\n", s->
out_gain);
257 switch (outlink->format) {
297 if (frame != out_frame)
384 .priv_class = &aecho_class,
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
This structure describes decoded (raw) audio or video data.
#define av_realloc_f(p, o, n)
#define AV_LOG_WARNING
Something somehow does not look correct.
Main libavfilter public API header.
AVFILTER_DEFINE_CLASS(aecho)
static void count_items(char *item_str, int *nb_items)
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a data pointers array, samples buffer for nb_samples samples, and fill data pointers and lin...
static void fill_items(char *item_str, int *nb_items, float *items)
const char * name
Pad name.
static av_cold void uninit(AVFilterContext *ctx)
void(* echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs, uint8_t *const *src, uint8_t **dst, int nb_samples, int channels)
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static const AVOption aecho_options[]
#define ECHO(name, type, min, max)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static int request_frame(AVFilterLink *outlink)
#define AVERROR_EOF
End of file.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
A filter pad used for either input or output.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
A link between two filters.
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
#define i(width, name, range_min, range_max)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int query_formats(AVFilterContext *ctx)
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void * priv
private data for use by the filter
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static int config_output(AVFilterLink *outlink)
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
simple assert() macros that are a bit more flexible than ISO C assert().
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
AVFilterContext * src
source filter
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
A list of supported channel layouts.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
AVSampleFormat
Audio sample formats.
typedef void(RENAME(mix_any_func_type))
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
static const AVFilterPad aecho_outputs[]
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Rational number (pair of numerator and denominator).
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
static av_cold int init(AVFilterContext *ctx)
static int activate(AVFilterContext *ctx)
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok()...
int channels
Number of channels.
AVFilterContext * dst
dest filter
static const AVFilterPad aecho_inputs[]
static enum AVSampleFormat sample_fmts[]
uint8_t ** extended_data
pointers to the data planes/channels.
int nb_samples
number of audio samples (per channel) described by this frame
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
#define AV_NOPTS_VALUE
Undefined timestamp value.