19 #include <rubberband/rubberband-c.h>
44 #define OFFSET(x) offsetof(RubberBandContext, x)
45 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
46 #define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
51 {
"transients",
"set transients",
OFFSET(transients),
AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX,
A,
"transients" },
52 {
"crisp", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsCrisp}, 0, 0,
A,
"transients" },
53 {
"mixed", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsMixed}, 0, 0,
A,
"transients" },
54 {
"smooth", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsSmooth}, 0, 0,
A,
"transients" },
56 {
"compound", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionDetectorCompound}, 0, 0,
A,
"detector" },
57 {
"percussive", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionDetectorPercussive}, 0, 0,
A,
"detector" },
58 {
"soft", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionDetectorSoft}, 0, 0,
A,
"detector" },
60 {
"laminar", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPhaseLaminar}, 0, 0,
A,
"phase" },
61 {
"independent", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPhaseIndependent}, 0, 0,
A,
"phase" },
63 {
"standard", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionWindowStandard}, 0, 0,
A,
"window" },
64 {
"short", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionWindowShort}, 0, 0,
A,
"window" },
65 {
"long", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionWindowLong}, 0, 0,
A,
"window" },
66 {
"smoothing",
"set smoothing",
OFFSET(smoothing),
AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX,
A,
"smoothing" },
67 {
"off", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionSmoothingOff}, 0, 0,
A,
"smoothing" },
68 {
"on", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionSmoothingOn}, 0, 0,
A,
"smoothing" },
70 {
"shifted", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionFormantShifted}, 0, 0,
A,
"formant" },
71 {
"preserved", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionFormantPreserved}, 0, 0,
A,
"formant" },
73 {
"quality", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPitchHighQuality}, 0, 0,
A,
"pitch" },
74 {
"speed", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPitchHighSpeed}, 0, 0,
A,
"pitch" },
75 {
"consistency", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPitchHighConsistency}, 0, 0,
A,
"pitch" },
77 {
"apart", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionChannelsApart}, 0, 0,
A,
"channels" },
78 {
"together", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionChannelsTogether}, 0, 0,
A,
"channels" },
89 rubberband_delete(
s->rbs);
128 int ret = 0, nb_samples;
131 s->first_pts =
in->pts;
134 s->nb_samples_in +=
in->nb_samples;
136 nb_samples = rubberband_available(
s->rbs);
137 if (nb_samples > 0) {
146 nb_samples = rubberband_retrieve(
s->rbs, (
float *
const *)
out->data, nb_samples);
147 out->nb_samples = nb_samples;
149 s->nb_samples_out += nb_samples;
155 return ret < 0 ? ret : nb_samples;
162 int opts =
s->transients|
s->detector|
s->phase|
s->window|
163 s->smoothing|
s->formant|
s->opitch|
s->channels|
164 RubberBandOptionProcessRealTime;
167 rubberband_delete(
s->rbs);
172 s->nb_samples = rubberband_get_samples_required(
s->rbs);
204 char *res,
int res_len,
int flags)
213 rubberband_set_time_ratio(
s->rbs, 1. /
s->tempo);
214 rubberband_set_pitch_scale(
s->rbs,
s->pitch);
215 s->nb_samples = rubberband_get_samples_required(
s->rbs);
238 .
name =
"rubberband",
242 .priv_class = &rubberband_class,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static const AVOption rubberband_options[]
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
AVFilter ff_af_rubberband
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static const AVFilterPad rubberband_inputs[]
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static int activate(AVFilterContext *ctx)
static av_cold void uninit(AVFilterContext *ctx)
AVFILTER_DEFINE_CLASS(rubberband)
static const AVFilterPad rubberband_outputs[]
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
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
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
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.
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
int ff_inlink_queued_samples(AVFilterLink *link)
Main libavfilter public API header.
#define flags(name, subs,...)
audio channel layout utility functions
common internal and external API header
static SDL_Window * window
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
#define FFERROR_NOT_READY
Filters implementation helper functions.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
#define AV_NOPTS_VALUE
Undefined timestamp value.
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
Describe the class of an AVClass context structure.
A list of supported channel layouts.
A link between two filters.
int channels
Number of channels.
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
int sample_rate
samples per second
AVFilterContext * dst
dest filter
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).