19 #include <rubberband/rubberband-c.h> 42 #define OFFSET(x) offsetof(RubberBandContext, x) 43 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 49 {
"crisp", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsCrisp}, 0, 0,
A,
"transients" },
50 {
"mixed", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsMixed}, 0, 0,
A,
"transients" },
51 {
"smooth", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsSmooth}, 0, 0,
A,
"transients" },
53 {
"compound", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionDetectorCompound}, 0, 0,
A,
"detector" },
54 {
"percussive", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionDetectorPercussive}, 0, 0,
A,
"detector" },
55 {
"soft", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionDetectorSoft}, 0, 0,
A,
"detector" },
57 {
"laminar", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPhaseLaminar}, 0, 0,
A,
"phase" },
58 {
"independent", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPhaseIndependent}, 0, 0,
A,
"phase" },
60 {
"standard", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionWindowStandard}, 0, 0,
A,
"window" },
61 {
"short", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionWindowShort}, 0, 0,
A,
"window" },
62 {
"long", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionWindowLong}, 0, 0,
A,
"window" },
64 {
"off", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionSmoothingOff}, 0, 0,
A,
"smoothing" },
65 {
"on", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionSmoothingOn}, 0, 0,
A,
"smoothing" },
67 {
"shifted", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionFormantShifted}, 0, 0,
A,
"formant" },
68 {
"preserved", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionFormantPreserved}, 0, 0,
A,
"formant" },
70 {
"quality", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPitchHighQuality}, 0, 0,
A,
"pitch" },
71 {
"speed", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPitchHighSpeed}, 0, 0,
A,
"pitch" },
72 {
"consistency", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionPitchHighConsistency}, 0, 0,
A,
"pitch" },
74 {
"apart", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionChannelsApart}, 0, 0,
A,
"channels" },
75 {
"together", 0, 0,
AV_OPT_TYPE_CONST, {.i64=RubberBandOptionChannelsTogether}, 0, 0,
A,
"channels" },
86 rubberband_delete(s->
rbs);
124 int ret = 0, nb_samples;
129 nb_samples = rubberband_available(s->
rbs);
130 if (nb_samples > 0) {
139 nb_samples = rubberband_retrieve(s->
rbs, (
float *
const *)out->
data, nb_samples);
155 RubberBandOptionProcessRealTime;
158 rubberband_delete(s->
rbs);
178 if (rubberband_available(s->
rbs) > 0) {
185 rubberband_process(s->
rbs, (
const float *
const *)out->
data, 1, 1);
187 nb_samples = rubberband_available(s->
rbs);
189 if (nb_samples > 0) {
196 nb_samples = rubberband_retrieve(s->
rbs, (
float *
const *)out->
data, nb_samples);
211 char *res,
int res_len,
int flags)
215 if (!strcmp(cmd,
"tempo")) {
218 sscanf(args,
"%lf", &arg);
219 if (arg < 0.01 || arg > 100) {
221 "Tempo scale factor '%f' out of range\n", arg);
224 rubberband_set_time_ratio(s->
rbs, 1. / arg);
227 if (!strcmp(cmd,
"pitch")) {
230 sscanf(args,
"%lf", &arg);
231 if (arg < 0.01 || arg > 100) {
233 "Pitch scale factor '%f' out of range\n", arg);
236 rubberband_set_pitch_scale(s->
rbs, arg);
262 .
name =
"rubberband",
266 .priv_class = &rubberband_class,
268 .
inputs = rubberband_inputs,
static const AVFilterPad rubberband_outputs[]
This structure describes decoded (raw) audio or video data.
static int config_input(AVFilterLink *inlink)
Main libavfilter public API header.
int max_samples
Maximum number of samples to filter at once.
const char * name
Pad name.
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 int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
#define AVERROR_EOF
End of file.
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.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int min_samples
Minimum number of samples to filter at once.
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.
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
static const AVOption rubberband_options[]
static const AVFilterPad rubberband_inputs[]
audio channel layout utility functions
const char AVS_Value args
static int query_formats(AVFilterContext *ctx)
AVFilterContext * src
source filter
int partial_buf_size
Size of the partial buffer to allocate.
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
A list of supported channel layouts.
AVSampleFormat
Audio sample formats.
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).
static av_cold void uninit(AVFilterContext *ctx)
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
AVFilter ff_af_rubberband
common internal and external API header
int channels
Number of channels.
AVFilterContext * dst
dest filter
AVFILTER_DEFINE_CLASS(rubberband)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static enum AVSampleFormat sample_fmts[]
static int request_frame(AVFilterLink *outlink)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
int nb_samples
number of audio samples (per channel) described by this frame