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 
   48     { 
"transients", 
"set transients", 
OFFSET(transients), 
AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, 
A, 
"transients" },
 
   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" },
 
   63     { 
"smoothing",  
"set smoothing",  
OFFSET(smoothing),  
AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, 
A, 
"smoothing" },
 
   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. 
static enum AVSampleFormat formats[]
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 
static int query_formats(AVFilterContext *ctx)
AVFilterContext * src
source filter 
int partial_buf_size
Size of the partial buffer to allocate. 
static const AVFilterPad outputs[]
A list of supported channel layouts. 
static const AVFilterPad inputs[]
AVSampleFormat
Audio sample formats. 
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Describe the class of an AVClass context structure. 
rational number numerator/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