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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavfilter/af_atempo.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

1187 lines
38 KiB
C

/*
* Copyright (c) 2012 Pavel Koshevoy <pkoshevoy at gmail dot com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* tempo scaling audio filter -- an implementation of WSOLA algorithm
*
* Based on MIT licensed yaeAudioTempoFilter.h and yaeAudioFragment.h
* from Apprentice Video player by Pavel Koshevoy.
* https://sourceforge.net/projects/apprenticevideo/
*
* An explanation of SOLA algorithm is available at
* http://www.surina.net/article/time-and-pitch-scaling.html
*
* WSOLA is very similar to SOLA, only one major difference exists between
* these algorithms. SOLA shifts audio fragments along the output stream,
* where as WSOLA shifts audio fragments along the input stream.
*
* The advantage of WSOLA algorithm is that the overlap region size is
* always the same, therefore the blending function is constant and
* can be precomputed.
*/
#include <float.h>
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavutil/tx.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"
/**
* A fragment of audio waveform
*/
typedef struct AudioFragment {
// index of the first sample of this fragment in the overall waveform;
// 0: input sample position
// 1: output sample position
int64_t position[2];
// original packed multi-channel samples:
uint8_t *data;
// number of samples in this fragment:
int nsamples;
// rDFT transform of the down-mixed mono fragment, used for
// fast waveform alignment via correlation in frequency domain:
float *xdat_in;
float *xdat;
} AudioFragment;
/**
* Filter state machine states
*/
typedef enum {
YAE_LOAD_FRAGMENT,
YAE_ADJUST_POSITION,
YAE_RELOAD_FRAGMENT,
YAE_OUTPUT_OVERLAP_ADD,
YAE_FLUSH_OUTPUT,
} FilterState;
/**
* Filter state machine
*/
typedef struct ATempoContext {
const AVClass *class;
// ring-buffer of input samples, necessary because some times
// input fragment position may be adjusted backwards:
uint8_t *buffer;
// ring-buffer maximum capacity, expressed in sample rate time base:
int ring;
// ring-buffer house keeping:
int size;
int head;
int tail;
// 0: input sample position corresponding to the ring buffer tail
// 1: output sample position
int64_t position[2];
// first input timestamp, all other timestamps are offset by this one
int64_t start_pts;
// sample format:
enum AVSampleFormat format;
// number of channels:
int channels;
// row of bytes to skip from one sample to next, across multple channels;
// stride = (number-of-channels * bits-per-sample-per-channel) / 8
int stride;
// fragment window size, power-of-two integer:
int window;
// Hann window coefficients, for feathering
// (blending) the overlapping fragment region:
float *hann;
// tempo scaling factor:
double tempo;
// a snapshot of previous fragment input and output position values
// captured when the tempo scale factor was set most recently:
int64_t origin[2];
// current/previous fragment ring-buffer:
AudioFragment frag[2];
// current fragment index:
uint64_t nfrag;
// current state:
FilterState state;
// for fast correlation calculation in frequency domain:
AVTXContext *real_to_complex;
AVTXContext *complex_to_real;
av_tx_fn r2c_fn, c2r_fn;
float *correlation_in;
float *correlation;
// for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
AVFrame *dst_buffer;
uint8_t *dst;
uint8_t *dst_end;
uint64_t nsamples_in;
uint64_t nsamples_out;
} ATempoContext;
#define YAE_ATEMPO_MIN 0.5
#define YAE_ATEMPO_MAX 100.0
#define OFFSET(x) offsetof(ATempoContext, x)
static const AVOption atempo_options[] = {
{ "tempo", "set tempo scale factor",
OFFSET(tempo), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 },
YAE_ATEMPO_MIN,
YAE_ATEMPO_MAX,
AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM },
{ NULL }
};
AVFILTER_DEFINE_CLASS(atempo);
inline static AudioFragment *yae_curr_frag(ATempoContext *atempo)
{
return &atempo->frag[atempo->nfrag % 2];
}
inline static AudioFragment *yae_prev_frag(ATempoContext *atempo)
{
return &atempo->frag[(atempo->nfrag + 1) % 2];
}
/**
* Reset filter to initial state, do not deallocate existing local buffers.
*/
static void yae_clear(ATempoContext *atempo)
{
atempo->size = 0;
atempo->head = 0;
atempo->tail = 0;
atempo->nfrag = 0;
atempo->state = YAE_LOAD_FRAGMENT;
atempo->start_pts = AV_NOPTS_VALUE;
atempo->position[0] = 0;
atempo->position[1] = 0;
atempo->origin[0] = 0;
atempo->origin[1] = 0;
atempo->frag[0].position[0] = 0;
atempo->frag[0].position[1] = 0;
atempo->frag[0].nsamples = 0;
atempo->frag[1].position[0] = 0;
atempo->frag[1].position[1] = 0;
atempo->frag[1].nsamples = 0;
// shift left position of 1st fragment by half a window
// so that no re-normalization would be required for
// the left half of the 1st fragment:
atempo->frag[0].position[0] = -(int64_t)(atempo->window / 2);
atempo->frag[0].position[1] = -(int64_t)(atempo->window / 2);
av_frame_free(&atempo->dst_buffer);
atempo->dst = NULL;
atempo->dst_end = NULL;
atempo->nsamples_in = 0;
atempo->nsamples_out = 0;
}
/**
* Reset filter to initial state and deallocate all buffers.
*/
static void yae_release_buffers(ATempoContext *atempo)
{
yae_clear(atempo);
av_freep(&atempo->frag[0].data);
av_freep(&atempo->frag[1].data);
av_freep(&atempo->frag[0].xdat_in);
av_freep(&atempo->frag[1].xdat_in);
av_freep(&atempo->frag[0].xdat);
av_freep(&atempo->frag[1].xdat);
av_freep(&atempo->buffer);
av_freep(&atempo->hann);
av_freep(&atempo->correlation_in);
av_freep(&atempo->correlation);
av_tx_uninit(&atempo->real_to_complex);
av_tx_uninit(&atempo->complex_to_real);
}
/* av_realloc is not aligned enough; fortunately, the data does not need to
* be preserved */
#define RE_MALLOC_OR_FAIL(field, field_size, element_size) \
do { \
av_freep(&field); \
field = av_calloc(field_size, element_size); \
if (!field) { \
yae_release_buffers(atempo); \
return AVERROR(ENOMEM); \
} \
} while (0)
/**
* Prepare filter for processing audio data of given format,
* sample rate and number of channels.
*/
static int yae_reset(ATempoContext *atempo,
enum AVSampleFormat format,
int sample_rate,
int channels)
{
const int sample_size = av_get_bytes_per_sample(format);
uint32_t nlevels = 0;
float scale = 1.f, iscale = 1.f;
uint32_t pot;
int i;
atempo->format = format;
atempo->channels = channels;
atempo->stride = sample_size * channels;
// pick a segment window size:
atempo->window = sample_rate / 24;
// adjust window size to be a power-of-two integer:
nlevels = av_log2(atempo->window);
pot = 1 << nlevels;
av_assert0(pot <= atempo->window);
if (pot < atempo->window) {
atempo->window = pot * 2;
nlevels++;
}
// initialize audio fragment buffers:
RE_MALLOC_OR_FAIL(atempo->frag[0].data, atempo->window, atempo->stride);
RE_MALLOC_OR_FAIL(atempo->frag[1].data, atempo->window, atempo->stride);
RE_MALLOC_OR_FAIL(atempo->frag[0].xdat_in, (atempo->window + 1), sizeof(AVComplexFloat));
RE_MALLOC_OR_FAIL(atempo->frag[1].xdat_in, (atempo->window + 1), sizeof(AVComplexFloat));
RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, (atempo->window + 1), sizeof(AVComplexFloat));
RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, (atempo->window + 1), sizeof(AVComplexFloat));
// initialize rDFT contexts:
av_tx_uninit(&atempo->real_to_complex);
av_tx_uninit(&atempo->complex_to_real);
av_tx_init(&atempo->real_to_complex, &atempo->r2c_fn, AV_TX_FLOAT_RDFT, 0, 1 << (nlevels + 1), &scale, 0);
if (!atempo->real_to_complex) {
yae_release_buffers(atempo);
return AVERROR(ENOMEM);
}
av_tx_init(&atempo->complex_to_real, &atempo->c2r_fn, AV_TX_FLOAT_RDFT, 1, 1 << (nlevels + 1), &iscale, 0);
if (!atempo->complex_to_real) {
yae_release_buffers(atempo);
return AVERROR(ENOMEM);
}
RE_MALLOC_OR_FAIL(atempo->correlation_in, (atempo->window + 1), sizeof(AVComplexFloat));
RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window, sizeof(AVComplexFloat));
atempo->ring = atempo->window * 3;
RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring, atempo->stride);
// initialize the Hann window function:
RE_MALLOC_OR_FAIL(atempo->hann, atempo->window, sizeof(float));
for (i = 0; i < atempo->window; i++) {
double t = (double)i / (double)(atempo->window - 1);
double h = 0.5 * (1.0 - cos(2.0 * M_PI * t));
atempo->hann[i] = (float)h;
}
yae_clear(atempo);
return 0;
}
static int yae_update(AVFilterContext *ctx)
{
const AudioFragment *prev;
ATempoContext *atempo = ctx->priv;
prev = yae_prev_frag(atempo);
atempo->origin[0] = prev->position[0] + atempo->window / 2;
atempo->origin[1] = prev->position[1] + atempo->window / 2;
return 0;
}
/**
* A helper macro for initializing complex data buffer with scalar data
* of a given type.
*/
#define yae_init_xdat(scalar_type, scalar_max) \
do { \
const uint8_t *src_end = src + \
frag->nsamples * atempo->channels * sizeof(scalar_type); \
\
float *xdat = frag->xdat_in; \
scalar_type tmp; \
\
if (atempo->channels == 1) { \
for (; src < src_end; xdat++) { \
tmp = *(const scalar_type *)src; \
src += sizeof(scalar_type); \
\
*xdat = (float)tmp; \
} \
} else { \
float s, max, ti, si; \
int i; \
\
for (; src < src_end; xdat++) { \
tmp = *(const scalar_type *)src; \
src += sizeof(scalar_type); \
\
max = (float)tmp; \
s = FFMIN((float)scalar_max, \
(float)fabsf(max)); \
\
for (i = 1; i < atempo->channels; i++) { \
tmp = *(const scalar_type *)src; \
src += sizeof(scalar_type); \
\
ti = (float)tmp; \
si = FFMIN((float)scalar_max, \
(float)fabsf(ti)); \
\
if (s < si) { \
s = si; \
max = ti; \
} \
} \
\
*xdat = max; \
} \
} \
} while (0)
/**
* Initialize complex data buffer of a given audio fragment
* with down-mixed mono data of appropriate scalar type.
*/
static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
{
// shortcuts:
const uint8_t *src = frag->data;
// init complex data buffer used for FFT and Correlation:
memset(frag->xdat_in, 0, sizeof(AVComplexFloat) * (atempo->window + 1));
if (atempo->format == AV_SAMPLE_FMT_U8) {
yae_init_xdat(uint8_t, 127);
} else if (atempo->format == AV_SAMPLE_FMT_S16) {
yae_init_xdat(int16_t, 32767);
} else if (atempo->format == AV_SAMPLE_FMT_S32) {
yae_init_xdat(int, 2147483647);
} else if (atempo->format == AV_SAMPLE_FMT_FLT) {
yae_init_xdat(float, 1);
} else if (atempo->format == AV_SAMPLE_FMT_DBL) {
yae_init_xdat(double, 1);
}
}
/**
* Populate the internal data buffer on as-needed basis.
*
* @return
* 0 if requested data was already available or was successfully loaded,
* AVERROR(EAGAIN) if more input data is required.
*/
static int yae_load_data(ATempoContext *atempo,
const uint8_t **src_ref,
const uint8_t *src_end,
int64_t stop_here)
{
// shortcut:
const uint8_t *src = *src_ref;
const int read_size = stop_here - atempo->position[0];
if (stop_here <= atempo->position[0]) {
return 0;
}
// samples are not expected to be skipped, unless tempo is greater than 2:
av_assert0(read_size <= atempo->ring || atempo->tempo > 2.0);
while (atempo->position[0] < stop_here && src < src_end) {
int src_samples = (src_end - src) / atempo->stride;
// load data piece-wise, in order to avoid complicating the logic:
int nsamples = FFMIN(read_size, src_samples);
int na;
int nb;
nsamples = FFMIN(nsamples, atempo->ring);
na = FFMIN(nsamples, atempo->ring - atempo->tail);
nb = FFMIN(nsamples - na, atempo->ring);
if (na) {
uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;
memcpy(a, src, na * atempo->stride);
src += na * atempo->stride;
atempo->position[0] += na;
atempo->size = FFMIN(atempo->size + na, atempo->ring);
atempo->tail = (atempo->tail + na) % atempo->ring;
atempo->head =
atempo->size < atempo->ring ?
atempo->tail - atempo->size :
atempo->tail;
}
if (nb) {
uint8_t *b = atempo->buffer;
memcpy(b, src, nb * atempo->stride);
src += nb * atempo->stride;
atempo->position[0] += nb;
atempo->size = FFMIN(atempo->size + nb, atempo->ring);
atempo->tail = (atempo->tail + nb) % atempo->ring;
atempo->head =
atempo->size < atempo->ring ?
atempo->tail - atempo->size :
atempo->tail;
}
}
// pass back the updated source buffer pointer:
*src_ref = src;
// sanity check:
av_assert0(atempo->position[0] <= stop_here);
return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);
}
/**
* Populate current audio fragment data buffer.
*
* @return
* 0 when the fragment is ready,
* AVERROR(EAGAIN) if more input data is required.
*/
static int yae_load_frag(ATempoContext *atempo,
const uint8_t **src_ref,
const uint8_t *src_end)
{
// shortcuts:
AudioFragment *frag = yae_curr_frag(atempo);
uint8_t *dst;
int64_t missing, start, zeros;
uint32_t nsamples;
const uint8_t *a, *b;
int i0, i1, n0, n1, na, nb;
int64_t stop_here = frag->position[0] + atempo->window;
if (src_ref && yae_load_data(atempo, src_ref, src_end, stop_here) != 0) {
return AVERROR(EAGAIN);
}
// calculate the number of samples we don't have:
missing =
stop_here > atempo->position[0] ?
stop_here - atempo->position[0] : 0;
nsamples =
missing < (int64_t)atempo->window ?
(uint32_t)(atempo->window - missing) : 0;
// setup the output buffer:
frag->nsamples = nsamples;
dst = frag->data;
start = atempo->position[0] - atempo->size;
// what we don't have we substitute with zeros:
zeros =
frag->position[0] < start ?
FFMIN(start - frag->position[0], (int64_t)nsamples) : 0;
if (zeros == nsamples) {
return 0;
}
if (frag->position[0] < start) {
memset(dst, 0, zeros * atempo->stride);
dst += zeros * atempo->stride;
}
// get the remaining data from the ring buffer:
na = (atempo->head < atempo->tail ?
atempo->tail - atempo->head :
atempo->ring - atempo->head);
nb = atempo->head < atempo->tail ? 0 : atempo->tail;
// sanity check:
av_assert0(nsamples <= zeros + na + nb);
a = atempo->buffer + atempo->head * atempo->stride;
b = atempo->buffer;
i0 = frag->position[0] + zeros - start;
i1 = i0 < na ? 0 : i0 - na;
n0 = i0 < na ? FFMIN(na - i0, (int)(nsamples - zeros)) : 0;
n1 = nsamples - zeros - n0;
if (n0) {
memcpy(dst, a + i0 * atempo->stride, n0 * atempo->stride);
dst += n0 * atempo->stride;
}
if (n1) {
memcpy(dst, b + i1 * atempo->stride, n1 * atempo->stride);
}
return 0;
}
/**
* Prepare for loading next audio fragment.
*/
static void yae_advance_to_next_frag(ATempoContext *atempo)
{
const double fragment_step = atempo->tempo * (double)(atempo->window / 2);
const AudioFragment *prev;
AudioFragment *frag;
atempo->nfrag++;
prev = yae_prev_frag(atempo);
frag = yae_curr_frag(atempo);
frag->position[0] = prev->position[0] + (int64_t)fragment_step;
frag->position[1] = prev->position[1] + atempo->window / 2;
frag->nsamples = 0;
}
/**
* Calculate cross-correlation via rDFT.
*
* Multiply two vectors of complex numbers (result of real_to_complex rDFT)
* and transform back via complex_to_real rDFT.
*/
static void yae_xcorr_via_rdft(float *xcorr_in,
float *xcorr,
AVTXContext *complex_to_real,
av_tx_fn c2r_fn,
const AVComplexFloat *xa,
const AVComplexFloat *xb,
const int window)
{
AVComplexFloat *xc = (AVComplexFloat *)xcorr_in;
int i;
for (i = 0; i <= window; i++, xa++, xb++, xc++) {
xc->re = (xa->re * xb->re + xa->im * xb->im);
xc->im = (xa->im * xb->re - xa->re * xb->im);
}
// apply inverse rDFT:
c2r_fn(complex_to_real, xcorr, xcorr_in, sizeof(*xc));
}
/**
* Calculate alignment offset for given fragment
* relative to the previous fragment.
*
* @return alignment offset of current fragment relative to previous.
*/
static int yae_align(AudioFragment *frag,
const AudioFragment *prev,
const int window,
const int delta_max,
const int drift,
float *correlation_in,
float *correlation,
AVTXContext *complex_to_real,
av_tx_fn c2r_fn)
{
int best_offset = -drift;
float best_metric = -FLT_MAX;
float *xcorr;
int i0;
int i1;
int i;
yae_xcorr_via_rdft(correlation_in,
correlation,
complex_to_real,
c2r_fn,
(const AVComplexFloat *)prev->xdat,
(const AVComplexFloat *)frag->xdat,
window);
// identify search window boundaries:
i0 = FFMAX(window / 2 - delta_max - drift, 0);
i0 = FFMIN(i0, window);
i1 = FFMIN(window / 2 + delta_max - drift, window - window / 16);
i1 = FFMAX(i1, 0);
// identify cross-correlation peaks within search window:
xcorr = correlation + i0;
for (i = i0; i < i1; i++, xcorr++) {
float metric = *xcorr;
// normalize:
float drifti = (float)(drift + i);
metric *= drifti * (float)(i - i0) * (float)(i1 - i);
if (metric > best_metric) {
best_metric = metric;
best_offset = i - window / 2;
}
}
return best_offset;
}
/**
* Adjust current fragment position for better alignment
* with previous fragment.
*
* @return alignment correction.
*/
static int yae_adjust_position(ATempoContext *atempo)
{
const AudioFragment *prev = yae_prev_frag(atempo);
AudioFragment *frag = yae_curr_frag(atempo);
const double prev_output_position =
(double)(prev->position[1] - atempo->origin[1] + atempo->window / 2) *
atempo->tempo;
const double ideal_output_position =
(double)(prev->position[0] - atempo->origin[0] + atempo->window / 2);
const int drift = (int)(prev_output_position - ideal_output_position);
const int delta_max = atempo->window / 2;
const int correction = yae_align(frag,
prev,
atempo->window,
delta_max,
drift,
atempo->correlation_in,
atempo->correlation,
atempo->complex_to_real,
atempo->c2r_fn);
if (correction) {
// adjust fragment position:
frag->position[0] -= correction;
// clear so that the fragment can be reloaded:
frag->nsamples = 0;
}
return correction;
}
/**
* A helper macro for blending the overlap region of previous
* and current audio fragment.
*/
#define yae_blend(scalar_type) \
do { \
const scalar_type *aaa = (const scalar_type *)a; \
const scalar_type *bbb = (const scalar_type *)b; \
\
scalar_type *out = (scalar_type *)dst; \
scalar_type *out_end = (scalar_type *)dst_end; \
int64_t i; \
\
for (i = 0; i < overlap && out < out_end; \
i++, atempo->position[1]++, wa++, wb++) { \
float w0 = *wa; \
float w1 = *wb; \
int j; \
\
for (j = 0; j < atempo->channels; \
j++, aaa++, bbb++, out++) { \
float t0 = (float)*aaa; \
float t1 = (float)*bbb; \
\
*out = \
frag->position[0] + i < 0 ? \
*aaa : \
(scalar_type)(t0 * w0 + t1 * w1); \
} \
} \
dst = (uint8_t *)out; \
} while (0)
/**
* Blend the overlap region of previous and current audio fragment
* and output the results to the given destination buffer.
*
* @return
* 0 if the overlap region was completely stored in the dst buffer,
* AVERROR(EAGAIN) if more destination buffer space is required.
*/
static int yae_overlap_add(ATempoContext *atempo,
uint8_t **dst_ref,
uint8_t *dst_end)
{
// shortcuts:
const AudioFragment *prev = yae_prev_frag(atempo);
const AudioFragment *frag = yae_curr_frag(atempo);
const int64_t start_here = FFMAX(atempo->position[1],
frag->position[1]);
const int64_t stop_here = FFMIN(prev->position[1] + prev->nsamples,
frag->position[1] + frag->nsamples);
const int64_t overlap = stop_here - start_here;
const int64_t ia = start_here - prev->position[1];
const int64_t ib = start_here - frag->position[1];
const float *wa = atempo->hann + ia;
const float *wb = atempo->hann + ib;
const uint8_t *a = prev->data + ia * atempo->stride;
const uint8_t *b = frag->data + ib * atempo->stride;
uint8_t *dst = *dst_ref;
av_assert0(start_here <= stop_here &&
frag->position[1] <= start_here &&
overlap <= frag->nsamples);
if (atempo->format == AV_SAMPLE_FMT_U8) {
yae_blend(uint8_t);
} else if (atempo->format == AV_SAMPLE_FMT_S16) {
yae_blend(int16_t);
} else if (atempo->format == AV_SAMPLE_FMT_S32) {
yae_blend(int);
} else if (atempo->format == AV_SAMPLE_FMT_FLT) {
yae_blend(float);
} else if (atempo->format == AV_SAMPLE_FMT_DBL) {
yae_blend(double);
}
// pass-back the updated destination buffer pointer:
*dst_ref = dst;
return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
}
/**
* Feed as much data to the filter as it is able to consume
* and receive as much processed data in the destination buffer
* as it is able to produce or store.
*/
static void
yae_apply(ATempoContext *atempo,
const uint8_t **src_ref,
const uint8_t *src_end,
uint8_t **dst_ref,
uint8_t *dst_end)
{
while (1) {
if (atempo->state == YAE_LOAD_FRAGMENT) {
// load additional data for the current fragment:
if (yae_load_frag(atempo, src_ref, src_end) != 0) {
break;
}
// down-mix to mono:
yae_downmix(atempo, yae_curr_frag(atempo));
// apply rDFT:
atempo->r2c_fn(atempo->real_to_complex, yae_curr_frag(atempo)->xdat, yae_curr_frag(atempo)->xdat_in, sizeof(float));
// must load the second fragment before alignment can start:
if (!atempo->nfrag) {
yae_advance_to_next_frag(atempo);
continue;
}
atempo->state = YAE_ADJUST_POSITION;
}
if (atempo->state == YAE_ADJUST_POSITION) {
// adjust position for better alignment:
if (yae_adjust_position(atempo)) {
// reload the fragment at the corrected position, so that the
// Hann window blending would not require normalization:
atempo->state = YAE_RELOAD_FRAGMENT;
} else {
atempo->state = YAE_OUTPUT_OVERLAP_ADD;
}
}
if (atempo->state == YAE_RELOAD_FRAGMENT) {
// load additional data if necessary due to position adjustment:
if (yae_load_frag(atempo, src_ref, src_end) != 0) {
break;
}
// down-mix to mono:
yae_downmix(atempo, yae_curr_frag(atempo));
// apply rDFT:
atempo->r2c_fn(atempo->real_to_complex, yae_curr_frag(atempo)->xdat, yae_curr_frag(atempo)->xdat_in, sizeof(float));
atempo->state = YAE_OUTPUT_OVERLAP_ADD;
}
if (atempo->state == YAE_OUTPUT_OVERLAP_ADD) {
// overlap-add and output the result:
if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
break;
}
// advance to the next fragment, repeat:
yae_advance_to_next_frag(atempo);
atempo->state = YAE_LOAD_FRAGMENT;
}
}
}
/**
* Flush any buffered data from the filter.
*
* @return
* 0 if all data was completely stored in the dst buffer,
* AVERROR(EAGAIN) if more destination buffer space is required.
*/
static int yae_flush(ATempoContext *atempo,
uint8_t **dst_ref,
uint8_t *dst_end)
{
AudioFragment *frag = yae_curr_frag(atempo);
int64_t overlap_end;
int64_t start_here;
int64_t stop_here;
int64_t offset;
const uint8_t *src;
uint8_t *dst;
int src_size;
int dst_size;
int nbytes;
atempo->state = YAE_FLUSH_OUTPUT;
if (!atempo->nfrag) {
// there is nothing to flush:
return 0;
}
if (atempo->position[0] == frag->position[0] + frag->nsamples &&
atempo->position[1] == frag->position[1] + frag->nsamples) {
// the current fragment is already flushed:
return 0;
}
if (frag->position[0] + frag->nsamples < atempo->position[0]) {
// finish loading the current (possibly partial) fragment:
yae_load_frag(atempo, NULL, NULL);
if (atempo->nfrag) {
// down-mix to mono:
yae_downmix(atempo, frag);
// apply rDFT:
atempo->r2c_fn(atempo->real_to_complex, frag->xdat, frag->xdat_in, sizeof(float));
// align current fragment to previous fragment:
if (yae_adjust_position(atempo)) {
// reload the current fragment due to adjusted position:
yae_load_frag(atempo, NULL, NULL);
}
}
}
// flush the overlap region:
overlap_end = frag->position[1] + FFMIN(atempo->window / 2,
frag->nsamples);
while (atempo->position[1] < overlap_end) {
if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
return AVERROR(EAGAIN);
}
}
// check whether all of the input samples have been consumed:
if (frag->position[0] + frag->nsamples < atempo->position[0]) {
yae_advance_to_next_frag(atempo);
return AVERROR(EAGAIN);
}
// flush the remainder of the current fragment:
start_here = FFMAX(atempo->position[1], overlap_end);
stop_here = frag->position[1] + frag->nsamples;
offset = start_here - frag->position[1];
av_assert0(start_here <= stop_here && frag->position[1] <= start_here);
src = frag->data + offset * atempo->stride;
dst = (uint8_t *)*dst_ref;
src_size = (int)(stop_here - start_here) * atempo->stride;
dst_size = dst_end - dst;
nbytes = FFMIN(src_size, dst_size);
memcpy(dst, src, nbytes);
dst += nbytes;
atempo->position[1] += (nbytes / atempo->stride);
// pass-back the updated destination buffer pointer:
*dst_ref = (uint8_t *)dst;
return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
}
static av_cold int init(AVFilterContext *ctx)
{
ATempoContext *atempo = ctx->priv;
atempo->format = AV_SAMPLE_FMT_NONE;
atempo->state = YAE_LOAD_FRAGMENT;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ATempoContext *atempo = ctx->priv;
yae_release_buffers(atempo);
}
// WSOLA necessitates an internal sliding window ring buffer
// for incoming audio stream.
//
// Planar sample formats are too cumbersome to store in a ring buffer,
// therefore planar sample formats are not supported.
//
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
static int config_props(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ATempoContext *atempo = ctx->priv;
enum AVSampleFormat format = inlink->format;
int sample_rate = (int)inlink->sample_rate;
return yae_reset(atempo, format, sample_rate, inlink->ch_layout.nb_channels);
}
static int push_samples(ATempoContext *atempo,
AVFilterLink *outlink,
int n_out)
{
int ret;
atempo->dst_buffer->sample_rate = outlink->sample_rate;
atempo->dst_buffer->nb_samples = n_out;
// adjust the PTS:
atempo->dst_buffer->pts = atempo->start_pts +
av_rescale_q(atempo->nsamples_out,
(AVRational){ 1, outlink->sample_rate },
outlink->time_base);
ret = ff_filter_frame(outlink, atempo->dst_buffer);
atempo->dst_buffer = NULL;
atempo->dst = NULL;
atempo->dst_end = NULL;
if (ret < 0)
return ret;
atempo->nsamples_out += n_out;
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
{
AVFilterContext *ctx = inlink->dst;
ATempoContext *atempo = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret = 0;
int n_in = src_buffer->nb_samples;
int n_out = (int)(0.5 + ((double)n_in) / atempo->tempo);
const uint8_t *src = src_buffer->data[0];
const uint8_t *src_end = src + n_in * atempo->stride;
if (atempo->start_pts == AV_NOPTS_VALUE)
atempo->start_pts = av_rescale_q(src_buffer->pts,
inlink->time_base,
outlink->time_base);
while (src < src_end) {
if (!atempo->dst_buffer) {
atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);
if (!atempo->dst_buffer) {
av_frame_free(&src_buffer);
return AVERROR(ENOMEM);
}
av_frame_copy_props(atempo->dst_buffer, src_buffer);
atempo->dst = atempo->dst_buffer->data[0];
atempo->dst_end = atempo->dst + n_out * atempo->stride;
}
yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);
if (atempo->dst == atempo->dst_end) {
int n_samples = ((atempo->dst - atempo->dst_buffer->data[0]) /
atempo->stride);
ret = push_samples(atempo, outlink, n_samples);
if (ret < 0)
goto end;
}
}
atempo->nsamples_in += n_in;
end:
av_frame_free(&src_buffer);
return ret;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ATempoContext *atempo = ctx->priv;
int ret;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF) {
// flush the filter:
int n_max = atempo->ring;
int n_out;
int err = AVERROR(EAGAIN);
while (err == AVERROR(EAGAIN)) {
if (!atempo->dst_buffer) {
atempo->dst_buffer = ff_get_audio_buffer(outlink, n_max);
if (!atempo->dst_buffer)
return AVERROR(ENOMEM);
atempo->dst = atempo->dst_buffer->data[0];
atempo->dst_end = atempo->dst + n_max * atempo->stride;
}
err = yae_flush(atempo, &atempo->dst, atempo->dst_end);
n_out = ((atempo->dst - atempo->dst_buffer->data[0]) /
atempo->stride);
if (n_out) {
ret = push_samples(atempo, outlink, n_out);
if (ret < 0)
return ret;
}
}
av_frame_free(&atempo->dst_buffer);
atempo->dst = NULL;
atempo->dst_end = NULL;
return AVERROR_EOF;
}
return ret;
}
static int process_command(AVFilterContext *ctx,
const char *cmd,
const char *arg,
char *res,
int res_len,
int flags)
{
int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
if (ret < 0)
return ret;
return yae_update(ctx);
}
static const AVFilterPad atempo_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_props,
},
};
static const AVFilterPad atempo_outputs[] = {
{
.name = "default",
.request_frame = request_frame,
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_atempo = {
.name = "atempo",
.description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
.init = init,
.uninit = uninit,
.process_command = process_command,
.priv_size = sizeof(ATempoContext),
.priv_class = &atempo_class,
FILTER_INPUTS(atempo_inputs),
FILTER_OUTPUTS(atempo_outputs),
FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
};