[FFmpeg-devel] [PATCH] libavfilter: add atempo filter (revised patch v5)
Pavel Koshevoy
pkoshevoy at gmail.com
Tue Jun 12 05:18:02 CEST 2012
Add atempo audio filter for adjusting audio tempo without affecting
pitch. This filter implements WSOLA algorithm with fast cross
correlation calculation in frequency domain.
Signed-off-by: Pavel Koshevoy <pavel at homestead.aragog.com>
---
Changelog | 1 +
MAINTAINERS | 1 +
configure | 1 +
doc/filters.texi | 18 +
libavfilter/Makefile | 2 +
libavfilter/af_atempo.c | 1134 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
7 files changed, 1158 insertions(+), 0 deletions(-)
create mode 100644 libavfilter/af_atempo.c
diff --git a/Changelog b/Changelog
index 41b0bdc..a639c71 100644
--- a/Changelog
+++ b/Changelog
@@ -5,6 +5,7 @@ version next:
- INI and flat output in ffprobe
- Scene detection in libavfilter
- Indeo Audio decoder
+- atempo filter
version 0.11:
diff --git a/MAINTAINERS b/MAINTAINERS
index aa1b5ed..3c5e9c5 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -275,6 +275,7 @@ Video filters:
graphdump.c Nicolas George
af_amerge.c Nicolas George
af_astreamsync.c Nicolas George
+ af_atempo.c Pavel Koshevoy
af_pan.c Nicolas George
vsrc_mandelbrot.c Michael Niedermayer
vf_yadif.c Michael Niedermayer
diff --git a/configure b/configure
index f95a204..8a2a5de 100755
--- a/configure
+++ b/configure
@@ -1702,6 +1702,7 @@ amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
ass_filter_deps="libass"
asyncts_filter_deps="avresample"
+atempo_filter_deps="avcodec"
blackframe_filter_deps="gpl"
boxblur_filter_deps="gpl"
colormatrix_filter_deps="gpl"
diff --git a/doc/filters.texi b/doc/filters.texi
index ac79c4c..e862a21 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -273,6 +273,24 @@ For example, to resample the input audio to 44100Hz:
aresample=44100
@end example
+ at section atempo
+
+Adjust audio tempo.
+
+The filter accepts exactly one parameter, the audio tempo. If not
+specified then the filter will assume nominal 1.0 tempo. Tempo must
+be in the [0.5, 2.0] range.
+
+For example, to slow down audio to 80% tempo:
+ at example
+atempo=0.8
+ at end example
+
+For example, to speed up audio to 125% tempo:
+ at example
+atempo=1.25
+ at end example
+
@section ashowinfo
Show a line containing various information for each input audio frame.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 29345fc..a1ced51 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -8,6 +8,7 @@ FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample
FFLIBS-$(CONFIG_ACONVERT_FILTER) += swresample
FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample
+FFLIBS-$(CONFIG_ATEMPO_FILTER) += avcodec
FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
FFLIBS-$(CONFIG_PAN_FILTER) += swresample
FFLIBS-$(CONFIG_REMOVELOGO_FILTER) += avformat avcodec
@@ -54,6 +55,7 @@ OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o
+OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c
new file mode 100644
index 0000000..94b5b58
--- /dev/null
+++ b/libavfilter/af_atempo.c
@@ -0,0 +1,1134 @@
+/*
+ * 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 "libavcodec/avfft.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/eval.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+/**
+ * A fragment of audio waveform
+ */
+typedef struct {
+ // 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;
+
+ // FFT transform of the down-mixed mono fragment, used for
+ // fast waveform alignment via correlation in frequency domain:
+ FFTComplex *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 {
+ // 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];
+
+ // 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;
+
+ // cumulative alignment drift:
+ int drift;
+
+ // 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:
+ FFTContext *fft_forward;
+ FFTContext *fft_inverse;
+ FFTComplex *correlation;
+
+ // for managing AVFilterPad.request_frame and AVFilterPad.filter_samples
+ int request_fulfilled;
+ AVFilterBufferRef *dst_buffer;
+ uint8_t *dst;
+ uint8_t *dst_end;
+ uint64_t nsamples_in;
+ uint64_t nsamples_out;
+
+} ATempoContext;
+
+/**
+ * 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->drift = 0;
+ atempo->nfrag = 0;
+ atempo->state = YAE_LOAD_FRAGMENT;
+
+ atempo->position[0] = 0;
+ atempo->position[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);
+
+ avfilter_unref_bufferp(&atempo->dst_buffer);
+ atempo->dst = NULL;
+ atempo->dst_end = NULL;
+
+ atempo->request_fulfilled = 0;
+ atempo->nsamples_in = 0;
+ atempo->nsamples_out = 0;
+}
+
+#define REALLOC_OR_FAIL(field, field_size) \
+ do { \
+ field = av_realloc(field, (field_size)); \
+ if (!field) \
+ 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;
+ 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:
+ REALLOC_OR_FAIL(atempo->frag[0].data,
+ atempo->window * atempo->stride);
+
+ REALLOC_OR_FAIL(atempo->frag[1].data,
+ atempo->window * atempo->stride);
+
+ REALLOC_OR_FAIL(atempo->frag[0].xdat,
+ atempo->window * 2 * sizeof(FFTComplex));
+
+ REALLOC_OR_FAIL(atempo->frag[1].xdat,
+ atempo->window * 2 * sizeof(FFTComplex));
+
+ // initialize FFT contexts:
+ av_fft_end(atempo->fft_forward);
+ av_fft_end(atempo->fft_inverse);
+
+ atempo->fft_forward = av_fft_init(nlevels + 1, 0);
+ if (!atempo->fft_forward) {
+ return AVERROR(ENOMEM);
+ }
+
+ atempo->fft_inverse = av_fft_init(nlevels + 1, 1);
+ if (!atempo->fft_inverse) {
+ return AVERROR(ENOMEM);
+ }
+
+ REALLOC_OR_FAIL(atempo->correlation,
+ atempo->window * 2 * sizeof(FFTComplex));
+
+ atempo->ring = atempo->window * 3;
+ REALLOC_OR_FAIL(atempo->buffer, atempo->ring * atempo->stride);
+
+ // initialize the Hann window function:
+ REALLOC_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_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
+{
+ ATempoContext *atempo = ctx->priv;
+ char *tail = NULL;
+ double tempo = av_strtod(arg_tempo, &tail);
+
+ if (tail && *tail) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid tempo value '%s'\n", arg_tempo);
+ return AVERROR(EINVAL);
+ }
+
+ if (tempo < 0.5 || tempo > 2.0) {
+ av_log(ctx, AV_LOG_ERROR, "Tempo value %f exceeds [0.5, 2.0] range\n",
+ tempo);
+ return AVERROR(EINVAL);
+ }
+
+ atempo->tempo = tempo;
+ return 0;
+}
+
+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];
+}
+
+inline static void yae_transform(FFTComplex *xdat, FFTContext *fft)
+{
+ av_fft_permute(fft, xdat);
+ av_fft_calc(fft, xdat);
+}
+
+/**
+ * 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); \
+ \
+ FFTComplex *xdat = frag->xdat; \
+ scalar_type tmp; \
+ \
+ if (atempo->channels == 1) { \
+ for (; src < src_end; blend++) { \
+ tmp = *(const scalar_type *)src; \
+ src += sizeof(scalar_type); \
+ \
+ xdat->re = (FFTSample)tmp; \
+ xdat->im = 0; \
+ xdat++; \
+ } \
+ } else { \
+ FFTSample s, max, ti, si; \
+ int i; \
+ \
+ for (; src < src_end; blend++) { \
+ tmp = *(const scalar_type *)src; \
+ src += sizeof(scalar_type); \
+ \
+ max = (FFTSample)tmp; \
+ s = FFMIN((FFTSample)scalar_max, \
+ (FFTSample)fabsf(max)); \
+ \
+ for (i = 1; i < atempo->channels; i++) { \
+ tmp = *(const scalar_type *)src; \
+ src += sizeof(scalar_type); \
+ \
+ ti = (FFTSample)tmp; \
+ si = FFMIN((FFTSample)scalar_max, \
+ (FFTSample)fabsf(ti)); \
+ \
+ if (s < si) { \
+ s = si; \
+ max = ti; \
+ } \
+ } \
+ \
+ xdat->re = max; \
+ xdat->im = 0; \
+ xdat++; \
+ } \
+ } \
+ } 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;
+ const float *blend = atempo->hann;
+
+ // init complex data buffer used for FFT and Correlation:
+ memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window * 2);
+
+ 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:
+ av_assert0(read_size <= atempo->ring);
+
+ 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;
+ zeros = 0;
+
+ if (frag->position[0] < start) {
+ // what we don't have we substitute with zeros:
+ zeros = FFMIN(start - frag->position[0], (int64_t)nsamples);
+ av_assert0(zeros != nsamples);
+
+ memset(dst, 0, zeros * atempo->stride);
+ dst += zeros * atempo->stride;
+ }
+
+ if (zeros == nsamples) {
+ return 0;
+ }
+
+ // 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);
+ dst += 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 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,
+ FFTComplex *correlation,
+ FFTContext *fft_inverse)
+{
+ const FFTComplex *xa = prev->xdat;
+ const FFTComplex *xb = frag->xdat;
+ FFTComplex *xc = correlation;
+
+ int best_offset = -drift;
+ FFTSample best_metric = -FLT_MAX;
+
+ int i0;
+ int i1;
+ int i;
+
+ for (i = 0; i < window * 2; 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 FFT:
+ yae_transform(correlation, fft_inverse);
+
+ // identify cross-correlation peaks:
+
+ 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);
+
+ xc = correlation + i0;
+ for (i = i0; i < i1; i++, xc++) {
+ FFTSample metric = xc->re;
+
+ // normalize:
+ FFTSample drifti = (FFTSample)(drift + i);
+ metric *= drifti * drifti;
+
+ 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 int delta_max = atempo->window / 2;
+ const int correction = yae_align(frag,
+ prev,
+ atempo->window,
+ delta_max,
+ atempo->drift,
+ atempo->correlation,
+ atempo->fft_inverse);
+
+ if (correction) {
+ // adjust fragment position:
+ frag->position[0] -= correction;
+
+ // clear so that the fragment can be reloaded:
+ frag->nsamples = 0;
+
+ // update cumulative correction drift counter:
+ atempo->drift += correction;
+ }
+
+ 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;
+ }
+
+ // build a multi-resolution pyramid for fragment alignment:
+ yae_downmix(atempo, yae_curr_frag(atempo));
+
+ // apply FFT:
+ yae_transform(yae_curr_frag(atempo)->xdat, atempo->fft_forward);
+
+ // 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;
+ }
+
+ // build a multi-resolution pyramid for fragment alignment:
+ yae_downmix(atempo, yae_curr_frag(atempo));
+
+ // apply FFT:
+ yae_transform(yae_curr_frag(atempo)->xdat, atempo->fft_forward);
+
+ 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->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) {
+ // build a multi-resolution pyramid for fragment alignment:
+ yae_downmix(atempo, frag);
+
+ // apply FFT:
+ yae_transform(frag->xdat, atempo->fft_forward);
+
+ // 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);
+ }
+ }
+
+ // flush the remaininder 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, const char *args, void *opaque)
+{
+ ATempoContext *atempo = ctx->priv;
+
+ // NOTE: this assumes that the caller has memset ctx->priv to 0:
+ atempo->format = AV_SAMPLE_FMT_NONE;
+ atempo->tempo = 1.0;
+ atempo->state = YAE_LOAD_FRAGMENT;
+
+ return args ? yae_set_tempo(ctx, args) : 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ATempoContext *atempo = ctx->priv;
+ yae_clear(atempo);
+
+ av_freep(&atempo->frag[0].data);
+ av_freep(&atempo->frag[1].data);
+ av_freep(&atempo->frag[0].xdat);
+ av_freep(&atempo->frag[1].xdat);
+
+ av_freep(&atempo->buffer);
+ av_freep(&atempo->hann);
+ av_freep(&atempo->correlation);
+
+ av_fft_end(atempo->fft_forward);
+ atempo->fft_forward = NULL;
+
+ av_fft_end(atempo->fft_inverse);
+ atempo->fft_inverse = NULL;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterChannelLayouts *layouts = NULL;
+ AVFilterFormats *formats = NULL;
+
+ // 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.
+ //
+ 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
+ };
+
+ layouts = ff_all_channel_layouts();
+ if (!layouts) {
+ return AVERROR(ENOMEM);
+ }
+ ff_set_common_channel_layouts(ctx, layouts);
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats) {
+ return AVERROR(ENOMEM);
+ }
+ ff_set_common_formats(ctx, formats);
+
+ formats = ff_all_samplerates();
+ if (!formats) {
+ return AVERROR(ENOMEM);
+ }
+ ff_set_common_samplerates(ctx, formats);
+
+ return 0;
+}
+
+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;
+ int channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
+
+ return yae_reset(atempo, format, sample_rate, channels);
+}
+
+static void push_samples(ATempoContext *atempo,
+ AVFilterLink *outlink,
+ int n_out)
+{
+ atempo->dst_buffer->audio->sample_rate = outlink->sample_rate;
+ atempo->dst_buffer->audio->nb_samples = n_out;
+
+ // adjust the PTS:
+ atempo->dst_buffer->pts =
+ av_rescale_q(atempo->nsamples_out,
+ (AVRational){ 1, outlink->sample_rate },
+ outlink->time_base);
+
+ ff_filter_samples(outlink, atempo->dst_buffer);
+ atempo->dst_buffer = NULL;
+ atempo->dst = NULL;
+ atempo->dst_end = NULL;
+
+ atempo->nsamples_out += n_out;
+}
+
+static void filter_samples(AVFilterLink *inlink,
+ AVFilterBufferRef *src_buffer)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ATempoContext *atempo = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+
+ int n_in = src_buffer->audio->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;
+
+ while (src < src_end) {
+ if (!atempo->dst_buffer) {
+ atempo->dst_buffer = ff_get_audio_buffer(outlink,
+ AV_PERM_WRITE,
+ n_out);
+ avfilter_copy_buffer_ref_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) {
+ push_samples(atempo, outlink, n_out);
+ atempo->request_fulfilled = 1;
+ }
+ }
+
+ atempo->nsamples_in += n_in;
+ avfilter_unref_bufferp(&src_buffer);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ ATempoContext *atempo = ctx->priv;
+ int ret;
+
+ atempo->request_fulfilled = 0;
+ do {
+ ret = avfilter_request_frame(ctx->inputs[0]);
+ }
+ while (!atempo->request_fulfilled && ret >= 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,
+ AV_PERM_WRITE,
+ n_max);
+
+ 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) {
+ push_samples(atempo, outlink, n_out);
+ }
+ }
+
+ avfilter_unref_bufferp(&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)
+{
+ return !strcmp(cmd, "tempo") ? yae_set_tempo(ctx, arg) : AVERROR(ENOSYS);
+}
+
+AVFilter avfilter_af_atempo = {
+ .name = "atempo",
+ .description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .process_command = process_command,
+ .priv_size = sizeof(ATempoContext),
+
+ .inputs = (const AVFilterPad[]) {
+ { .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .config_props = config_props,
+ .min_perms = AV_PERM_READ, },
+ { .name = NULL}
+ },
+
+ .outputs = (const AVFilterPad[]) {
+ { .name = "default",
+ .request_frame = request_frame,
+ .type = AVMEDIA_TYPE_AUDIO, },
+ { .name = NULL}
+ },
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index b9d44f2..e8c8406 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -44,6 +44,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ASPLIT, asplit, af);
REGISTER_FILTER (ASTREAMSYNC, astreamsync, af);
REGISTER_FILTER (ASYNCTS, asyncts, af);
+ REGISTER_FILTER (ATEMPO, atempo, af);
REGISTER_FILTER (EARWAX, earwax, af);
REGISTER_FILTER (PAN, pan, af);
REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
--
1.7.7
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