mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-02 03:06:28 +02:00
185d1f3bfc
Signed-off-by: Paul B Mahol <onemda@gmail.com>
1170 lines
37 KiB
C
1170 lines
37 KiB
C
/*
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* Copyright (c) 2012 Pavel Koshevoy <pkoshevoy at gmail dot com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* tempo scaling audio filter -- an implementation of WSOLA algorithm
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*
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* Based on MIT licensed yaeAudioTempoFilter.h and yaeAudioFragment.h
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* from Apprentice Video player by Pavel Koshevoy.
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* https://sourceforge.net/projects/apprenticevideo/
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*
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* An explanation of SOLA algorithm is available at
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* http://www.surina.net/article/time-and-pitch-scaling.html
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*
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* WSOLA is very similar to SOLA, only one major difference exists between
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* these algorithms. SOLA shifts audio fragments along the output stream,
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* where as WSOLA shifts audio fragments along the input stream.
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*
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* The advantage of WSOLA algorithm is that the overlap region size is
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* always the same, therefore the blending function is constant and
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* can be precomputed.
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*/
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#include <float.h>
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#include "libavcodec/avfft.h"
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#include "libavutil/avassert.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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/**
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* A fragment of audio waveform
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*/
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typedef struct {
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// index of the first sample of this fragment in the overall waveform;
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// 0: input sample position
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// 1: output sample position
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int64_t position[2];
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// original packed multi-channel samples:
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uint8_t *data;
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// number of samples in this fragment:
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int nsamples;
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// rDFT transform of the down-mixed mono fragment, used for
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// fast waveform alignment via correlation in frequency domain:
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FFTSample *xdat;
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} AudioFragment;
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/**
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* Filter state machine states
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*/
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typedef enum {
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YAE_LOAD_FRAGMENT,
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YAE_ADJUST_POSITION,
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YAE_RELOAD_FRAGMENT,
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YAE_OUTPUT_OVERLAP_ADD,
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YAE_FLUSH_OUTPUT,
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} FilterState;
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/**
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* Filter state machine
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*/
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typedef struct {
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// ring-buffer of input samples, necessary because some times
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// input fragment position may be adjusted backwards:
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uint8_t *buffer;
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// ring-buffer maximum capacity, expressed in sample rate time base:
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int ring;
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// ring-buffer house keeping:
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int size;
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int head;
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int tail;
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// 0: input sample position corresponding to the ring buffer tail
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// 1: output sample position
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int64_t position[2];
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// sample format:
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enum AVSampleFormat format;
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// number of channels:
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int channels;
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// row of bytes to skip from one sample to next, across multple channels;
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// stride = (number-of-channels * bits-per-sample-per-channel) / 8
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int stride;
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// fragment window size, power-of-two integer:
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int window;
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// Hann window coefficients, for feathering
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// (blending) the overlapping fragment region:
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float *hann;
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// tempo scaling factor:
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double tempo;
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// cumulative alignment drift:
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int drift;
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// current/previous fragment ring-buffer:
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AudioFragment frag[2];
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// current fragment index:
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uint64_t nfrag;
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// current state:
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FilterState state;
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// for fast correlation calculation in frequency domain:
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RDFTContext *real_to_complex;
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RDFTContext *complex_to_real;
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FFTSample *correlation;
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// for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
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int request_fulfilled;
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AVFilterBufferRef *dst_buffer;
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uint8_t *dst;
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uint8_t *dst_end;
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uint64_t nsamples_in;
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uint64_t nsamples_out;
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} ATempoContext;
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/**
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* Reset filter to initial state, do not deallocate existing local buffers.
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*/
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static void yae_clear(ATempoContext *atempo)
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{
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atempo->size = 0;
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atempo->head = 0;
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atempo->tail = 0;
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atempo->drift = 0;
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atempo->nfrag = 0;
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atempo->state = YAE_LOAD_FRAGMENT;
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atempo->position[0] = 0;
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atempo->position[1] = 0;
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atempo->frag[0].position[0] = 0;
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atempo->frag[0].position[1] = 0;
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atempo->frag[0].nsamples = 0;
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atempo->frag[1].position[0] = 0;
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atempo->frag[1].position[1] = 0;
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atempo->frag[1].nsamples = 0;
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// shift left position of 1st fragment by half a window
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// so that no re-normalization would be required for
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// the left half of the 1st fragment:
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atempo->frag[0].position[0] = -(int64_t)(atempo->window / 2);
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atempo->frag[0].position[1] = -(int64_t)(atempo->window / 2);
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avfilter_unref_bufferp(&atempo->dst_buffer);
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atempo->dst = NULL;
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atempo->dst_end = NULL;
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atempo->request_fulfilled = 0;
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atempo->nsamples_in = 0;
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atempo->nsamples_out = 0;
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}
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/**
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* Reset filter to initial state and deallocate all buffers.
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*/
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static void yae_release_buffers(ATempoContext *atempo)
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{
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yae_clear(atempo);
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av_freep(&atempo->frag[0].data);
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av_freep(&atempo->frag[1].data);
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av_freep(&atempo->frag[0].xdat);
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av_freep(&atempo->frag[1].xdat);
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av_freep(&atempo->buffer);
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av_freep(&atempo->hann);
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av_freep(&atempo->correlation);
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av_rdft_end(atempo->real_to_complex);
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atempo->real_to_complex = NULL;
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av_rdft_end(atempo->complex_to_real);
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atempo->complex_to_real = NULL;
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}
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/* av_realloc is not aligned enough; fortunately, the data does not need to
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* be preserved */
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#define RE_MALLOC_OR_FAIL(field, field_size) \
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do { \
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av_freep(&field); \
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field = av_malloc(field_size); \
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if (!field) { \
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yae_release_buffers(atempo); \
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return AVERROR(ENOMEM); \
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} \
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} while (0)
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/**
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* Prepare filter for processing audio data of given format,
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* sample rate and number of channels.
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*/
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static int yae_reset(ATempoContext *atempo,
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enum AVSampleFormat format,
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int sample_rate,
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int channels)
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{
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const int sample_size = av_get_bytes_per_sample(format);
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uint32_t nlevels = 0;
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uint32_t pot;
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int i;
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atempo->format = format;
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atempo->channels = channels;
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atempo->stride = sample_size * channels;
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// pick a segment window size:
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atempo->window = sample_rate / 24;
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// adjust window size to be a power-of-two integer:
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nlevels = av_log2(atempo->window);
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pot = 1 << nlevels;
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av_assert0(pot <= atempo->window);
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if (pot < atempo->window) {
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atempo->window = pot * 2;
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nlevels++;
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}
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// initialize audio fragment buffers:
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RE_MALLOC_OR_FAIL(atempo->frag[0].data, atempo->window * atempo->stride);
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RE_MALLOC_OR_FAIL(atempo->frag[1].data, atempo->window * atempo->stride);
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RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, atempo->window * sizeof(FFTComplex));
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RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, atempo->window * sizeof(FFTComplex));
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// initialize rDFT contexts:
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av_rdft_end(atempo->real_to_complex);
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atempo->real_to_complex = NULL;
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av_rdft_end(atempo->complex_to_real);
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atempo->complex_to_real = NULL;
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atempo->real_to_complex = av_rdft_init(nlevels + 1, DFT_R2C);
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if (!atempo->real_to_complex) {
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yae_release_buffers(atempo);
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return AVERROR(ENOMEM);
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}
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atempo->complex_to_real = av_rdft_init(nlevels + 1, IDFT_C2R);
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if (!atempo->complex_to_real) {
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yae_release_buffers(atempo);
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return AVERROR(ENOMEM);
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}
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RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window * sizeof(FFTComplex));
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atempo->ring = atempo->window * 3;
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RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring * atempo->stride);
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// initialize the Hann window function:
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RE_MALLOC_OR_FAIL(atempo->hann, atempo->window * sizeof(float));
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for (i = 0; i < atempo->window; i++) {
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double t = (double)i / (double)(atempo->window - 1);
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double h = 0.5 * (1.0 - cos(2.0 * M_PI * t));
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atempo->hann[i] = (float)h;
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}
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yae_clear(atempo);
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return 0;
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}
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static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
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{
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ATempoContext *atempo = ctx->priv;
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char *tail = NULL;
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double tempo = av_strtod(arg_tempo, &tail);
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if (tail && *tail) {
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av_log(ctx, AV_LOG_ERROR, "Invalid tempo value '%s'\n", arg_tempo);
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return AVERROR(EINVAL);
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}
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if (tempo < 0.5 || tempo > 2.0) {
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av_log(ctx, AV_LOG_ERROR, "Tempo value %f exceeds [0.5, 2.0] range\n",
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tempo);
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return AVERROR(EINVAL);
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}
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atempo->tempo = tempo;
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return 0;
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}
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inline static AudioFragment *yae_curr_frag(ATempoContext *atempo)
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{
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return &atempo->frag[atempo->nfrag % 2];
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}
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inline static AudioFragment *yae_prev_frag(ATempoContext *atempo)
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{
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return &atempo->frag[(atempo->nfrag + 1) % 2];
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}
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/**
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* A helper macro for initializing complex data buffer with scalar data
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* of a given type.
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*/
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#define yae_init_xdat(scalar_type, scalar_max) \
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do { \
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const uint8_t *src_end = src + \
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frag->nsamples * atempo->channels * sizeof(scalar_type); \
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\
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FFTSample *xdat = frag->xdat; \
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scalar_type tmp; \
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\
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if (atempo->channels == 1) { \
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for (; src < src_end; xdat++) { \
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tmp = *(const scalar_type *)src; \
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src += sizeof(scalar_type); \
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\
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*xdat = (FFTSample)tmp; \
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} \
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} else { \
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FFTSample s, max, ti, si; \
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int i; \
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\
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for (; src < src_end; xdat++) { \
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tmp = *(const scalar_type *)src; \
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src += sizeof(scalar_type); \
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\
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max = (FFTSample)tmp; \
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s = FFMIN((FFTSample)scalar_max, \
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(FFTSample)fabsf(max)); \
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\
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for (i = 1; i < atempo->channels; i++) { \
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tmp = *(const scalar_type *)src; \
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src += sizeof(scalar_type); \
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\
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ti = (FFTSample)tmp; \
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si = FFMIN((FFTSample)scalar_max, \
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(FFTSample)fabsf(ti)); \
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\
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if (s < si) { \
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s = si; \
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max = ti; \
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} \
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} \
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\
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*xdat = max; \
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} \
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} \
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} while (0)
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/**
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* Initialize complex data buffer of a given audio fragment
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* with down-mixed mono data of appropriate scalar type.
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*/
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static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
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{
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// shortcuts:
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const uint8_t *src = frag->data;
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// init complex data buffer used for FFT and Correlation:
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memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window);
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if (atempo->format == AV_SAMPLE_FMT_U8) {
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yae_init_xdat(uint8_t, 127);
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} else if (atempo->format == AV_SAMPLE_FMT_S16) {
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yae_init_xdat(int16_t, 32767);
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} else if (atempo->format == AV_SAMPLE_FMT_S32) {
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yae_init_xdat(int, 2147483647);
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} else if (atempo->format == AV_SAMPLE_FMT_FLT) {
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yae_init_xdat(float, 1);
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} else if (atempo->format == AV_SAMPLE_FMT_DBL) {
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yae_init_xdat(double, 1);
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}
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}
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/**
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* Populate the internal data buffer on as-needed basis.
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*
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* @return
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* 0 if requested data was already available or was successfully loaded,
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* AVERROR(EAGAIN) if more input data is required.
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*/
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static int yae_load_data(ATempoContext *atempo,
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const uint8_t **src_ref,
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const uint8_t *src_end,
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int64_t stop_here)
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{
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// shortcut:
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const uint8_t *src = *src_ref;
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const int read_size = stop_here - atempo->position[0];
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if (stop_here <= atempo->position[0]) {
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return 0;
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}
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// samples are not expected to be skipped:
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av_assert0(read_size <= atempo->ring);
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while (atempo->position[0] < stop_here && src < src_end) {
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int src_samples = (src_end - src) / atempo->stride;
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// load data piece-wise, in order to avoid complicating the logic:
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int nsamples = FFMIN(read_size, src_samples);
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int na;
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int nb;
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nsamples = FFMIN(nsamples, atempo->ring);
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na = FFMIN(nsamples, atempo->ring - atempo->tail);
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nb = FFMIN(nsamples - na, atempo->ring);
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if (na) {
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uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;
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memcpy(a, src, na * atempo->stride);
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src += na * atempo->stride;
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atempo->position[0] += na;
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atempo->size = FFMIN(atempo->size + na, atempo->ring);
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atempo->tail = (atempo->tail + na) % atempo->ring;
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atempo->head =
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atempo->size < atempo->ring ?
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atempo->tail - atempo->size :
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atempo->tail;
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}
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if (nb) {
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uint8_t *b = atempo->buffer;
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memcpy(b, src, nb * atempo->stride);
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src += nb * atempo->stride;
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atempo->position[0] += nb;
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atempo->size = FFMIN(atempo->size + nb, atempo->ring);
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atempo->tail = (atempo->tail + nb) % atempo->ring;
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atempo->head =
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atempo->size < atempo->ring ?
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atempo->tail - atempo->size :
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atempo->tail;
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}
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}
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// pass back the updated source buffer pointer:
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*src_ref = src;
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// sanity check:
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av_assert0(atempo->position[0] <= stop_here);
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return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);
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}
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/**
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* Populate current audio fragment data buffer.
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*
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* @return
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* 0 when the fragment is ready,
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* AVERROR(EAGAIN) if more input data is required.
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*/
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static int yae_load_frag(ATempoContext *atempo,
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const uint8_t **src_ref,
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const uint8_t *src_end)
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{
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// shortcuts:
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AudioFragment *frag = yae_curr_frag(atempo);
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uint8_t *dst;
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int64_t missing, start, zeros;
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uint32_t nsamples;
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const uint8_t *a, *b;
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int i0, i1, n0, n1, na, nb;
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int64_t stop_here = frag->position[0] + atempo->window;
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if (src_ref && yae_load_data(atempo, src_ref, src_end, stop_here) != 0) {
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return AVERROR(EAGAIN);
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}
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// calculate the number of samples we don't have:
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missing =
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stop_here > atempo->position[0] ?
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stop_here - atempo->position[0] : 0;
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nsamples =
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missing < (int64_t)atempo->window ?
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(uint32_t)(atempo->window - missing) : 0;
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// setup the output buffer:
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frag->nsamples = nsamples;
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dst = frag->data;
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|
|
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);
|
|
}
|
|
|
|
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(FFTSample *xcorr,
|
|
RDFTContext *complex_to_real,
|
|
const FFTComplex *xa,
|
|
const FFTComplex *xb,
|
|
const int window)
|
|
{
|
|
FFTComplex *xc = (FFTComplex *)xcorr;
|
|
int i;
|
|
|
|
// NOTE: first element requires special care -- Given Y = rDFT(X),
|
|
// Im(Y[0]) and Im(Y[N/2]) are always zero, therefore av_rdft_calc
|
|
// stores Re(Y[N/2]) in place of Im(Y[0]).
|
|
|
|
xc->re = xa->re * xb->re;
|
|
xc->im = xa->im * xb->im;
|
|
xa++;
|
|
xb++;
|
|
xc++;
|
|
|
|
for (i = 1; 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:
|
|
av_rdft_calc(complex_to_real, xcorr);
|
|
}
|
|
|
|
/**
|
|
* 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,
|
|
FFTSample *correlation,
|
|
RDFTContext *complex_to_real)
|
|
{
|
|
int best_offset = -drift;
|
|
FFTSample best_metric = -FLT_MAX;
|
|
FFTSample *xcorr;
|
|
|
|
int i0;
|
|
int i1;
|
|
int i;
|
|
|
|
yae_xcorr_via_rdft(correlation,
|
|
complex_to_real,
|
|
(const FFTComplex *)prev->xdat,
|
|
(const FFTComplex *)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++) {
|
|
FFTSample metric = *xcorr;
|
|
|
|
// normalize:
|
|
FFTSample drifti = (FFTSample)(drift + i);
|
|
metric *= drifti * (FFTSample)(i - i0) * (FFTSample)(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 int delta_max = atempo->window / 2;
|
|
const int correction = yae_align(frag,
|
|
prev,
|
|
atempo->window,
|
|
delta_max,
|
|
atempo->drift,
|
|
atempo->correlation,
|
|
atempo->complex_to_real);
|
|
|
|
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;
|
|
}
|
|
|
|
// down-mix to mono:
|
|
yae_downmix(atempo, yae_curr_frag(atempo));
|
|
|
|
// apply rDFT:
|
|
av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
|
|
|
|
// 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:
|
|
av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
|
|
|
|
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) {
|
|
// down-mix to mono:
|
|
yae_downmix(atempo, frag);
|
|
|
|
// apply rDFT:
|
|
av_rdft_calc(atempo->real_to_complex, frag->xdat);
|
|
|
|
// 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)
|
|
{
|
|
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_release_buffers(atempo);
|
|
}
|
|
|
|
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.
|
|
//
|
|
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
|
|
};
|
|
|
|
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_frame(outlink, atempo->dst_buffer);
|
|
atempo->dst_buffer = NULL;
|
|
atempo->dst = NULL;
|
|
atempo->dst_end = NULL;
|
|
|
|
atempo->nsamples_out += n_out;
|
|
}
|
|
|
|
static int filter_frame(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);
|
|
return 0;
|
|
}
|
|
|
|
static int request_frame(AVFilterLink *outlink)
|
|
{
|
|
AVFilterContext *ctx = outlink->src;
|
|
ATempoContext *atempo = ctx->priv;
|
|
int ret;
|
|
|
|
atempo->request_fulfilled = 0;
|
|
do {
|
|
ret = ff_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);
|
|
}
|
|
|
|
static const AVFilterPad atempo_inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = filter_frame,
|
|
.config_props = config_props,
|
|
.min_perms = AV_PERM_READ,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad atempo_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.request_frame = request_frame,
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
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 = atempo_inputs,
|
|
.outputs = atempo_outputs,
|
|
};
|