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/*
* Audio FIFO
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.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
* Audio FIFO
*/
#include "avutil.h"
#include "audio_fifo.h"
#include "common.h"
#include "fifo.h"
#include "mem.h"
#include "samplefmt.h"
struct AVAudioFifo {
AVFifoBuffer **buf; /**< single buffer for interleaved, per-channel buffers for planar */
int nb_buffers; /**< number of buffers */
int nb_samples; /**< number of samples currently in the FIFO */
int allocated_samples; /**< current allocated size, in samples */
int channels; /**< number of channels */
enum AVSampleFormat sample_fmt; /**< sample format */
int sample_size; /**< size, in bytes, of one sample in a buffer */
};
void av_audio_fifo_free(AVAudioFifo *af)
{
if (af) {
if (af->buf) {
int i;
for (i = 0; i < af->nb_buffers; i++) {
if (af->buf[i])
av_fifo_free(af->buf[i]);
}
av_freep(&af->buf);
}
av_free(af);
}
}
AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
int nb_samples)
{
AVAudioFifo *af;
int buf_size, i;
/* get channel buffer size (also validates parameters) */
if (av_samples_get_buffer_size(&buf_size, channels, nb_samples, sample_fmt, 1) < 0)
return NULL;
af = av_mallocz(sizeof(*af));
if (!af)
return NULL;
af->channels = channels;
af->sample_fmt = sample_fmt;
af->sample_size = buf_size / nb_samples;
af->nb_buffers = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
af->buf = av_mallocz_array(af->nb_buffers, sizeof(*af->buf));
if (!af->buf)
goto error;
for (i = 0; i < af->nb_buffers; i++) {
af->buf[i] = av_fifo_alloc(buf_size);
if (!af->buf[i])
goto error;
}
af->allocated_samples = nb_samples;
return af;
error:
av_audio_fifo_free(af);
return NULL;
}
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
{
int i, ret, buf_size;
if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples,
af->sample_fmt, 1)) < 0)
return ret;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_realloc2(af->buf[i], buf_size)) < 0)
return ret;
}
af->allocated_samples = nb_samples;
return 0;
}
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
{
int i, ret, size;
/* automatically reallocate buffers if needed */
if (av_audio_fifo_space(af) < nb_samples) {
int current_size = av_audio_fifo_size(af);
/* check for integer overflow in new size calculation */
if (INT_MAX / 2 - current_size < nb_samples)
return AVERROR(EINVAL);
/* reallocate buffers */
if ((ret = av_audio_fifo_realloc(af, 2 * (current_size + nb_samples))) < 0)
return ret;
}
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
ret = av_fifo_generic_write(af->buf[i], data[i], size, NULL);
if (ret != size)
return AVERROR_BUG;
}
af->nb_samples += nb_samples;
return nb_samples;
}
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
{
int i, ret, size;
if (nb_samples < 0)
return AVERROR(EINVAL);
nb_samples = FFMIN(nb_samples, af->nb_samples);
if (!nb_samples)
return 0;
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_generic_peek(af->buf[i], data[i], size, NULL)) < 0)
return AVERROR_BUG;
}
return nb_samples;
}
int av_audio_fifo_peek_at(AVAudioFifo *af, void **data, int nb_samples, int offset)
{
int i, ret, size;
if (offset < 0 || offset >= af->nb_samples)
return AVERROR(EINVAL);
if (nb_samples < 0)
return AVERROR(EINVAL);
nb_samples = FFMIN(nb_samples, af->nb_samples);
if (!nb_samples)
return 0;
if (offset > af->nb_samples - nb_samples)
return AVERROR(EINVAL);
offset *= af->sample_size;
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_generic_peek_at(af->buf[i], data[i], offset, size, NULL)) < 0)
return AVERROR_BUG;
}
return nb_samples;
}
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
{
int i, ret, size;
if (nb_samples < 0)
return AVERROR(EINVAL);
nb_samples = FFMIN(nb_samples, af->nb_samples);
if (!nb_samples)
return 0;
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_generic_read(af->buf[i], data[i], size, NULL)) < 0)
return AVERROR_BUG;
}
af->nb_samples -= nb_samples;
return nb_samples;
}
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
{
int i, size;
if (nb_samples < 0)
return AVERROR(EINVAL);
nb_samples = FFMIN(nb_samples, af->nb_samples);
if (nb_samples) {
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++)
av_fifo_drain(af->buf[i], size);
af->nb_samples -= nb_samples;
}
return 0;
}
void av_audio_fifo_reset(AVAudioFifo *af)
{
int i;
for (i = 0; i < af->nb_buffers; i++)
av_fifo_reset(af->buf[i]);
af->nb_samples = 0;
}
int av_audio_fifo_size(AVAudioFifo *af)
{
return af->nb_samples;
}
int av_audio_fifo_space(AVAudioFifo *af)
{
return af->allocated_samples - af->nb_samples;
}
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