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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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* This file is part of Libav.
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* Libav 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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* Libav 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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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; 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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#include "audio_fifo.h"
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#include "samplefmt.h"
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AVFifoBuffer **buf; /**< single buffer for interleaved, per-channel buffers for planar */
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int nb_buffers; /**< number of buffers */
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int nb_samples; /**< number of samples currently in the FIFO */
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int allocated_samples; /**< current allocated size, in samples */
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int channels; /**< number of channels */
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enum AVSampleFormat sample_fmt; /**< sample format */
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int sample_size; /**< size, in bytes, of one sample in a buffer */
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void av_audio_fifo_free(AVAudioFifo *af)
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for (i = 0; i < af->nb_buffers; i++) {
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av_fifo_free(af->buf[i]);
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AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
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/* get channel buffer size (also validates parameters) */
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if (av_samples_get_buffer_size(&buf_size, channels, nb_samples, sample_fmt, 1) < 0)
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af = av_mallocz(sizeof(*af));
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af->channels = channels;
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af->sample_fmt = sample_fmt;
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af->sample_size = buf_size / nb_samples;
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af->nb_buffers = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
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af->buf = av_mallocz(af->nb_buffers * sizeof(*af->buf));
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for (i = 0; i < af->nb_buffers; i++) {
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af->buf[i] = av_fifo_alloc(buf_size);
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af->allocated_samples = nb_samples;
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av_audio_fifo_free(af);
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int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
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if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples,
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af->sample_fmt, 1)) < 0)
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for (i = 0; i < af->nb_buffers; i++) {
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if ((ret = av_fifo_realloc2(af->buf[i], buf_size)) < 0)
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af->allocated_samples = nb_samples;
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int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
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/* automatically reallocate buffers if needed */
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if (av_audio_fifo_space(af) < nb_samples) {
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int current_size = av_audio_fifo_size(af);
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/* check for integer overflow in new size calculation */
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if (INT_MAX / 2 - current_size < nb_samples)
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return AVERROR(EINVAL);
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/* reallocate buffers */
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if ((ret = av_audio_fifo_realloc(af, 2 * (current_size + nb_samples))) < 0)
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size = nb_samples * af->sample_size;
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for (i = 0; i < af->nb_buffers; i++) {
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ret = av_fifo_generic_write(af->buf[i], data[i], size, NULL);
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af->nb_samples += nb_samples;
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int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
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return AVERROR(EINVAL);
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nb_samples = FFMIN(nb_samples, af->nb_samples);
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size = nb_samples * af->sample_size;
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for (i = 0; i < af->nb_buffers; i++) {
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if ((ret = av_fifo_generic_read(af->buf[i], data[i], size, NULL)) < 0)
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af->nb_samples -= nb_samples;
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int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
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return AVERROR(EINVAL);
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nb_samples = FFMIN(nb_samples, af->nb_samples);
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size = nb_samples * af->sample_size;
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for (i = 0; i < af->nb_buffers; i++)
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av_fifo_drain(af->buf[i], size);
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af->nb_samples -= nb_samples;
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void av_audio_fifo_reset(AVAudioFifo *af)
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for (i = 0; i < af->nb_buffers; i++)
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av_fifo_reset(af->buf[i]);
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int av_audio_fifo_size(AVAudioFifo *af)
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return af->nb_samples;
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int av_audio_fifo_space(AVAudioFifo *af)
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return af->allocated_samples - af->nb_samples;