From 0c0d1bce7c582b82e49843acaa7d0fb4b1774b21 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 23 Mar 2012 17:45:48 -0400 Subject: avutil: add audio fifo buffer The functions operate on the sample level rather than the byte level and work with all audio sample formats. --- libavutil/audio_fifo.c | 193 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 193 insertions(+) create mode 100644 libavutil/audio_fifo.c (limited to 'libavutil/audio_fifo.c') diff --git a/libavutil/audio_fifo.c b/libavutil/audio_fifo.c new file mode 100644 index 0000000000..97c51a72c1 --- /dev/null +++ b/libavutil/audio_fifo.c @@ -0,0 +1,193 @@ +/* + * Audio FIFO + * Copyright (c) 2012 Justin Ruggles + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; 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 "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_free(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(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_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; +} -- cgit v1.2.3