/* * 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++) { av_fifo_freep(&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, 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 (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; }