mirror of https://github.com/FFmpeg/FFmpeg.git
The functions operate on the sample level rather than the byte level and work with all audio sample formats.pull/5/head
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/*
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* Audio FIFO |
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
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* |
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* This file is part of Libav. |
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* |
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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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* |
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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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* |
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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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*/ |
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/**
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* @file |
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* Audio FIFO |
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*/ |
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#include "avutil.h" |
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#include "audio_fifo.h" |
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#include "fifo.h" |
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#include "mem.h" |
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#include "samplefmt.h" |
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struct AVAudioFifo { |
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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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}; |
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void av_audio_fifo_free(AVAudioFifo *af) |
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{ |
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if (af) { |
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if (af->buf) { |
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int i; |
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for (i = 0; i < af->nb_buffers; i++) { |
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if (af->buf[i]) |
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av_fifo_free(af->buf[i]); |
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} |
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av_free(af->buf); |
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} |
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av_free(af); |
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} |
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} |
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AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, |
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int nb_samples) |
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{ |
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AVAudioFifo *af; |
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int buf_size, i; |
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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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return NULL; |
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af = av_mallocz(sizeof(*af)); |
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if (!af) |
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return NULL; |
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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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if (!af->buf) |
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goto error; |
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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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if (!af->buf[i]) |
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goto error; |
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} |
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af->allocated_samples = nb_samples; |
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return af; |
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error: |
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av_audio_fifo_free(af); |
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return NULL; |
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} |
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int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples) |
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{ |
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int i, ret, buf_size; |
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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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return ret; |
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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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return ret; |
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} |
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af->allocated_samples = nb_samples; |
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return 0; |
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} |
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int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples) |
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{ |
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int i, ret, size; |
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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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return ret; |
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} |
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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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if (ret != size) |
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return AVERROR_BUG; |
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} |
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af->nb_samples += nb_samples; |
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return nb_samples; |
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} |
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int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples) |
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{ |
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int i, ret, size; |
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if (nb_samples < 0) |
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return AVERROR(EINVAL); |
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nb_samples = FFMIN(nb_samples, af->nb_samples); |
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if (!nb_samples) |
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return 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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if ((ret = av_fifo_generic_read(af->buf[i], data[i], size, NULL)) < 0) |
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return AVERROR_BUG; |
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} |
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af->nb_samples -= nb_samples; |
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return nb_samples; |
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} |
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int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples) |
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{ |
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int i, size; |
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if (nb_samples < 0) |
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return AVERROR(EINVAL); |
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nb_samples = FFMIN(nb_samples, af->nb_samples); |
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if (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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} |
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return 0; |
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} |
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void av_audio_fifo_reset(AVAudioFifo *af) |
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{ |
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int i; |
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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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af->nb_samples = 0; |
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} |
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int av_audio_fifo_size(AVAudioFifo *af) |
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{ |
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return af->nb_samples; |
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} |
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int av_audio_fifo_space(AVAudioFifo *af) |
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{ |
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return af->allocated_samples - af->nb_samples; |
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} |
@ -0,0 +1,146 @@ |
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/*
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* Audio FIFO |
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
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* |
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* This file is part of Libav. |
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* |
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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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* |
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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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* |
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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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*/ |
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/**
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* @file |
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* Audio FIFO Buffer |
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*/ |
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#ifndef AVUTIL_AUDIO_FIFO_H |
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#define AVUTIL_AUDIO_FIFO_H |
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#include "avutil.h" |
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#include "fifo.h" |
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#include "samplefmt.h" |
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/**
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* @addtogroup lavu_audio |
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* @{ |
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*/ |
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/**
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* Context for an Audio FIFO Buffer. |
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* |
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* - Operates at the sample level rather than the byte level. |
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* - Supports multiple channels with either planar or packed sample format. |
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* - Automatic reallocation when writing to a full buffer. |
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*/ |
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typedef struct AVAudioFifo AVAudioFifo; |
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/**
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* Free an AVAudioFifo. |
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* |
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* @param af AVAudioFifo to free |
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*/ |
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void av_audio_fifo_free(AVAudioFifo *af); |
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/**
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* Allocate an AVAudioFifo. |
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* |
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* @param sample_fmt sample format |
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* @param channels number of channels |
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* @param nb_samples initial allocation size, in samples |
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* @return newly allocated AVAudioFifo, or NULL on error |
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*/ |
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AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, |
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int nb_samples); |
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/**
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* Reallocate an AVAudioFifo. |
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* |
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* @param af AVAudioFifo to reallocate |
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* @param nb_samples new allocation size, in samples |
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* @return 0 if OK, or negative AVERROR code on failure |
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*/ |
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int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples); |
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/**
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* Write data to an AVAudioFifo. |
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* |
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* The AVAudioFifo will be reallocated automatically if the available space |
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* is less than nb_samples. |
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* |
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* @see enum AVSampleFormat |
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* The documentation for AVSampleFormat describes the data layout. |
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* |
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* @param af AVAudioFifo to write to |
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* @param data audio data plane pointers |
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* @param nb_samples number of samples to write |
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* @return number of samples actually written, or negative AVERROR |
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* code on failure. |
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*/ |
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int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples); |
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/**
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* Read data from an AVAudioFifo. |
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* |
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* @see enum AVSampleFormat |
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* The documentation for AVSampleFormat describes the data layout. |
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* |
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* @param af AVAudioFifo to read from |
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* @param data audio data plane pointers |
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* @param nb_samples number of samples to read |
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* @return number of samples actually read, or negative AVERROR code |
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* on failure. |
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*/ |
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int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples); |
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/**
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* Drain data from an AVAudioFifo. |
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* |
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* Removes the data without reading it. |
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* |
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* @param af AVAudioFifo to drain |
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* @param nb_samples number of samples to drain |
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* @return 0 if OK, or negative AVERROR code on failure |
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*/ |
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int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples); |
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/**
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* Reset the AVAudioFifo buffer. |
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* |
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* This empties all data in the buffer. |
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* |
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* @param af AVAudioFifo to reset |
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*/ |
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void av_audio_fifo_reset(AVAudioFifo *af); |
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/**
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* Get the current number of samples in the AVAudioFifo available for reading. |
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* |
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* @param af the AVAudioFifo to query |
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* @return number of samples available for reading |
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*/ |
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int av_audio_fifo_size(AVAudioFifo *af); |
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/**
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* Get the current number of samples in the AVAudioFifo available for writing. |
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* |
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* @param af the AVAudioFifo to query |
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* @return number of samples available for writing |
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*/ |
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int av_audio_fifo_space(AVAudioFifo *af); |
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/**
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* @} |
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*/ |
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#endif /* AVUTIL_AUDIO_FIFO_H */ |
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