On Mon,  9 Apr 2012 17:27:51 -0400, Justin Ruggles <[email protected]> 
wrote:
> The functions operate on the sample level rather than the byte level and work
> with all audio sample formats.
> ---
>  libavutil/Makefile     |    2 +
>  libavutil/audio_fifo.c |  197 
> ++++++++++++++++++++++++++++++++++++++++++++++++
>  libavutil/audio_fifo.h |  140 ++++++++++++++++++++++++++++++++++
>  3 files changed, 339 insertions(+), 0 deletions(-)
>  create mode 100644 libavutil/audio_fifo.c
>  create mode 100644 libavutil/audio_fifo.h
> 
> diff --git a/libavutil/Makefile b/libavutil/Makefile
> index a73fb79..168799d 100644
> --- a/libavutil/Makefile
> +++ b/libavutil/Makefile
> @@ -3,6 +3,7 @@ NAME = avutil
>  HEADERS = adler32.h                                                     \
>            aes.h                                                         \
>            attributes.h                                                  \
> +          audio_fifo.h                                                  \
>            audioconvert.h                                                \
>            avassert.h                                                    \
>            avstring.h                                                    \
> @@ -40,6 +41,7 @@ BUILT_HEADERS = avconfig.h
>  
>  OBJS = adler32.o                                                        \
>         aes.o                                                            \
> +       audio_fifo.o                                                     \
>         audioconvert.o                                                   \
>         avstring.o                                                       \
>         base64.o                                                         \
> diff --git a/libavutil/audio_fifo.c b/libavutil/audio_fifo.c
> new file mode 100644
> index 0000000..551c8bf
> --- /dev/null
> +++ b/libavutil/audio_fifo.c
> @@ -0,0 +1,197 @@
> +/*
> + * Audio FIFO
> + * Copyright (c) 2012 Justin Ruggles <[email protected]>
> + *
> + * 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 planar;                     /**< is the sample format planar */
> +    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(AVAudioFifo));

sizeof(*af) is better karma

> +    if (!af)
> +        return NULL;
> +
> +    af->channels    = channels;
> +    af->sample_fmt  = sample_fmt;
> +    af->planar      = av_sample_fmt_is_planar(sample_fmt);

Why does this variable exist? I don't see it used anywhere except two
lines below.

> +    af->sample_size = buf_size / nb_samples;
> +    af->nb_buffers  = af->planar ? 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;
> +    af->nb_samples        = 0;

No need, you're already using mallocz

> +
> +    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);
> +    if (nb_samples > av_audio_fifo_size(af))
> +        nb_samples = av_audio_fifo_size(af);
> +    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);
> +    if (nb_samples > av_audio_fifo_size(af))
> +        nb_samples = av_audio_fifo_size(af);
> +
> +    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 av_fifo_size(af->buf[0]) / af->sample_size;
> +}
> +
> +int av_audio_fifo_space(AVAudioFifo *af)
> +{
> +    return av_fifo_space(af->buf[0]) / af->sample_size;
> +}
> diff --git a/libavutil/audio_fifo.h b/libavutil/audio_fifo.h
> new file mode 100644
> index 0000000..3e1535e
> --- /dev/null
> +++ b/libavutil/audio_fifo.h
> @@ -0,0 +1,140 @@
> +/*
> + * Audio FIFO
> + * Copyright (c) 2012 Justin Ruggles <[email protected]>
> + *
> + * 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 Buffer
> + */
> +
> +#ifndef AVUTIL_AUDIO_FIFO_H
> +#define AVUTIL_AUDIO_FIFO_H
> +
> +#include "avutil.h"
> +#include "fifo.h"
> +#include "samplefmt.h"
> +
> +/**
> + * @addtogroup lavu_audio
> + * @{
> + */
> +
> +/**
> + * Context for an Audio FIFO Buffer.
> + *
> + * - Operates at the sample level rather than the byte level.
> + * - Supports multiple channels with either planar and packed sample format.
> + * - Automatic reallocation when writing to a full buffer.
> + */
> +typedef struct AVAudioFifo AVAudioFifo;
> +
> +/**
> + * Free an AVAudioFifo
> + *
> + * @param af  AVAudioFifo to free
> + */
> +void av_audio_fifo_free(AVAudioFifo *af);
> +
> +/**
> + * Allocate an AVAudioFifo
> + *
> + * @param sample_fmt  sample format
> + * @param channels    number of channels
> + * @param nb_samples  initial allocation size, in samples
> + * @return            newly allocated AVAudioFifo, or NULL on error
> + */
> +AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int 
> channels,
> +                                 int nb_samples);
> +
> +/**
> + * Reallocate an AVAudioFifo
> + *
> + * @param af          AVAudioFifo to reallocate
> + * @param nb_samples  new allocation size, in samples
> + * @return            0 if ok, or negative AVERROR code on failure
> + */
> +int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples);
> +
> +/**
> + * Write data to an AVAudioFifo
> + *
> + * The AVAudioFifo will be reallocated automatically if the available space
> + * is less than nb_samples.
> + *
> + * @param af          AVAudioFifo to write to
> + * @param data        audio data plane pointers
> + * @param nb_samples  number of samples to write
> + * @return            number of samples actually written, or negative AVERROR
> +                      code on failure.
> + */
> +int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples);
> +
> +/**
> + * Read data from an AVAudioFifo
> + *
> + * @param af          AVAudioFifo to read from
> + * @param data        audio data plane pointers

Please be more verbose here. This can be confusing to people who are
less familiar with lavc API.

Also you're missing the bump/APIchanges dance.
Otherwise looks fine.

-- 
Anton Khirnov
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