avutil: add audio fifo buffer

The functions operate on the sample level rather than the byte level and work
with all audio sample formats.
pull/5/head
Justin Ruggles 13 years ago
parent 2b98377935
commit 0c0d1bce7c
  1. 12
      doc/APIchanges
  2. 2
      libavutil/Makefile
  3. 193
      libavutil/audio_fifo.c
  4. 146
      libavutil/audio_fifo.h
  5. 4
      libavutil/avutil.h

@ -12,6 +12,18 @@ libavutil: 2011-04-18
API changes, most recent first: API changes, most recent first:
2012-xx-xx - xxxxxxx - lavu 51.28.0 - audio_fifo.h
Add audio FIFO functions:
av_audio_fifo_free()
av_audio_fifo_alloc()
av_audio_fifo_realloc()
av_audio_fifo_write()
av_audio_fifo_read()
av_audio_fifo_drain()
av_audio_fifo_reset()
av_audio_fifo_size()
av_audio_fifo_space()
2012-xx-xx - xxxxxxx - lavfi 2.16.0 - avfiltergraph.h 2012-xx-xx - xxxxxxx - lavfi 2.16.0 - avfiltergraph.h
Add avfilter_graph_parse2(), avfilter_inout_alloc() and Add avfilter_graph_parse2(), avfilter_inout_alloc() and
avfilter_inout_free() functions. avfilter_inout_free() functions.

@ -3,6 +3,7 @@ NAME = avutil
HEADERS = adler32.h \ HEADERS = adler32.h \
aes.h \ aes.h \
attributes.h \ attributes.h \
audio_fifo.h \
audioconvert.h \ audioconvert.h \
avassert.h \ avassert.h \
avstring.h \ avstring.h \
@ -40,6 +41,7 @@ BUILT_HEADERS = avconfig.h
OBJS = adler32.o \ OBJS = adler32.o \
aes.o \ aes.o \
audio_fifo.o \
audioconvert.o \ audioconvert.o \
avstring.o \ avstring.o \
base64.o \ base64.o \

@ -0,0 +1,193 @@
/*
* Audio FIFO
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* 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;
}

@ -0,0 +1,146 @@
/*
* Audio FIFO
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* 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 or 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.
*
* @see enum AVSampleFormat
* The documentation for AVSampleFormat describes the data layout.
*
* @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.
*
* @see enum AVSampleFormat
* The documentation for AVSampleFormat describes the data layout.
*
* @param af AVAudioFifo to read from
* @param data audio data plane pointers
* @param nb_samples number of samples to read
* @return number of samples actually read, or negative AVERROR code
* on failure.
*/
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples);
/**
* Drain data from an AVAudioFifo.
*
* Removes the data without reading it.
*
* @param af AVAudioFifo to drain
* @param nb_samples number of samples to drain
* @return 0 if OK, or negative AVERROR code on failure
*/
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples);
/**
* Reset the AVAudioFifo buffer.
*
* This empties all data in the buffer.
*
* @param af AVAudioFifo to reset
*/
void av_audio_fifo_reset(AVAudioFifo *af);
/**
* Get the current number of samples in the AVAudioFifo available for reading.
*
* @param af the AVAudioFifo to query
* @return number of samples available for reading
*/
int av_audio_fifo_size(AVAudioFifo *af);
/**
* Get the current number of samples in the AVAudioFifo available for writing.
*
* @param af the AVAudioFifo to query
* @return number of samples available for writing
*/
int av_audio_fifo_space(AVAudioFifo *af);
/**
* @}
*/
#endif /* AVUTIL_AUDIO_FIFO_H */

@ -152,8 +152,8 @@
*/ */
#define LIBAVUTIL_VERSION_MAJOR 51 #define LIBAVUTIL_VERSION_MAJOR 51
#define LIBAVUTIL_VERSION_MINOR 27 #define LIBAVUTIL_VERSION_MINOR 28
#define LIBAVUTIL_VERSION_MICRO 2 #define LIBAVUTIL_VERSION_MICRO 0
#define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \ #define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
LIBAVUTIL_VERSION_MINOR, \ LIBAVUTIL_VERSION_MINOR, \

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