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
Add avfilter_graph_parse2(), avfilter_inout_alloc() and
avfilter_inout_free() functions.
HEADERS = adler32.h \
aes.h \
attributes.h \
+ audio_fifo.h \
audioconvert.h \
avassert.h \
avstring.h \
OBJS = adler32.o \
aes.o \
+ audio_fifo.o \
audioconvert.o \
avstring.o \
base64.o \
--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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 */
*/
#define LIBAVUTIL_VERSION_MAJOR 51
-#define LIBAVUTIL_VERSION_MINOR 27
-#define LIBAVUTIL_VERSION_MICRO 2
+#define LIBAVUTIL_VERSION_MINOR 28
+#define LIBAVUTIL_VERSION_MICRO 0
#define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
LIBAVUTIL_VERSION_MINOR, \
projects / platform / upstream / libav.git / commitdiff