[libav-commits] avutil: add audio fifo buffer

Justin Ruggles git at libav.org
Fri Apr 20 15:55:38 CEST 2012


Module: libav
Branch: master
Commit: 0c0d1bce7c582b82e49843acaa7d0fb4b1774b21

Author:    Justin Ruggles <justin.ruggles at gmail.com>
Committer: Justin Ruggles <justin.ruggles at gmail.com>
Date:      Fri Mar 23 17:45:48 2012 -0400

avutil: add audio fifo buffer

The functions operate on the sample level rather than the byte level and work
with all audio sample formats.

---

 doc/APIchanges         |   12 +++
 libavutil/Makefile     |    2 +
 libavutil/audio_fifo.c |  193 ++++++++++++++++++++++++++++++++++++++++++++++++
 libavutil/audio_fifo.h |  146 ++++++++++++++++++++++++++++++++++++
 libavutil/avutil.h     |    4 +-
 5 files changed, 355 insertions(+), 2 deletions(-)

diff --git a/doc/APIchanges b/doc/APIchanges
index 5114e14..fed77b0 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -12,6 +12,18 @@ libavutil:   2011-04-18
 
 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.
diff --git a/libavutil/Makefile b/libavutil/Makefile
index 820abb1..69f2acd 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..97c51a7
--- /dev/null
+++ b/libavutil/audio_fifo.c
@@ -0,0 +1,193 @@
+/*
+ * Audio FIFO
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles at 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;
+}
diff --git a/libavutil/audio_fifo.h b/libavutil/audio_fifo.h
new file mode 100644
index 0000000..8c76388
--- /dev/null
+++ b/libavutil/audio_fifo.h
@@ -0,0 +1,146 @@
+/*
+ * Audio FIFO
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles at 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 */
diff --git a/libavutil/avutil.h b/libavutil/avutil.h
index 7319c22..6673f0f 100644
--- a/libavutil/avutil.h
+++ b/libavutil/avutil.h
@@ -152,8 +152,8 @@
  */
 
 #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, \



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