avutil: add audio fifo buffer

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

1 files changed, 193 insertions, 0 deletions

diff --git a/libavutil/audio_fifo.c b/libavutil/audio_fifo.c
new file mode 100644
index 0000000000..97c51a72c1
--- /dev/null
+++ b/libavutil/audio_fifo.c
@@ -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;
+}