1 files changed, 117 insertions, 0 deletions
diff --git a/libavdevice/alsa_enc.c b/libavdevice/alsa_enc.c
new file mode 100644
index 0000000000..30c1719203
--- /dev/null
+++ b/libavdevice/alsa_enc.c
@@ -0,0 +1,117 @@
+/*
+ * ALSA input and output
+ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
+ * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
+ *
+ * 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
+ * ALSA input and output: output
+ * @author Luca Abeni ( lucabe72 email it )
+ * @author Benoit Fouet ( benoit fouet free fr )
+ *
+ * This avdevice encoder allows to play audio to an ALSA (Advanced Linux
+ * Sound Architecture) device.
+ *
+ * The filename parameter is the name of an ALSA PCM device capable of
+ * capture, for example "default" or "plughw:1"; see the ALSA documentation
+ * for naming conventions. The empty string is equivalent to "default".
+ *
+ * The playback period is set to the lower value available for the device,
+ * which gives a low latency suitable for real-time playback.
+ */
+
+#include <alsa/asoundlib.h>
+
+#include "libavutil/internal.h"
+
+#include "libavformat/avformat.h"
+
+#include "alsa.h"
+
+static av_cold int audio_write_header(AVFormatContext *s1)
+{
+    AlsaData *s = s1->priv_data;
+    AVStream *st;
+    unsigned int sample_rate;
+    enum AVCodecID codec_id;
+    int res;
+
+    st = s1->streams[0];
+    sample_rate = st->codec->sample_rate;
+    codec_id    = st->codec->codec_id;
+    res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
+        st->codec->channels, &codec_id);
+    if (sample_rate != st->codec->sample_rate) {
+        av_log(s1, AV_LOG_ERROR,
+               "sample rate %d not available, nearest is %d\n",
+               st->codec->sample_rate, sample_rate);
+        goto fail;
+    }
+
+    return res;
+
+fail:
+    snd_pcm_close(s->h);
+    return AVERROR(EIO);
+}
+
+static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+    AlsaData *s = s1->priv_data;
+    int res;
+    int size     = pkt->size;
+    uint8_t *buf = pkt->data;
+
+    size /= s->frame_size;
+    if (s->reorder_func) {
+        if (size > s->reorder_buf_size)
+            if (ff_alsa_extend_reorder_buf(s, size))
+                return AVERROR(ENOMEM);
+        s->reorder_func(buf, s->reorder_buf, size);
+        buf = s->reorder_buf;
+    }
+    while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
+        if (res == -EAGAIN) {
+
+            return AVERROR(EAGAIN);
+        }
+
+        if (ff_alsa_xrun_recover(s1, res) < 0) {
+            av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
+                   snd_strerror(res));
+
+            return AVERROR(EIO);
+        }
+    }
+
+    return 0;
+}
+
+AVOutputFormat ff_alsa_muxer = {
+    .name           = "alsa",
+    .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
+    .priv_data_size = sizeof(AlsaData),
+    .audio_codec    = DEFAULT_CODEC_ID,
+    .video_codec    = AV_CODEC_ID_NONE,
+    .write_header   = audio_write_header,
+    .write_packet   = audio_write_packet,
+    .write_trailer  = ff_alsa_close,
+    .flags          = AVFMT_NOFILE,
+};