Add ALSA support in libavdevice.

Patch by Nicolas George: name surname normalesup org
Original thread: [FFmpeg-devel] [PATCH] ALSA for libavdevice
Date: 12/09/2008 07:17 PM

Originally committed as revision 16800 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 108 insertions, 0 deletions

diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c
new file mode 100644
index 0000000000..375ad63765
--- /dev/null
+++ b/libavdevice/alsa-audio-enc.c
@@ -0,0 +1,108 @@
+/*
+ * 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 FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file alsa-audio-enc.c
+ * 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 "libavformat/avformat.h"
+#include <alsa/asoundlib.h>
+
+#include "alsa-audio.h"
+
+av_cold static int audio_write_header(AVFormatContext *s1)
+{
+    AlsaData *s = s1->priv_data;
+    AVStream *st;
+    unsigned int sample_rate;
+    int 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;
+
+    while((res = snd_pcm_writei(s->h, buf, size / s->frame_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 alsa_muxer = {
+    "alsa",
+    NULL_IF_CONFIG_SMALL("ALSA audio output"),
+    "",
+    "",
+    sizeof(AlsaData),
+    DEFAULT_CODEC_ID,
+    CODEC_ID_NONE,
+    audio_write_header,
+    audio_write_packet,
+    ff_alsa_close,
+    .flags = AVFMT_NOFILE,
+};