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, 175 insertions, 0 deletions

diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c
new file mode 100644
index 0000000000..1be0a6c7eb
--- /dev/null
+++ b/libavdevice/alsa-audio-dec.c
@@ -0,0 +1,175 @@
+/*
+ * 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-dec.c
+ * ALSA input and output: input
+ * @author Luca Abeni ( lucabe72 email it )
+ * @author Benoit Fouet ( benoit fouet free fr )
+ * @author Nicolas George ( nicolas george normalesup org )
+ *
+ * This avdevice decoder allows to capture audio from 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 capture period is set to the lower value available for the device,
+ * which gives a low latency suitable for real-time capture.
+ *
+ * The PTS are an Unix time in microsecond.
+ *
+ * Due to a bug in the ALSA library
+ * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
+ * decoder does not work with certain ALSA plugins, especially the dsnoop
+ * plugin.
+ */
+
+#include "libavformat/avformat.h"
+#include <alsa/asoundlib.h>
+
+#include "alsa-audio.h"
+
+av_cold static int audio_read_header(AVFormatContext *s1,
+                                     AVFormatParameters *ap)
+{
+    AlsaData *s = s1->priv_data;
+    AVStream *st;
+    int ret;
+    unsigned int sample_rate;
+    int codec_id;
+    snd_pcm_sw_params_t *sw_params;
+
+    if (ap->sample_rate <= 0) {
+        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
+
+        return AVERROR(EIO);
+    }
+
+    if (ap->channels <= 0) {
+        av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
+
+        return AVERROR(EIO);
+    }
+
+    st = av_new_stream(s1, 0);
+    if (!st) {
+        av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
+
+        return AVERROR(ENOMEM);
+    }
+    sample_rate = ap->sample_rate;
+    codec_id    = ap->audio_codec_id;
+
+    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
+        &codec_id);
+    if (ret < 0) {
+        return AVERROR(EIO);
+    }
+
+    if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
+        av_log(s1, AV_LOG_WARNING,
+               "capture with some ALSA plugins, especially dsnoop, "
+               "may hang.\n");
+
+    ret = snd_pcm_sw_params_malloc(&sw_params);
+    if (ret < 0) {
+        av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
+               snd_strerror(ret));
+        goto fail;
+    }
+
+    snd_pcm_sw_params_current(s->h, sw_params);
+    snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
+
+    ret = snd_pcm_sw_params(s->h, sw_params);
+    snd_pcm_sw_params_free(sw_params);
+    if (ret < 0) {
+        av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
+               snd_strerror(ret));
+        goto fail;
+    }
+
+    /* take real parameters */
+    st->codec->codec_type  = CODEC_TYPE_AUDIO;
+    st->codec->codec_id    = codec_id;
+    st->codec->sample_rate = sample_rate;
+    st->codec->channels    = ap->channels;
+    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
+
+    return 0;
+
+fail:
+    snd_pcm_close(s->h);
+    return AVERROR(EIO);
+}
+
+static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+    AlsaData *s  = s1->priv_data;
+    AVStream *st = s1->streams[0];
+    int res;
+    snd_htimestamp_t timestamp;
+    snd_pcm_uframes_t ts_delay;
+
+    if (av_new_packet(pkt, s->period_size) < 0) {
+        return AVERROR(EIO);
+    }
+
+    while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
+        if (res == -EAGAIN) {
+            av_free_packet(pkt);
+
+            return AVERROR(EAGAIN);
+        }
+        if (ff_alsa_xrun_recover(s1, res) < 0) {
+            av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
+                   snd_strerror(res));
+            av_free_packet(pkt);
+
+            return AVERROR(EIO);
+        }
+    }
+
+    snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
+    ts_delay += res;
+    pkt->pts = timestamp.tv_sec * 1000000LL
+               + (timestamp.tv_nsec * st->codec->sample_rate
+                  - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
+               / (st->codec->sample_rate * 1000LL);
+
+    pkt->size = res * s->frame_size;
+
+    return 0;
+}
+
+AVInputFormat alsa_demuxer = {
+    "alsa",
+    NULL_IF_CONFIG_SMALL("ALSA audio input"),
+    sizeof(AlsaData),
+    NULL,
+    audio_read_header,
+    audio_read_packet,
+    ff_alsa_close,
+    .flags = AVFMT_NOFILE,
+};