Merge commit '8d26c193fb42d08602ac93ece039d4718d029adc'

* commit '8d26c193fb42d08602ac93ece039d4718d029adc':
  avdevice: Apply a more consistent file naming scheme

Conflicts:
	libavdevice/Makefile
	libavdevice/alsa.h
	libavdevice/alsa_dec.c
	libavdevice/alsa_enc.c
	libavdevice/sndio_enc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 168 insertions, 0 deletions

diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c
new file mode 100644
index 0000000000..286af650c7
--- /dev/null
+++ b/libavdevice/alsa_dec.c
@@ -0,0 +1,168 @@
+/*
+ * 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 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 <alsa/asoundlib.h>
+
+#include "libavutil/internal.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+
+#include "libavformat/internal.h"
+
+#include "avdevice.h"
+#include "alsa.h"
+
+static av_cold int audio_read_header(AVFormatContext *s1)
+{
+    AlsaData *s = s1->priv_data;
+    AVStream *st;
+    int ret;
+    enum AVCodecID codec_id;
+
+    st = avformat_new_stream(s1, NULL);
+    if (!st) {
+        av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
+
+        return AVERROR(ENOMEM);
+    }
+    codec_id    = s1->audio_codec_id;
+
+    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
+        &codec_id);
+    if (ret < 0) {
+        return AVERROR(EIO);
+    }
+
+    /* take real parameters */
+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
+    st->codec->codec_id    = codec_id;
+    st->codec->sample_rate = s->sample_rate;
+    st->codec->channels    = s->channels;
+    st->codec->frame_size = s->frame_size;
+    avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
+    /* microseconds instead of seconds, MHz instead of Hz */
+    s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
+                                      s->period_size, 1.5E-6);
+    if (!s->timefilter)
+        goto fail;
+
+    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;
+    int res;
+    int64_t dts;
+    snd_pcm_sframes_t delay = 0;
+
+    if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
+        return AVERROR(EIO);
+    }
+
+    while ((res = snd_pcm_readi(s->h, pkt->data, s->period_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);
+        }
+        ff_timefilter_reset(s->timefilter);
+    }
+
+    dts = av_gettime();
+    snd_pcm_delay(s->h, &delay);
+    dts -= av_rescale(delay + res, 1000000, s->sample_rate);
+    pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
+    s->last_period = res;
+
+    pkt->size = res * s->frame_size;
+
+    return 0;
+}
+
+static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
+{
+    return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
+}
+
+static const AVOption options[] = {
+    { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { "channels",    "", offsetof(AlsaData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { NULL },
+};
+
+static const AVClass alsa_demuxer_class = {
+    .class_name     = "ALSA demuxer",
+    .item_name      = av_default_item_name,
+    .option         = options,
+    .version        = LIBAVUTIL_VERSION_INT,
+    .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
+};
+
+AVInputFormat ff_alsa_demuxer = {
+    .name           = "alsa",
+    .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
+    .priv_data_size = sizeof(AlsaData),
+    .read_header    = audio_read_header,
+    .read_packet    = audio_read_packet,
+    .read_close     = ff_alsa_close,
+    .get_device_list = audio_get_device_list,
+    .flags          = AVFMT_NOFILE,
+    .priv_class     = &alsa_demuxer_class,
+};