1 files changed, 63 insertions, 6 deletions

diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c
index bb4575fa02..43d097de53 100644
--- a/libavdevice/alsa-audio-enc.c
+++ b/libavdevice/alsa-audio-enc.c
@@ -3,20 +3,20 @@
  * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
  * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * 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.
  *
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -38,19 +38,26 @@
  */
 
 #include <alsa/asoundlib.h>
-#include "libavformat/avformat.h"
 
+#include "libavutil/time.h"
+#include "libavformat/internal.h"
+#include "avdevice.h"
 #include "alsa-audio.h"
 
 static av_cold int audio_write_header(AVFormatContext *s1)
 {
     AlsaData *s = s1->priv_data;
-    AVStream *st;
+    AVStream *st = NULL;
     unsigned int sample_rate;
     enum AVCodecID codec_id;
     int res;
 
+    if (s1->nb_streams != 1 || s1->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
+        av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
+        return AVERROR(EINVAL);
+    }
     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,
@@ -61,6 +68,7 @@ static av_cold int audio_write_header(AVFormatContext *s1)
                st->codec->sample_rate, sample_rate);
         goto fail;
     }
+    avpriv_set_pts_info(st, 64, 1, sample_rate);
 
     return res;
 
@@ -77,6 +85,10 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
     uint8_t *buf = pkt->data;
 
     size /= s->frame_size;
+    if (pkt->dts != AV_NOPTS_VALUE)
+        s->timestamp = pkt->dts;
+    s->timestamp += pkt->duration ? pkt->duration : size;
+
     if (s->reorder_func) {
         if (size > s->reorder_buf_size)
             if (ff_alsa_extend_reorder_buf(s, size))
@@ -101,6 +113,47 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
     return 0;
 }
 
+static int audio_write_frame(AVFormatContext *s1, int stream_index,
+                             AVFrame **frame, unsigned flags)
+{
+    AlsaData *s = s1->priv_data;
+    AVPacket pkt;
+
+    /* ff_alsa_open() should have accepted only supported formats */
+    if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
+        return av_sample_fmt_is_planar(s1->streams[stream_index]->codec->sample_fmt) ?
+               AVERROR(EINVAL) : 0;
+    /* set only used fields */
+    pkt.data     = (*frame)->data[0];
+    pkt.size     = (*frame)->nb_samples * s->frame_size;
+    pkt.dts      = (*frame)->pkt_dts;
+    pkt.duration = av_frame_get_pkt_duration(*frame);
+    return audio_write_packet(s1, &pkt);
+}
+
+static void
+audio_get_output_timestamp(AVFormatContext *s1, int stream,
+    int64_t *dts, int64_t *wall)
+{
+    AlsaData *s  = s1->priv_data;
+    snd_pcm_sframes_t delay = 0;
+    *wall = av_gettime();
+    snd_pcm_delay(s->h, &delay);
+    *dts = s->timestamp - delay;
+}
+
+static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
+{
+    return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
+}
+
+static const AVClass alsa_muxer_class = {
+    .class_name     = "ALSA muxer",
+    .item_name      = av_default_item_name,
+    .version        = LIBAVUTIL_VERSION_INT,
+    .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
+};
+
 AVOutputFormat ff_alsa_muxer = {
     .name           = "alsa",
     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
@@ -110,5 +163,9 @@ AVOutputFormat ff_alsa_muxer = {
     .write_header   = audio_write_header,
     .write_packet   = audio_write_packet,
     .write_trailer  = ff_alsa_close,
+    .write_uncoded_frame = audio_write_frame,
+    .get_device_list = audio_get_device_list,
+    .get_output_timestamp = audio_get_output_timestamp,
     .flags          = AVFMT_NOFILE,
+    .priv_class     = &alsa_muxer_class,
 };