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-rw-r--r--libavdevice/alsa-audio-enc.c24
1 files changed, 19 insertions, 5 deletions
diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c
index bb4575f..403a44f 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
*/
@@ -38,8 +38,9 @@
*/
#include <alsa/asoundlib.h>
-#include "libavformat/avformat.h"
+#include "libavformat/internal.h"
+#include "avdevice.h"
#include "alsa-audio.h"
static av_cold int audio_write_header(AVFormatContext *s1)
@@ -61,6 +62,7 @@ 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;
@@ -101,6 +103,17 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
return 0;
}
+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 = s1->streams[0]->cur_dts - delay;
+}
+
AVOutputFormat ff_alsa_muxer = {
.name = "alsa",
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
@@ -110,5 +123,6 @@ AVOutputFormat ff_alsa_muxer = {
.write_header = audio_write_header,
.write_packet = audio_write_packet,
.write_trailer = ff_alsa_close,
+ .get_output_timestamp = audio_get_output_timestamp,
.flags = AVFMT_NOFILE,
};
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