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authorFrançois Revol <revol@free.fr>2003-03-27 15:08:16 +0000
committerFrançois Revol <revol@free.fr>2003-03-27 15:08:16 +0000
commit335338e88411c2f11960d597682b589af7f24024 (patch)
tree5e3e883d71941c3bf3db7a121ee18d41a0942409
parent788a2d3d8bcfff3c33aff24e99ea6599043825c3 (diff)
downloadffmpeg-335338e88411c2f11960d597682b589af7f24024.tar.gz
the media node now won't connect itself to the default audio input with -ad wait:
(allows redirecting another node to it) Originally committed as revision 1713 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavformat/beosaudio.cpp14
1 files changed, 10 insertions, 4 deletions
diff --git a/libavformat/beosaudio.cpp b/libavformat/beosaudio.cpp
index c34bb61826..2fe5e4c57e 100644
--- a/libavformat/beosaudio.cpp
+++ b/libavformat/beosaudio.cpp
@@ -181,7 +181,7 @@ static void audiorecord_callback(void *cookie, bigtime_t timestamp, void *buffer
}
#endif
-static int audio_open(AudioData *s, int is_output)
+static int audio_open(AudioData *s, int is_output, const char *audio_device)
{
int p[2];
int ret;
@@ -210,7 +210,13 @@ static int audio_open(AudioData *s, int is_output)
set_thread_priority(find_thread(NULL), B_DISPLAY_PRIORITY+1);
#ifdef HAVE_BSOUNDRECORDER
if (!is_output) {
- s->recorder = new BSoundRecorder(&iformat, false, "ffmpeg input", audiorecord_callback);
+ bool wait_for_input = false;
+ if (audio_device && !strcmp(audio_device, "wait:"))
+ wait_for_input = true;
+ s->recorder = new BSoundRecorder(&iformat, wait_for_input, "ffmpeg input", audiorecord_callback);
+ if (wait_for_input && (s->recorder->InitCheck() == B_OK)) {
+ s->recorder->WaitForIncomingConnection(&iformat);
+ }
if (s->recorder->InitCheck() != B_OK || iformat.format != media_raw_audio_format::B_AUDIO_SHORT) {
delete s->recorder;
s->recorder = NULL;
@@ -283,7 +289,7 @@ static int audio_write_header(AVFormatContext *s1)
st = s1->streams[0];
s->sample_rate = st->codec.sample_rate;
s->channels = st->codec.channels;
- ret = audio_open(s, 1);
+ ret = audio_open(s, 1, NULL);
if (ret < 0)
return -EIO;
return 0;
@@ -345,7 +351,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
s->sample_rate = ap->sample_rate;
s->channels = ap->channels;
- ret = audio_open(s, 0);
+ ret = audio_open(s, 0, ap->device);
if (ret < 0) {
av_free(st);
return -EIO;