Problem with scaletemo element
Maik Scholz
Scholz.Maik at t-online.de
Wed Apr 18 15:09:50 UTC 2018
Hi,
I like to change the playback speed of an audio pipeline.
For that, I wrote an example which should change the playback speed to the
ratio to 0.5 in the segment (T=11s => T=16s)
The audio samples are generated from an app source. This generates a 300Hz
sine tone with a tick every 500ms.
My seek event:
{
gboolean update = TRUE;
GstSegment segment;
gst_segment_init (&segment, GST_FORMAT_TIME);
printf("gst_segment_do_seek ...\n");
if( gst_segment_do_seek (&segment,
0.5, GST_FORMAT_TIME,
GST_SEEK_FLAG_NONE,
GST_SEEK_TYPE_SET, pos + 1*GST_SECOND,
GST_SEEK_TYPE_SET, pos + 1*GST_SECOND + 5*GST_SECOND,
&update)) {
GstEvent *ev = gst_event_new_segment (&segment);
if( ev != NULL ) {
if( !gst_element_send_event(data->audio_scaletempo,ev) ) {
}
}
}
}
Instead of a changed playback speed, I get a complete mute. I don't know
why?
Any hint?
Maik
My test:
gcc -o gstseektest gstseektest.c `pkg-config --cflags --libs gstreamer-1.0
gstreamer-app-1.0 gstreamer-audio-1.0` -lm
./gstseektest
0:00:00.017054010 32199 0xbd6430 INFO GST_EVENT
gstevent.c:1511:gst_event_new_reconfigure: creating reconfigure event
0:00:00.017461765 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbbd2e0 reconfigure
61441
0:00:00.017503466 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<sink:proxypad0> have event type
reconfigure event: 0xbbd2e0, time 99:99:99.999999999, seq-num 0, (NULL)
0:00:00.017778341 32199 0xbd6430 INFO GST_EVENT
gstevent.c:1511:gst_event_new_reconfigure: creating reconfigure event
0:00:00.017830456 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbbd350 reconfigure
61441
0:00:00.017843365 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_source:src> have event
type reconfigure event: 0xbbd350, time 99:99:99.999999999, seq-num 2, (NULL)
0:00:00.017855729 32199 0xbd6430 INFO GST_EVENT
gstpad.c:5652:gst_pad_send_event_unchecked:<audio_source:src> Received event
on flushing pad. Discarding
0:00:00.017984762 32199 0xbd6430 INFO GST_EVENT
gstevent.c:1511:gst_event_new_reconfigure: creating reconfigure event
0:00:00.018048776 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbcb000 reconfigure
61441
0:00:00.018061161 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_queue:src> have event type
reconfigure event: 0xbcb000, time 99:99:99.999999999, seq-num 5, (NULL)
0:00:00.018094548 32199 0xbd6430 INFO GST_EVENT
gstpad.c:5652:gst_pad_send_event_unchecked:<audio_queue:src> Received event
on flushing pad. Discarding
0:00:00.018698948 32199 0xbd6430 INFO GST_EVENT
gstevent.c:1511:gst_event_new_reconfigure: creating reconfigure event
0:00:00.018768177 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbcb070 reconfigure
61441
0:00:00.018781683 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_scaletempo:src> have event
type reconfigure event: 0xbcb070, time 99:99:99.999999999, seq-num 8, (NULL)
0:00:00.018813472 32199 0xbd6430 INFO GST_EVENT
gstpad.c:5652:gst_pad_send_event_unchecked:<audio_scaletempo:src> Received
event on flushing pad. Discarding
0:00:00.019507583 32199 0xbd6430 INFO GST_EVENT
gstevent.c:1511:gst_event_new_reconfigure: creating reconfigure event
0:00:00.019554719 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbcb0e0 reconfigure
61441
0:00:00.019623702 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_convert1:src> have event
type reconfigure event: 0xbcb0e0, time 99:99:99.999999999, seq-num 11,
(NULL)
0:00:00.019655183 32199 0xbd6430 INFO GST_EVENT
gstpad.c:5652:gst_pad_send_event_unchecked:<audio_convert1:src> Received
event on flushing pad. Discarding
0:00:00.019702936 32199 0xbd6430 INFO GST_EVENT
gstevent.c:1511:gst_event_new_reconfigure: creating reconfigure event
0:00:00.019729220 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbcb150 reconfigure
61441
0:00:00.019778134 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_resample:src> have event
type reconfigure event: 0xbcb150, time 99:99:99.999999999, seq-num 14,
(NULL)
0:00:00.019789796 32199 0xbd6430 INFO GST_EVENT
gstpad.c:5652:gst_pad_send_event_unchecked:<audio_resample:src> Received
event on flushing pad. Discarding
0:00:00.039213483 32199 0xbd6430 INFO GST_EVENT
gstevent.c:1511:gst_event_new_reconfigure: creating reconfigure event
0:00:00.039277891 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbcb2a0 reconfigure
61441
0:00:00.039295173 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<sink:proxypad0> have event type
reconfigure event: 0xbcb2a0, time 99:99:99.999999999, seq-num 18, (NULL)
0:00:00.039957859 32199 0xbcf320 FIXME default
gstutils.c:3826:gst_pad_create_stream_id_internal:<audio_source:src>
Creating random stream-id, consider implementing a deterministic way of
creating a stream-id
0:00:00.040056874 32199 0xbcf320 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbde0f0
stream-start 10254
0:00:00.040167590 32199 0xbcf320 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_queue:sink> have event
type stream-start event: 0xbde0f0, time 99:99:99.999999999, seq-num 40,
GstEventStreamStart, stream-id=(string)964c8ceeaaf0f109aecdf3a51767d017,
flags=(GstStreamFlags)GST_STREAM_FLAG_NONE, group-id=(uint)0;
0:00:00.040282212 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_scaletempo:sink> have
event type stream-start event: 0xbde0f0, time 99:99:99.999999999, seq-num
40, GstEventStreamStart, stream-id=(string)964c8ceeaaf0f109aecdf3a51767d017,
flags=(GstStreamFlags)GST_STREAM_FLAG_NONE, group-id=(uint)0;
0:00:00.040360015 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_convert1:sink> have event
type stream-start event: 0xbde0f0, time 99:99:99.999999999, seq-num 40,
GstEventStreamStart, stream-id=(string)964c8ceeaaf0f109aecdf3a51767d017,
flags=(GstStreamFlags)GST_STREAM_FLAG_NONE, group-id=(uint)0;
0:00:00.040377651 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_resample:sink> have event
type stream-start event: 0xbde0f0, time 99:99:99.999999999, seq-num 40,
GstEventStreamStart, stream-id=(string)964c8ceeaaf0f109aecdf3a51767d017,
flags=(GstStreamFlags)GST_STREAM_FLAG_NONE, group-id=(uint)0;
0:00:00.040414557 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink:sink> have event type
stream-start event: 0xbde0f0, time 99:99:99.999999999, seq-num 40,
GstEventStreamStart, stream-id=(string)964c8ceeaaf0f109aecdf3a51767d017,
flags=(GstStreamFlags)GST_STREAM_FLAG_NONE, group-id=(uint)0;
0:00:00.040442030 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink-actual-sink-alsa:sink>
have event type stream-start event: 0xbde0f0, time 99:99:99.999999999,
seq-num 40, GstEventStreamStart,
stream-id=(string)964c8ceeaaf0f109aecdf3a51767d017,
flags=(GstStreamFlags)GST_STREAM_FLAG_NONE, group-id=(uint)0;
0:00:00.040535589 32199 0xbcf320 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps: creating caps event audio/x-raw,
format=(string)S16LE, layout=(string)interleaved, rate=(int)44100,
channels=(int)1
0:00:00.040553150 32199 0xbcf320 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbde160 caps 12814
0:00:00.040568920 32199 0xbcf320 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_queue:sink> have event
type caps event: 0xbde160, time 99:99:99.999999999, seq-num 44,
GstEventCaps, caps=(GstCaps)"audio/x-raw\,\ format\=\(string\)S16LE\,\
layout\=\(string\)interleaved\,\ rate\=\(int\)44100\,\ channels\=\(int\)1";
0:00:00.040749803 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_scaletempo:sink> have
event type caps event: 0xbde160, time 99:99:99.999999999, seq-num 44,
GstEventCaps, caps=(GstCaps)"audio/x-raw\,\ format\=\(string\)S16LE\,\
layout\=\(string\)interleaved\,\ rate\=\(int\)44100\,\ channels\=\(int\)1";
0:00:00.040906240 32199 0xbcf4a0 DEBUG scaletempo
gstscaletempo.c:679:gst_scaletempo_set_caps: caps: audio/x-raw,
format=(string)S16LE, layout=(string)interleaved, rate=(int)44100,
channels=(int)1, 2 bps
0:00:00.040950315 32199 0xbcf4a0 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps: creating caps event audio/x-raw,
format=(string)S16LE, layout=(string)interleaved, rate=(int)44100,
channels=(int)1
0:00:00.040971495 32199 0xbcf4a0 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0x7fad54004410 caps
12814
0:00:00.040990706 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_convert1:sink> have event
type caps event: 0x7fad54004410, time 99:99:99.999999999, seq-num 45,
GstEventCaps, caps=(GstCaps)"audio/x-raw\,\ format\=\(string\)S16LE\,\
layout\=\(string\)interleaved\,\ rate\=\(int\)44100\,\ channels\=\(int\)1";
0:00:00.041585263 32199 0xbcf4a0 WARN alsa
conf.c:4694:snd_config_expand: alsalib error: Unknown parameters {AES0 0x02
AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.041608410 32199 0xbcf4a0 WARN alsa
pcm.c:2239:snd_pcm_open_noupdate: alsalib error: Unknown PCM default:{AES0
0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.040931204 32199 0xbcf320 INFO GST_EVENT
gstevent.c:889:gst_event_new_segment: creating segment event time segment
start=0:00:00.000000000, offset=0:00:00.000000000, stop=99:99:99.999999999,
rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000,
base=0:00:00.000000000, position 0:00:00.000000000, duration
99:99:99.999999999
0:00:00.042804712 32199 0xbcf320 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbde1d0 segment
17934
0:00:00.042823820 32199 0xbcf320 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_queue:sink> have event
type segment event: 0xbde1d0, time 99:99:99.999999999, seq-num 35,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, base=(guint64)0,
offset=(guint64)0, start=(guint64)0, stop=(guint64)18446744073709551615,
time=(guint64)0, position=(guint64)0,
duration=(guint64)18446744073709551615;";
0:00:00.049389403 32199 0xbcf4a0 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps: creating caps event audio/x-raw,
format=(string)S16LE, layout=(string)interleaved, rate=(int)44100,
channels=(int)1
0:00:00.049533091 32199 0xbcf4a0 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0x7fad540044f0 caps
12814
0:00:00.049624092 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_resample:sink> have event
type caps event: 0x7fad540044f0, time 99:99:99.999999999, seq-num 48,
GstEventCaps, caps=(GstCaps)"audio/x-raw\,\ format\=\(string\)S16LE\,\
layout\=\(string\)interleaved\,\ rate\=\(int\)44100\,\ channels\=\(int\)1";
0:00:00.052439365 32199 0xbcf4a0 WARN audio-resampler
audio-resampler.c:273:convert_taps_gint16_c: can't find exact taps
0:00:00.052530887 32199 0xbcf4a0 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps: creating caps event audio/x-raw,
format=(string)S16LE, layout=(string)interleaved, rate=(int)44100,
channels=(int)1
0:00:00.052554080 32199 0xbcf4a0 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0x7fad54004640 caps
12814
0:00:00.052610405 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink:sink> have event type
caps event: 0x7fad54004640, time 99:99:99.999999999, seq-num 49,
GstEventCaps, caps=(GstCaps)"audio/x-raw\,\ format\=\(string\)S16LE\,\
layout\=\(string\)interleaved\,\ rate\=\(int\)44100\,\ channels\=\(int\)1";
0:00:00.052702883 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink-actual-sink-alsa:sink>
have event type caps event: 0x7fad54004640, time 99:99:99.999999999, seq-num
49, GstEventCaps, caps=(GstCaps)"audio/x-raw\,\ format\=\(string\)S16LE\,\
layout\=\(string\)interleaved\,\ rate\=\(int\)44100\,\ channels\=\(int\)1";
0:00:00.062901170 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_scaletempo:sink> have
event type segment event: 0xbde1d0, time 99:99:99.999999999, seq-num 35,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, base=(guint64)0,
offset=(guint64)0, start=(guint64)0, stop=(guint64)18446744073709551615,
time=(guint64)0, position=(guint64)0,
duration=(guint64)18446744073709551615;";
0:00:00.062966702 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_convert1:sink> have event
type segment event: 0xbde1d0, time 99:99:99.999999999, seq-num 35,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, base=(guint64)0,
offset=(guint64)0, start=(guint64)0, stop=(guint64)18446744073709551615,
time=(guint64)0, position=(guint64)0,
duration=(guint64)18446744073709551615;";
0:00:00.063005484 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_resample:sink> have event
type segment event: 0xbde1d0, time 99:99:99.999999999, seq-num 35,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, base=(guint64)0,
offset=(guint64)0, start=(guint64)0, stop=(guint64)18446744073709551615,
time=(guint64)0, position=(guint64)0,
duration=(guint64)18446744073709551615;";
0:00:00.063042013 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink:sink> have event type
segment event: 0xbde1d0, time 99:99:99.999999999, seq-num 35,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, base=(guint64)0,
offset=(guint64)0, start=(guint64)0, stop=(guint64)18446744073709551615,
time=(guint64)0, position=(guint64)0,
duration=(guint64)18446744073709551615;";
0:00:00.063078654 32199 0xbcf4a0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink-actual-sink-alsa:sink>
have event type segment event: 0xbde1d0, time 99:99:99.999999999, seq-num
35, GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, base=(guint64)0,
offset=(guint64)0, start=(guint64)0, stop=(guint64)18446744073709551615,
time=(guint64)0, position=(guint64)0,
duration=(guint64)18446744073709551615;";
0:00:00.063479840 32199 0x7fad54003cf0 DEBUG scaletempo
gstscaletempo.c:747:gst_scaletempo_query:<audio_scaletempo> Peer latency:
min 0:00:00.000000000 max 0:01:40.000000000
0:00:00.063503013 32199 0x7fad54003cf0 DEBUG scaletempo
gstscaletempo.c:751:gst_scaletempo_query:<audio_scaletempo> Our latency:
0:00:00.000000000
0:00:00.063515248 32199 0x7fad54003cf0 DEBUG scaletempo
gstscaletempo.c:758:gst_scaletempo_query:<audio_scaletempo> Calculated total
latency : min 0:00:00.000000000 max 0:01:40.000000000
0:00:00.063547392 32199 0x7fad54003cf0 INFO GST_EVENT
gstevent.c:1382:gst_event_new_latency: creating latency event
0:00:00.000000000
0:00:00.063564230 32199 0x7fad54003cf0 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0x7fad54004ac0
latency 56321
0:00:00.063587530 32199 0x7fad54003cf0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<sink:proxypad0> have event type
latency event: 0x7fad54004ac0, time 99:99:99.999999999, seq-num 59,
GstEventLatency, latency=(guint64)0;
0:00:00.063626992 32199 0x7fad54003cf0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_resample:src> have event
type latency event: 0x7fad54004ac0, time 99:99:99.999999999, seq-num 59,
GstEventLatency, latency=(guint64)0;
0:00:00.063642462 32199 0x7fad54003cf0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_convert1:src> have event
type latency event: 0x7fad54004ac0, time 99:99:99.999999999, seq-num 59,
GstEventLatency, latency=(guint64)0;
0:00:00.063656774 32199 0x7fad54003cf0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_scaletempo:src> have event
type latency event: 0x7fad54004ac0, time 99:99:99.999999999, seq-num 59,
GstEventLatency, latency=(guint64)0;
0:00:00.063670622 32199 0x7fad54003cf0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_queue:src> have event type
latency event: 0x7fad54004ac0, time 99:99:99.999999999, seq-num 59,
GstEventLatency, latency=(guint64)0;
0:00:00.063686127 32199 0x7fad54003cf0 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_source:src> have event
type latency event: 0x7fad54004ac0, time 99:99:99.999999999, seq-num 59,
GstEventLatency, latency=(guint64)0;
gst_segment_do_seek ...
0:00:10.022508572 32199 0xbd6430 INFO GST_EVENT
gstevent.c:889:gst_event_new_segment: creating segment event time segment
start=0:00:11.134149659, offset=0:00:00.000000000, stop=0:00:16.134149659,
rate=0.500000, applied_rate=1.000000, flags=0x00, time=0:00:11.134149659,
base=0:00:00.000000000, position 0:00:11.134149659, duration
99:99:99.999999999
0:00:10.022536518 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbcb310 segment
17934
0:00:10.022549032 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_scaletempo:sink> have
event type segment event: 0xbcb310, time 99:99:99.999999999, seq-num 68,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)0.5,
applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, base=(guint64)0,
offset=(guint64)0, start=(guint64)11134149659, stop=(guint64)16134149659,
time=(guint64)11134149659, position=(guint64)11134149659,
duration=(guint64)18446744073709551615;";
0:00:10.131871256 32199 0xbd6430 DEBUG scaletempo
gstscaletempo.c:620:gst_scaletempo_sink_event: 0.500 scale, 0.000 stride_in,
0 stride_out
0:00:10.131914177 32199 0xbd6430 INFO GST_EVENT
gstevent.c:889:gst_event_new_segment: creating segment event time segment
start=0:00:11.134149659, offset=0:00:00.000000000, stop=0:00:21.134149659,
rate=1.000000, applied_rate=0.500000, flags=0x00, time=0:00:11.134149659,
base=0:00:00.000000000, position 0:00:11.134149659, duration
99:99:99.999999999
0:00:10.131942226 32199 0xbd6430 DEBUG GST_EVENT
gstevent.c:305:gst_event_new_custom: creating new event 0xbde010 segment
17934
0:00:10.131959883 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_convert1:sink> have event
type segment event: 0xbde010, time 99:99:99.999999999, seq-num 68,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)0.5, format=(GstFormat)GST_FORMAT_TIME,
base=(guint64)0, offset=(guint64)0, start=(guint64)11134149659,
stop=(guint64)21134149659, time=(guint64)11134149659,
position=(guint64)11134149659, duration=(guint64)18446744073709551615;";
0:00:10.132050951 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_resample:sink> have event
type segment event: 0xbde010, time 99:99:99.999999999, seq-num 68,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)0.5, format=(GstFormat)GST_FORMAT_TIME,
base=(guint64)0, offset=(guint64)0, start=(guint64)11134149659,
stop=(guint64)21134149659, time=(guint64)11134149659,
position=(guint64)11134149659, duration=(guint64)18446744073709551615;";
0:00:10.132102178 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink:sink> have event type
segment event: 0xbde010, time 99:99:99.999999999, seq-num 68,
GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)0.5, format=(GstFormat)GST_FORMAT_TIME,
base=(guint64)0, offset=(guint64)0, start=(guint64)11134149659,
stop=(guint64)21134149659, time=(guint64)11134149659,
position=(guint64)11134149659, duration=(guint64)18446744073709551615;";
0:00:10.132149719 32199 0xbd6430 DEBUG GST_EVENT
gstpad.c:5546:gst_pad_send_event_unchecked:<audio_sink-actual-sink-alsa:sink>
have event type segment event: 0xbde010, time 99:99:99.999999999, seq-num
68, GstEventSegment, segment=(GstSegment)"GstSegment,
flags=(GstSegmentFlags)GST_SEGMENT_FLAG_NONE, rate=(double)1,
applied-rate=(double)0.5, format=(GstFormat)GST_FORMAT_TIME,
base=(guint64)0, offset=(guint64)0, start=(guint64)11134149659,
stop=(guint64)21134149659, time=(guint64)11134149659,
position=(guint64)11134149659, duration=(guint64)18446744073709551615;";
0:00:10.133330285 32199 0xbcf4a0 DEBUG scaletempo
gstscaletempo.c:424:reinit_buffers: 0.500 scale, 661.500 stride_in, 1323
stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, S16LE mode
0:00:11.095553961 32199 0xbcf4a0 WARN audiobasesink
gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<audio_sink-actual-sink-alsa>
Unexpected discontinuity in audio timestamps of -0:00:10.698526077,
resyncing
0:00:11.096069902 32199 0xbcf4a0 WARN audiobasesink
gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<audio_sink-actual-sink-alsa>
Unexpected discontinuity in audio timestamps of +0:00:00.002879818,
resyncing
0:00:11.097452921 32199 0xbcf4a0 WARN audiobasesink
gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<audio_sink-actual-sink-alsa>
Unexpected discontinuity in audio timestamps of +0:00:00.001859410,
resyncing
0:00:11.100180015 32199 0xbcf4a0 WARN audiobasesink
gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<audio_sink-actual-sink-alsa>
Unexpected discontinuity in audio timestamps of +0:00:00.000839002,
resyncing
0:00:11.120734946 32199 0xbcf4a0 WARN audiobasesink
gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<audio_sink-actual-sink-alsa>
Unexpected discontinuity in audio timestamps of +0:00:00.000113378,
resyncing
=> audio is muted?
My example code:
/* compile with:
* gcc -o gstseektest gstseektest.c `pkg-config --cflags --libs
gstreamer-1.0 gstreamer-app-1.0 gstreamer-audio-1.0` -lm
*/
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer
*/
#define SAMPLE_RATE 44100 /* Samples per second we are sending */
/* Structure to contain all our information, so we can pass it to callbacks
*/
typedef struct _CustomData {
GstElement *pipeline, *app_source, *audio_queue, *audio_scaletempo,
*audio_convert1, *audio_resample, *audio_sink;
guint64 num_samples; /* Number of samples generated so far (for
timestamp generation) */
guint sourceid; /* To control the GSource */
guint timeoutid;
GMainLoop *main_loop; /* GLib's Main Loop */
FILE* sink_pad_dumpfile;
} CustomData;
/* PAD probe for sink data capturing
*/
GstPadProbeReturn
sink_pad_probe (GstPad *pad, GstPadProbeInfo *info, gpointer user_data)
{
CustomData *data = (CustomData*)user_data;
if(((info->type&GST_PAD_PROBE_TYPE_BUFFER)!=0)&&(info->data != NULL)) {
GstMapInfo buf_info;
gst_buffer_map(GST_BUFFER(info->data), &buf_info, GST_MAP_READ);
if( data->sink_pad_dumpfile != NULL ) {
fwrite(buf_info.data,1,buf_info.size,data->sink_pad_dumpfile);
}
gst_buffer_unmap(GST_BUFFER(info->data), &buf_info);
}
return GST_PAD_PROBE_OK;
}
/* This method is called by the idle GSource in the mainloop, to feed
CHUNK_SIZE bytes into appsrc.
* The ide handler is added to the mainloop when appsrc requests us to start
sending data (need-data signal)
* and is removed when appsrc has enough data (enough-data signal).
*/
static gboolean push_data (CustomData *data) {
GstBuffer *buffer;
GstFlowReturn ret;
int i;
GstMapInfo map;
gint16 *raw;
gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
gfloat freq;
/* Create a new empty buffer */
buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);
/* Set its timestamp and duration */
GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale
(data->num_samples, GST_SECOND, SAMPLE_RATE);
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (num_samples,
GST_SECOND, SAMPLE_RATE);
/* Generate some psychodelic waveforms */
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
raw = (gint16 *)map.data;
for (i = 0; i < num_samples; i++) {
raw[i] =
(gint16)2000*sin((300.0*3.14*2.0*(1.0*i+data->num_samples))/(SAMPLE_RATE));
/*
* generate a click every 500ms for later tempo measurement
*/
if( ((i+data->num_samples+1)%(SAMPLE_RATE/2)) == 0 ) {
raw[i] = -30000;
} else if( ((i+data->num_samples)%(SAMPLE_RATE/2)) == 0 ) {
raw[i] = 30000;
}
}
gst_buffer_unmap (buffer, &map);
data->num_samples += num_samples;
/* Push the buffer into the appsrc */
g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret);
/* Free the buffer now that we are done with it */
gst_buffer_unref (buffer);
if (ret != GST_FLOW_OK) {
/* We got some error, stop sending data */
return FALSE;
}
return TRUE;
}
/* timer for scaletempo changes
*/
static gboolean timeout_10000ms (CustomData *data) {
GstClockTime pos;
gst_element_query_position(data->pipeline,GST_FORMAT_TIME,&pos);
/*
* install a scaletempo segment with custom ratio
*/
{
gboolean update = TRUE;
GstSegment segment;
gst_segment_init (&segment, GST_FORMAT_TIME);
printf("gst_segment_do_seek ...\n");
if( gst_segment_do_seek (&segment,
0.5, GST_FORMAT_TIME,
GST_SEEK_FLAG_NONE,
GST_SEEK_TYPE_SET, pos + 1*GST_SECOND,
GST_SEEK_TYPE_SET, pos + 1*GST_SECOND + 5*GST_SECOND,
&update)) {
GstEvent *ev = gst_event_new_segment (&segment);
if( ev != NULL ) {
if( !gst_element_send_event(data->audio_scaletempo,ev) ) {
}
}
}
}
return FALSE;
}
/* This signal callback triggers when appsrc needs data. Here, we add an
idle handler
* to the mainloop to start pushing data into the appsrc */
static void start_feed (GstElement *source, guint size, CustomData *data) {
if (data->sourceid == 0) {
//g_print ("Start feeding\n");
data->sourceid = g_idle_add ((GSourceFunc) push_data, data);
}
}
/* This callback triggers when appsrc has enough data and we can stop
sending.
* We remove the idle handler from the mainloop */
static void stop_feed (GstElement *source, CustomData *data) {
if (data->sourceid != 0) {
//g_print ("Stop feeding\n");
g_source_remove (data->sourceid);
data->sourceid = 0;
}
}
/* This function is called when an error message is posted on the bus */
static void error_cb (GstBus *bus, GstMessage *msg, CustomData *data) {
GError *err;
gchar *debug_info;
/* Print error details on the screen */
gst_message_parse_error (msg, &err, &debug_info);
g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME
(msg->src), err->message);
g_printerr ("Debugging information: %s\n", debug_info ? debug_info :
"none");
g_clear_error (&err);
g_free (debug_info);
g_main_loop_quit (data->main_loop);
}
int main(int argc, char *argv[]) {
CustomData data;
GstAudioInfo info;
GstCaps *audio_caps;
GstBus *bus;
/* Initialize cumstom data structure */
memset (&data, 0, sizeof (data));
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
data.app_source = gst_element_factory_make ("appsrc", "audio_source");
data.audio_queue = gst_element_factory_make ("queue", "audio_queue");
g_object_set (data.audio_queue, "max-size-time", 100*GST_SECOND, NULL);
data.audio_scaletempo = gst_element_factory_make ("scaletempo",
"audio_scaletempo");
data.audio_convert1 = gst_element_factory_make ("audioconvert",
"audio_convert1");
data.audio_resample = gst_element_factory_make ("audioresample",
"audio_resample");
#if 1
data.audio_sink = gst_element_factory_make ("autoaudiosink",
"audio_sink");
#else
data.audio_sink = gst_element_factory_make ("alsasink", "audio_sink");
g_object_set (data.audio_sink, "sync", TRUE, "slave-method", 1, NULL);
#endif
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.app_source || !data.audio_queue ||
!data.audio_scaletempo || !data.audio_convert1 ||
!data.audio_resample || !data.audio_sink ) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Configure appsrc */
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1,
NULL);
audio_caps = gst_audio_info_to_caps (&info);
g_object_set (data.app_source, "is-live", FALSE, "caps", audio_caps,
"format", GST_FORMAT_TIME, NULL);
g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed),
&data);
g_signal_connect (data.app_source, "enough-data", G_CALLBACK
(stop_feed), &data);
/* Link all elements that can be automatically linked because they have
"Always" pads */
gst_bin_add_many (GST_BIN (data.pipeline), data.app_source,
data.audio_queue, data.audio_scaletempo, data.audio_convert1,
data.audio_resample,
data.audio_sink, NULL);
if (gst_element_link_many (data.app_source, data.audio_queue,
data.audio_scaletempo, data.audio_convert1, data.audio_resample,
data.audio_sink, NULL) != TRUE ) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/*
* pad probe for offline analysis
*/
data.sink_pad_dumpfile = fopen("gstseektest.raw","wb");
gst_pad_add_probe( gst_element_get_static_pad(data.audio_sink, "sink"),
GST_PAD_PROBE_TYPE_BUFFER,
sink_pad_probe,
&data,
NULL);
/* Instruct the bus to emit signals for each received message, and
connect to the interesting signals */
bus = gst_element_get_bus (data.pipeline);
gst_bus_add_signal_watch (bus);
g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb,
&data);
gst_object_unref (bus);
/*
* timer for scaletempo changes
*/
data.timeoutid = g_timeout_add ( 10000, (GSourceFunc) timeout_10000ms,
&data);
/* Start playing the pipeline */
gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
/* Create a GLib Main Loop and set it to run */
data.main_loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (data.main_loop);
/* Free resources */
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
--
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