I use rtsp server.
Pipeline for audio:
"appsrc ! rawaudioparse ! audioconvert ! alawenc ! rtppcmapay name=pay1
pt=97"
...
GstAudioChannelPosition pos = GST_AUDIO_CHANNEL_POSITION_MONO;
gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_S16, 16000, 1, &pos);
...
std::uint32_t bSize = sample->Header().nBodySize;
GST_BUFFER_DTS(buf) = GST_BUFFER_TIMESTAMP(buf) =
gst_util_uint64_scale(m_audioSampleCount, GST_SECOND, 16000);
GST_BUFFER_DURATION(buf) = gst_util_uint64_scale(bSize / 2,
GST_SECOND, 16000);
...
I get log:
~#[17180]; 2019-09-04; 16:49:15.959; WARN; 0; /HttpServer.0/GStreamer:
basetransform
gstbasetransform.c:1364:gst_base_transform_setcaps:<audioconvert0> transform
could not transform audio/x-raw, format=(string)S16LE,
layout=(string)interleaved, rate=(int)16000, channels=(int)1 in anything we
support
How can I properly play mp2l2 ?
I used almost all parametrs for gst_audio_info_set_format (..),
GST_AUDIO_LAYOUT_INTERLEAVED/GST_AUDIO_LAYOUT_NON_INTERLEAVED
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