java - Playing an rtp stream on android published with gstreamer -


i'm trying rtp connection between microphone on desktop pc , android smartphone.

i grab data using gstreamer. because of other applications using microphone @ same time in same system, there tcpsink, in data published.

this done call:

gst-launch-0.10 -v alsasrc ! 'audio/x-raw-int, depth=16, width=16, \               endianness=1234, channels=1, rate=16000' ! \               tcpserversink host=localhost port=20000

then create second stream, grabs tcp connection , convert rtp stream publish data on udp 

gst-launch-0.10 tcpclientsrc host=localhost protocol=0 port=20000 ! \              audio/x-raw-int,depth=16, width=16,endianness=1234, channels=1,\              rate=16000 ! lamemp3enc target=1 bitrate=64 cbr=true ! mad ! \              audioconvert ! audioresample ! mulawenc ! rtppcmupay pt=96 ! \              udpsink host=129.70.134.128 port=6000

this works while playing whith vlc player on localhost

vlc rtp://129.70.134.128:6000

now change host in udpsink android's phone one. shout while playing mplayer app.

after this, last step should play sound own app.

i'm trying stream android.net.rtp class.

audiomanager audiomanager = (audiomanager); mcontext.getsystemservice(mcontext.audio_service); audiomanager.setmode(audiomanager.mode_in_communication); audiostream inrtpstream = new audiostream(createinet("127.0.0.1"));  inrtpstream.associate(createinet(url), 6000); inrtpstream.setmode(rtpstream.mode_receive_only);   inrtpstream.setcodec(audiocodec.pcmu); inrtpstream.setdtmftype(96); // initialize audiogroup , attach audiostream audiogroup main_grp = new audiogroup(); main_grp.setmode(audiogroup.mode_normal); inrtpstream.join(main_grp);

but there silence. logging output makes me think, there kind of data, application trying play.

debug   audiogroup  stream[57] configured pcmu 8khz 20ms mode 2 debug   audiogroup  stream[64] configured raw 8khz 32ms mode 0 debug   audiogroup  stream[64] joins group[63] debug   audiogroup  group[63] switches mode 0 2 debug   audiogroup  stream[57] joins group[63] debug   audiogroup  reported frame count: output 1149, input 384 debug   audiogroup  adjusted frame count: output 1149, input 512 debug   audiogroup  latency: output 302, input 64

am missing starting stream, or switching speaker on?

all available volume sliders turned maximum. requested internet , record_audio permissions in manifest file.

the codecs should same.

thanks answers

you should pass in actual ip address , not loop address 127.0.0.1 in "new audiostream(createinet("127.0.0.1"));"


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