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Trying to capture a Radio Stream
Posted: 09 Feb 2009 19:22
by RatPackSopra
For some reason I can't figure out how to set up recording for my radio stream. I previously used an older version that had a Wizard for setting up recording and everything I've found out on the net has either not addressed the lack of the Wizard button or has confused me.
Could someone give me some guidance ...I'd really like to set things up to record at a certain time, but lets just get me heading in the right direction first. LOL
I will also mention that I have no interest in high quality recordings of the radio feed, since it will be a listen to it once and delete type thing. The feed I'll be recording is (
http://www.mainstreamnetwork.com/listen ... on=wccp-fm) in case that makes things easier. It's a asp stream in case the details don't show in the link.
I'm using 0.9.4 and I'm aware that I could get a newer version, but I'm using Ubuntu (linux) and that is the suggested version for now, so I need to stay with 0.9.4.
Thanks in advance!
Re: Trying to capture a Radio Stream
Posted: 09 Feb 2009 19:34
by Arite
Go to:
Media >> Convert / Save...
Click on the "Network" tab. Enter the URL there, then press "Convert / Save". If you are happy with e.g. 128kbps MP3 select "MP3" from the Profile drop-down box. Tick "File" and enter a destination (with *.mp3 extension ideally). If you want to listen to it at the same time tick "Play locally".
Then press "Stream" and it should stream to a file.
Arite.
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 17:50
by RatPackSopra
Thank you for the reply! I'm still having some issues getting the file to save though. I have tried to name it multiple ways (thinking that it might not like spaces in the file name or numbers,) but I've still not had any luck getting to save. I'm aware that the problem has to be "operator error," but maybe I'm missing a small part of the steps. I would prefer to save the stream at a very low quality like 20kbps or so, but I went ahead and stayed with the default settings (128kbps) for my testing, so it's not about that either. I have gotten it to save files, but none of them show any info in them (bytes.) I've went ahead and tried to play them just in case it was not showing the correct file size, but when totem or VLC tries to play them it confirms that they are empty. The stream IS playing because I can hear it playing, but it's not compiling any info.
Sorry to ask for more hand holding, but I'm a little frustrated and I'm sure it's something simple.
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 18:03
by Arite
The method above work both fine on Linux (Debian sid) and Windows for me. Post VLC's messages output. Go to "Tools >> Messages..." and set the "Verbosity Level" to "2". Then open/setup your stream etc. Paste the message here (use "Code" tags prefably).
If you want to change the bitrate etc. to e.g. 32kbps then don't select a profile, select "RAW" as the Encapsulation and e.g. "MP3" with 32kbps for the Audio codec. Or you can change the sout string directly (at the bottom of the window).
VLC 1.0.0 (not released yet) should be able to stream to Speex which is ideal for low bitrate encoding for speech.
Arite.
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 18:15
by RatPackSopra
I hope this is the correct info you are asking for. I set it to make a message of the short recording I did and then I changed the encap. to raw, 32kbps and then recorded 10 seconds or so.
main info: Running vlc with the default interface. Use 'cvlc' to use vlc without interface.
access_mms info: selecting stream[0x1] audio (20 kb/s)
access_mms info: connection successful
access_mms debug: recv command start_sequence:0x4a000001 command_id:0xb00bface length:32 len8:4 sequence 0x00000025 len8_II:2 dir_comm:0x0004001b
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 18:30
by Arite
Did you set the "Verbosity Level" to 2? There should be a lot more debug info that than if you did.
Cheers, Arite.
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 18:56
by RatPackSopra
I did set it to 2 and made a file name ...now when I made that recording it was only a few seconds, but after I shut down the recording I went and c&p'ed the info from the file and that was all it had.
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 19:17
by Arite
Just leave the messages window open for the duration of the streaming/conversion. The key bit is what happens after it has parsed the ASF source and streams out.
Arite.
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 20:04
by RatPackSopra
Ok, maybe this will shed some light on things ...do I need to have "dump raw file checked?"
The disappointing thing is ...I need to run some errands, but I'll be in about 8pm EST tonight.
Code: Select all
main info: Running vlc with the default interface. Use 'cvlc' to use vlc without interface.
qt4 debug: MRL passed to the Sout: http://www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm
qt4 debug: Profile Used: mp3
qt4 debug: Profile Used: Custom
qt4 debug: Profile Used: mp3
qt4 debug: Sout mrl :sout=#transcode{acodec=mp3,ab=32,channels=2}:duplicate{dst=display,dst=std{access=file,mux=raw,dst=WCCPtest7.mp3}}
main debug: adding item `Streaming' ( http://www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm )
main debug: rebuilding array of current - root Playlist
main debug: rebuild done - 1 items, index -1
main debug: starting new item
main debug: processing request item Streaming node Playlist skip 0
main debug: resyncing on Streaming
main debug: Streaming is at 0
main debug: creating new input thread
main debug: Creating an input for 'Streaming'
main debug: waiting for thread initialization
main debug: thread started
main debug: thread 2956364688 (input) created at priority 10 (input/input.c:370)
main debug: stream=`transcode'
main debug: looking for sout stream module: 1 candidate
main debug: stream=`duplicate'
main debug: looking for sout stream module: 1 candidate
stream_out_duplicate debug: creating 'duplicate'
stream_out_duplicate debug: * adding `display'
main debug: stream=`display'
main debug: looking for sout stream module: 1 candidate
main debug: using sout stream module "stream_out_display"
main debug: TIMER module_Need() : 1.046 ms - Total 1.046 ms / 1 intvls (Avg 1.046 ms)
stream_out_duplicate debug: * adding `std{access=file,mux=raw,dst=WCCPtest7.mp3}'
main debug: stream=`std'
main debug: looking for sout stream module: 1 candidate
qt4 debug: Updating the stream status: 3
main debug: set config option: sout-standard-access to file
main debug: set config option: sout-standard-mux to raw
main debug: set config option: sout-standard-dst to WCCPtest7.mp3
stream_out_standard debug: creating `file/raw://WCCPtest7.mp3'
stream_out_standard debug: extension is mp3
stream_out_standard debug: extension -> mux=(null)
stream_out_standard debug: using `file/raw://WCCPtest7.mp3'
main debug: looking for sout access module: 1 candidate
access_output_file debug: file access output opened (WCCPtest7.mp3)
main debug: using sout access module "access_output_file"
main debug: TIMER module_Need() : 19.360 ms - Total 19.360 ms / 1 intvls (Avg 19.360 ms)
stream_out_standard debug: access opened
main debug: looking for sout mux module: 1 candidate
mux_dummy debug: Dummy/Raw muxer opened
mux_dummy info: Open
main debug: using sout mux module "mux_dummy"
main debug: TIMER module_Need() : 0.847 ms - Total 0.847 ms / 1 intvls (Avg 0.847 ms)
main debug: muxer support adding stream at any time
stream_out_standard debug: mux opened
main debug: using sout stream module "stream_out_standard"
main debug: TIMER module_Need() : 36.835 ms - Total 36.835 ms / 1 intvls (Avg 36.835 ms)
main debug: using sout stream module "stream_out_duplicate"
main debug: TIMER module_Need() : 38.296 ms - Total 38.296 ms / 1 intvls (Avg 38.296 ms)
main debug: set config option: sout-transcode-acodec to mp3
main debug: set config option: sout-transcode-ab to 32
main debug: set config option: sout-transcode-channels to 2
stream_out_transcode debug: codec audio=mp3 0Hz 2 channels 32Kb/s
main debug: using sout stream module "stream_out_transcode"
main debug: TIMER module_Need() : 45.934 ms - Total 45.934 ms / 1 intvls (Avg 45.934 ms)
main debug: `http://www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm' gives access `http' demux `' path `www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm'
main debug: creating demux: access='http' demux='' path='www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm'
main debug: looking for access_demux module: 0 candidates
main warning: no access_demux module matched "http"
main debug: TIMER module_Need() : 0.163 ms - Total 0.163 ms / 1 intvls (Avg 0.163 ms)
main debug: creating access 'http' path='www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm'
main debug: looking for access module: 2 candidates
access_http debug: http: server='www.mainstreamnetwork.com' port=80 file='/listen/dynamicuasx.asp?station=wccp-fm
main debug: net: connecting to www.mainstreamnetwork.com port 80
qt4 debug: New Event: type 1103
qt4 debug: Updating the stream status: 1
main debug: connection: Operation now in progress
main debug: connection succeeded (socket = 26)
access_http debug: protocol 'HTTP' answer code 200
access_http debug: Server: Microsoft-IIS/6.0
access_http debug: this frame size=227
access_http debug: Content-Type: video/x-ms-asf
access_http debug: We have a Cookie we won't remember: ASPSESSIONIDASATDADB=KLBDNKHBPKMDNAIEPLOMHHIF; path=/
main debug: using access module "access_http"
main debug: TIMER module_Need() : 684.614 ms - Total 684.614 ms / 1 intvls (Avg 684.614 ms)
main debug: Using AStream*Stream
main debug: pre-buffering...
qt4 debug: New Event: type 1103
qt4 debug: Updating the stream status: 2
main debug: received first data for our buffer
main debug: creating demux: access='http' demux='' path='www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm'
main debug: looking for demux module: 52 candidates
playlist debug: found valid ASX playlist
main debug: using demux module "playlist"
main debug: TIMER module_Need() : 21.938 ms - Total 21.938 ms / 1 intvls (Avg 21.938 ms)
main debug: starting in async mode
main debug: `http://www.mainstreamnetwork.com/listen/dynamicuasx.asp?station=wccp-fm' successfully opened
playlist debug: param name strlen: 8
playlist debug: param name: htmlview
main debug: deleting item `Streaming'
main debug: rebuilding array of current - root Playlist
main debug: rebuild done - 1 items, index -1
main debug: EOF reached
main debug: control type=1
main debug: no fetch required for 1 (art currently (null))
main debug: finished input
main debug: waitpipe: object killed
main debug: dying input
main debug: removing module "playlist"
main debug: removing module "access_http"
main debug: thread ended
qt4 debug: Updating the stream status: 8
main debug: dead input
main debug: thread 2956364688 joined (playlist/engine.c:244)
main debug: TIMER input launching for 'Streaming' : 762.977 ms - Total 762.977 ms / 1 intvls (Avg 762.977 ms)
main debug: destroying chain... (name=transcode)
main debug: destroying chain... (name=duplicate)
stream_out_duplicate debug: closing a duplication
main debug: destroying chain... (name=display)
main debug: removing module "stream_out_display"
main debug: destroying chain done
main debug: destroying chain... (name=std)
mux_dummy debug: Dummy/Raw muxer closed
main debug: removing module "mux_dummy"
access_output_file debug: file access output closed
main debug: removing module "access_output_file"
main debug: removing module "stream_out_standard"
main debug: destroying chain done
main debug: removing module "stream_out_duplicate"
main debug: destroying chain done
main debug: removing module "stream_out_transcode"
main debug: destroying chain done
main debug: starting new item
main debug: changing item without a request (current -1/1)
main debug: using item 0
main debug: creating new input thread
main debug: Creating an input for '1 '
main debug: thread started
main debug: waiting for thread initialization
main debug: stream=`transcode'
main debug: looking for sout stream module: 1 candidate
main debug: stream=`duplicate'
main debug: looking for sout stream module: 1 candidate
stream_out_duplicate debug: creating 'duplicate'
stream_out_duplicate debug: * adding `display'
main debug: stream=`display'
main debug: looking for sout stream module: 1 candidate
main debug: using sout stream module "stream_out_display"
main debug: TIMER module_Need() : 0.839 ms - Total 0.839 ms / 1 intvls (Avg 0.839 ms)
stream_out_duplicate debug: * adding `std{access=file,mux=raw,dst=WCCPtest7.mp3}'
main debug: stream=`std'
main debug: looking for sout stream module: 1 candidate
main debug: set config option: sout-standard-access to file
main debug: set config option: sout-standard-mux to raw
main debug: set config option: sout-standard-dst to WCCPtest7.mp3
stream_out_standard debug: creating `file/raw://WCCPtest7.mp3'
stream_out_standard debug: extension is mp3
stream_out_standard debug: extension -> mux=(null)
stream_out_standard debug: using `file/raw://WCCPtest7.mp3'
main debug: looking for sout access module: 1 candidate
main debug: thread 2956364688 (input) created at priority 10 (input/input.c:370)
access_output_file debug: file access output opened (WCCPtest7.mp3)
main debug: using sout access module "access_output_file"
main debug: TIMER module_Need() : 0.533 ms - Total 0.533 ms / 1 intvls (Avg 0.533 ms)
stream_out_standard debug: access opened
main debug: looking for sout mux module: 1 candidate
mux_dummy debug: Dummy/Raw muxer opened
mux_dummy info: Open
main debug: using sout mux module "mux_dummy"
main debug: TIMER module_Need() : 0.343 ms - Total 0.343 ms / 1 intvls (Avg 0.343 ms)
main debug: muxer support adding stream at any time
stream_out_standard debug: mux opened
main debug: using sout stream module "stream_out_standard"
main debug: TIMER module_Need() : 4.484 ms - Total 4.484 ms / 1 intvls (Avg 4.484 ms)
main debug: using sout stream module "stream_out_duplicate"
main debug: TIMER module_Need() : 5.544 ms - Total 5.544 ms / 1 intvls (Avg 5.544 ms)
qt4 debug: Updating the stream status: 3
main debug: set config option: sout-transcode-acodec to mp3
main debug: set config option: sout-transcode-ab to 32
main debug: set config option: sout-transcode-channels to 2
stream_out_transcode debug: codec audio=mp3 0Hz 2 channels 32Kb/s
main debug: using sout stream module "stream_out_transcode"
main debug: TIMER module_Need() : 13.598 ms - Total 13.598 ms / 1 intvls (Avg 13.598 ms)
main debug: `mms://208.43.60.68/wccp-fm' gives access `mms' demux `' path `208.43.60.68/wccp-fm'
main debug: creating demux: access='mms' demux='' path='208.43.60.68/wccp-fm'
main debug: looking for access_demux module: 0 candidates
main warning: no access_demux module matched "mms"
main debug: TIMER module_Need() : 0.167 ms - Total 0.167 ms / 1 intvls (Avg 0.167 ms)
main debug: creating access 'mms' path='208.43.60.68/wccp-fm'
main debug: looking for access module: 1 candidate
access_mms debug: waiting for connection...
main debug: net: connecting to 208.43.60.68 port 1755
main debug: connection: Operation now in progress
qt4 debug: New Event: type 1103
qt4 debug: Updating the stream status: 1
main debug: connection succeeded (socket = 26)
access_mms debug: connection(tcp) with "208.43.60.68:1755" successful
access_mms debug: generated guid: babac001-0d82-c86b-0e4632213f43c528
access_mms debug: recv command start_sequence:0x4a000001 command_id:0xb00bface length:128 len8:16 sequence 0x00000000 len8_II:14 dir_comm:0x00040001
access_mms debug: 0x01 --> server_version:"9.01.01.5000" tool_version:"" update_player_url:"" encryption_type:"NTLM"
access_mms debug: recv command start_sequence:0x00000001 command_id:0xb00bface length:80 len8:10 sequence 0x00000001 len8_II:8 dir_comm:0x00040002
access_mms debug: recv command start_sequence:0x4a000001 command_id:0xb00bface length:136 len8:17 sequence 0x00000002 len8_II:15 dir_comm:0x00040006
access_mms debug: media file name/path accepted
access_mms debug: answer 0x06 flags:0x02000000 media_length:0s packet_length:831l packet_count:0 max_bit_rate:20718 header_size:5131
access_mms debug: reading header
access_mms debug: recv command start_sequence:0x00000001 command_id:0xb00bface length:40 len8:5 sequence 0x00000003 len8_II:3 dir_comm:0x00040011
access_mms debug: header incomplete (823/5131), reading more
access_mms debug: header incomplete (1646/5131), reading more
access_mms debug: header incomplete (2469/5131), reading more
access_mms debug: header incomplete (3292/5131), reading more
access_mms debug: header incomplete (4115/5131), reading more
access_mms debug: header incomplete (4938/5131), reading more
access_mms debug: header complete(5131)
access_mms info: selecting stream[0x1] audio (20 kb/s)
access_mms debug: recv command start_sequence:0x4a000001 command_id:0xb00bface length:32 len8:4 sequence 0x00000004 len8_II:2 dir_comm:0x00040021
access_mms info: connection successful
access_mms debug: connected to 208.43.60.68:1755
access_mms debug: recv command start_sequence:0x4a000001 command_id:0xb00bface length:56 len8:7 sequence 0x00000005 len8_II:5 dir_comm:0x00040005
access_mms debug: streaming started
main debug: thread 2947570576 (mmstu keepalive thread) created at priority 0 (mmstu.c:211)
main debug: using access module "access_mms"
main debug: thread started
main debug: TIMER module_Need() : 2407.310 ms - Total 2407.310 ms / 1 intvls (Avg 2407.310 ms)
qt4 debug: New Event: type 1103
qt4 debug: Updating the stream status: 2
main debug: Using AStream*Stream
main debug: pre-buffering...
main debug: received first data for our buffer
main debug: pre-buffering done 9972 bytes in 1s - 7 kbytes/s
main debug: creating demux: access='mms' demux='' path='208.43.60.68/wccp-fm'
main debug: looking for demux module: 52 candidates
asf debug: found object guid: 0x75b22630-0x668e-0x11cf-0xa6d900aa0062ce6c size:5081
asf debug: read "header object" subobj:7, reserved1:1, reserved2:2
asf debug: found object guid: 0x75b22633-0x668e-0x11cf-0xa6d900aa0062ce6c size:188
asf debug: read "content description object" title:"The Super Sports FM WCCP- 104.9 Clemson, SC" artist:"Golden Corners Broadcasting" copyright:"2004" description:"" rating:""
asf debug: found object guid: 0x7bf875ce-0x468d-0x11d1-0x8d82006097c9a2b2 size:32
asf debug: read "stream bitrate properties object"
asf debug: - stream=1 bitrate=20718
asf debug: found object guid: 0x8cabdca1-0xa947-0x11cf-0x8ee400c00c205365 size:104
asf debug: read "file properties object" file_id:0xb94743a0-0xe841-0x4ffe-0xbd78d30f39d26dfc file_size:5131 creation_date:128786569736090000 data_packets_count:4294967295 play_duration:0 send_duration:0 preroll:1025 flags:9 min_data_packet_size:831 max_data_packet_size:831 max_bitrate:20718
asf debug: found object guid: 0x5fbf03b5-0xa92e-0x11cf-0x8ee300c00c205365 size:4273
asf debug: read "header extension object" reserved1:0xabd3d211-0xa9ba-0x11cf-0x8ee600c00c205365 reserved2:6 header_extension_size:4227
asf debug: found object guid: 0x7c4346a9-0xefe0-0x4bfc-0xb229393ede415c85 size:39
asf debug: read "language list object" 1 entries
asf debug: - 'en-us'
asf debug: found object guid: 0x14e6a5cb-0xc672-0x4332-0x8399a96952065b5a size:88
asf debug: read "extended stream properties object":
asf debug: - start=0 end=0
asf debug: - data bitrate=20000 buffer=1025 initial fullness=0
asf debug: - alternate data bitrate=20000 buffer=1025 initial fullness=0
asf debug: - maximum object size=800
asf debug: - flags=0x0
asf debug: - stream number=1 language=0
asf debug: - average time per frame=0
asf debug: - stream name count=0
asf debug: - payload extension system count=0
asf debug: found object guid: 0x26f18b5d-0x4584-0x47ec-0x9f5f0e651f0452c9 size:26
asf warning: unknown asf object (not loaded)
asf debug: found object guid: 0xc5f8cbea-0x5baf-0x4877-0x8467aa8c44fa4cca size:122
asf debug: read "metadata object" 2 entries
asf debug: - IsVBR=0
asf debug: - DeviceConformanceTemplate=L2
asf debug: found object guid: 0x1806d474-0xcadf-0x4509-0xa4ba9aabcb96aae8 size:3952
asf warning: unknown asf object (not loaded)
asf debug: found object guid: 0xb7dc0791-0xa9b7-0x11cf-0x8ee600c00c205365 size:114
asf debug: read "stream Properties object" stream_type:0xf8699e40-0x5b4d-0x11cf-0xa8fd00805f5c442b error_correction_type:0xbfc3cd50-0x618f-0x11cf-0x8bb200aa00b4e220 time_offset:0 type_specific_data_length:28 error_correction_data_length:8 flags:0x1 stream_number:1
asf debug: found object guid: 0xd2d0a440-0xe307-0x11d2-0x97f000a0c95ea850 size:166
asf debug: read "extended content description object"
asf debug: - 'WMFSDKVersion' = '11.0.5721.5251'
asf debug: - 'WMFSDKNeeded' = '0.0.0.0000'
asf debug: - 'IsVBR' = 'false'
asf debug: found object guid: 0x86d15240-0x311d-0x11d0-0xa3a400a0c90348f6 size:174
asf debug: read "codec list object" reserved_guid:0x86d15241-0x311d-0x11d0-0xa3a400a0c90348f6 codec_entries_count:1
asf debug: - codec[0] audio name:"Windows Media Audio 9.2" description:" 20 kbps, 16 kHz, stereo 1-pass CBR" information_length:2
asf debug: found object guid: 0x75b22636-0x668e-0x11cf-0xa6d900aa0062ce6c size:50
asf debug: read "data object" file_id:0xb94743a0-0xe841-0x4ffe-0xbd78d30f39d26dfc total data packet:0 reserved:257
asf debug: + 'Unknown' GUID 0x0-0x0-0x0-0x0000000000000000 size:0pos:0
asf debug: + 'Header' GUID 0x75b22630-0x668e-0x11cf-0xa6d900aa0062ce6c size:5081pos:0
asf debug: | + 'Content Description' GUID 0x75b22633-0x668e-0x11cf-0xa6d900aa0062ce6c size:188pos:30
asf debug: | + 'Stream Bitrate Properties' GUID 0x7bf875ce-0x468d-0x11d1-0x8d82006097c9a2b2 size:32pos:218
asf debug: | + 'File Properties' GUID 0x8cabdca1-0xa947-0x11cf-0x8ee400c00c205365 size:104pos:250
asf debug: | + 'Header Extension' GUID 0x5fbf03b5-0xa92e-0x11cf-0x8ee300c00c205365 size:4273pos:354
asf debug: | | + 'Language List' GUID 0x7c4346a9-0xefe0-0x4bfc-0xb229393ede415c85 size:39pos:400
asf debug: | | + 'Extended Stream Properties' GUID 0x14e6a5cb-0xc672-0x4332-0x8399a96952065b5a size:88pos:439
asf debug: | | + 'Unknown' GUID 0x26f18b5d-0x4584-0x47ec-0x9f5f0e651f0452c9 size:26pos:527
asf debug: | | + 'Metadata' GUID 0xc5f8cbea-0x5baf-0x4877-0x8467aa8c44fa4cca size:122pos:553
asf debug: | | + 'Padding' GUID 0x1806d474-0xcadf-0x4509-0xa4ba9aabcb96aae8 size:3952pos:675
asf debug: | + 'Stream Properties' GUID 0xb7dc0791-0xa9b7-0x11cf-0x8ee600c00c205365 size:114pos:4627
asf debug: | + 'Extended content description' GUID 0xd2d0a440-0xe307-0x11d2-0x97f000a0c95ea850 size:166pos:4741
asf debug: | + 'Codec List' GUID 0x86d15240-0x311d-0x11d0-0xa3a400a0c90348f6 size:174pos:4907
asf debug: + 'Data' GUID 0x75b22636-0x668e-0x11cf-0xa6d900aa0062ce6c size:50pos:5081
asf debug: found 1 streams
main debug: selecting program id=0
qt4 debug: New Event: type 1108
asf debug: added new audio stream(codec:0x161,ID:1)
main debug: using demux module "asf"
main debug: TIMER module_Need() : 19.098 ms - Total 19.098 ms / 1 intvls (Avg 19.098 ms)
main debug: looking for packetizer module: 18 candidates
main debug: using packetizer module "packetizer_copy"
main debug: TIMER module_Need() : 5.703 ms - Total 5.703 ms / 1 intvls (Avg 5.703 ms)
main debug: stream out mode -> no decoder thread
main debug: starting in async mode
access_mms warning: unimplemented query in control
main debug: `mms://208.43.60.68/wccp-fm' successfully opened
packetizer_copy debug: need dts > 0
main debug: control type=1
qt4 debug: New Event: type 1103
qt4 debug: Updating the stream status: 3
main debug: adding a new sout input (sout_input:0x995f0a8)
stream_out_transcode debug: creating audio transcoding from fcc=`wma2' to fcc=`mp3 '
main debug: looking for decoder module: 30 candidates
avcodec debug: libavcodec initialized (interface 3355136 )
avcodec warning: Physical channel configuration not set : guessing
avcodec debug: ffmpeg codec (Windows Media Audio 2) started
main debug: using decoder module "avcodec"
main debug: TIMER module_Need() : 15.916 ms - Total 15.916 ms / 1 intvls (Avg 15.916 ms)
main debug: looking for encoder module: 10 candidates
avcodec debug: libavcodec already initialized
avcodec debug: found encoder MPEG Audio layer 1/2/3
main debug: using encoder module "avcodec"
main debug: TIMER module_Need() : 12.733 ms - Total 12.733 ms / 1 intvls (Avg 12.733 ms)
stream_out_duplicate debug: duplicated a new stream codec=mpga (es=0 group=0)
main debug: looking for decoder module: 30 candidates
main debug: using decoder module "mpeg_audio"
main debug: TIMER module_Need() : 0.314 ms - Total 0.314 ms / 1 intvls (Avg 0.314 ms)
main debug: thread 2911243152 (decoder) created at priority 5 (input/decoder.c:217)
stream_out_duplicate debug: - added for output 0
main debug: adding a new input
mux_dummy debug: adding input
stream_out_duplicate debug: - added for output 1
main debug: thread started
main debug: switching to sync mode
mpeg_audio debug: MPGA channels:2 samplerate:16000 bitrate:32
main debug: no aout present, spawning one
main debug: looking for audio output module: 4 candidates
alsa debug: opening ALSA device `default'
main debug: thread 2891959184 (aout) created at priority 15 (alsa.c:687)
main debug: using audio output module "alsa"
main debug: thread started
main debug: TIMER module_Need() : 25.769 ms - Total 25.769 ms / 1 intvls (Avg 25.769 ms)
main debug: output 'fl32' 16000 Hz Stereo frame=1 samples/8 bytes
main debug: mixer 'fl32' 16000 Hz Stereo frame=1 samples/8 bytes
main debug: no need for any filter
main debug: looking for audio mixer module: 3 candidates
main debug: using audio mixer module "float32_mixer"
main debug: TIMER module_Need() : 0.524 ms - Total 0.524 ms / 1 intvls (Avg 0.524 ms)
main debug: input 'mpga' 16000 Hz Stereo frame=576 samples/729 bytes
main debug: filter(s) 'mpga'->'fl32' 16000 Hz->16000 Hz Stereo->Stereo
main debug: looking for audio filter module: 24 candidates
main debug: using audio filter module "mpgatofixed32"
main debug: TIMER module_Need() : 1.667 ms - Total 1.667 ms / 1 intvls (Avg 1.667 ms)
main debug: found a filter for the whole conversion
main debug: filter(s) 'fl32'->'fl32' 17600 Hz->16000 Hz Stereo->Stereo
main debug: looking for audio filter module: 24 candidates
main debug: using audio filter module "bandlimited_resampler"
main debug: TIMER module_Need() : 0.511 ms - Total 0.511 ms / 1 intvls (Avg 0.511 ms)
main debug: found a filter for the whole conversion
main warning: buffer is 108000 in advance, triggering downsampling
main warning: audio drift is too big (-140438), clearing out
main warning: timing screwed, stopping resampling
main warning: mixer start isn't output start (-24576)
main warning: output date isn't PTS date, requesting resampling (-41063)
main warning: buffer is 76563 in advance, triggering downsampling
main warning: output date isn't PTS date, requesting resampling (82070)
main warning: resampling stopped after 467959 usec (drift: -5257)
main debug: incoming request - stopping current input
main debug: waitpipe: object killed
main debug: dying input
main debug: removing module "mpeg_audio"
main debug: thread ended
main debug: dying input
main debug: dying input
main debug: dying input
main debug: dying input
main debug: dying input
main debug: control type=0
main debug: control: stopping input
asf debug: free asf object 0x75b22633-0x668e-0x11cf-0xa6d900aa0062ce6c
asf debug: free asf object 0x7bf875ce-0x468d-0x11d1-0x8d82006097c9a2b2
asf debug: free asf object 0x8cabdca1-0xa947-0x11cf-0x8ee400c00c205365
asf debug: free asf object 0x7c4346a9-0xefe0-0x4bfc-0xb229393ede415c85
asf debug: free asf object 0x14e6a5cb-0xc672-0x4332-0x8399a96952065b5a
asf warning: unknown asf object 0x26f18b5d-0x4584-0x47ec-0x9f5f0e651f0452c9
asf debug: free asf object 0xc5f8cbea-0x5baf-0x4877-0x8467aa8c44fa4cca
asf warning: unknown asf object 0x1806d474-0xcadf-0x4509-0xa4ba9aabcb96aae8
asf debug: free asf object 0x5fbf03b5-0xa92e-0x11cf-0x8ee300c00c205365
asf debug: free asf object 0xb7dc0791-0xa9b7-0x11cf-0x8ee600c00c205365
asf debug: free asf object 0xd2d0a440-0xe307-0x11d2-0x97f000a0c95ea850
asf debug: free asf object 0x86d15240-0x311d-0x11d0-0xa3a400a0c90348f6
asf debug: free asf object 0x75b22630-0x668e-0x11cf-0xa6d900aa0062ce6c
asf debug: free asf object 0x75b22636-0x668e-0x11cf-0xa6d900aa0062ce6c
main debug: removing module "packetizer_copy"
main debug: killing decoder fourcc `wma2', 0 PES in FIFO
main debug: removing a sout input (sout_input:0x995f0a8)
main debug: TIMER encoding audio frame : 2.524 ms - Total 596.522 ms / 109 intvls (Avg 5.473 ms)
avcodec debug: ffmpeg codec (Windows Media Audio 2) stopped
main debug: removing module "avcodec"
main debug: removing module "avcodec"
main debug: thread 2911243152 joined (input/decoder.c:248)
main debug: killing decoder fourcc `mpga', 142 PES in FIFO
main debug: removing module "mpgatofixed32"
main debug: removing module "bandlimited_resampler"
qt4 debug: Updating the stream status: 8
main debug: thread ended
main debug: thread 2891959184 joined (alsa.c:742)
main debug: removing module "alsa"
main debug: removing module "float32_mixer"
mux_dummy debug: removing input
main warning: no more input streams for this mux
main debug: Program doesn't contain anymore ES
main debug: removing module "asf"
main debug: thread ended
main debug: thread 2947570576 joined (mmstu.c:228)
access_mms debug: Connection closed
main debug: removing module "access_mms"
main debug: thread ended
main debug: dead input
main debug: thread 2956364688 joined (playlist/engine.c:244)
main debug: TIMER input launching for '1 ' : 3716.724 ms - Total 3716.724 ms / 1 intvls (Avg 3716.724 ms)
main debug: destroying chain... (name=transcode)
main debug: destroying chain... (name=duplicate)
stream_out_duplicate debug: closing a duplication
main debug: destroying chain... (name=display)
main debug: removing module "stream_out_display"
main debug: destroying chain done
main debug: destroying chain... (name=std)
mux_dummy debug: Dummy/Raw muxer closed
main debug: removing module "mux_dummy"
access_output_file debug: file access output closed
main debug: removing module "access_output_file"
main debug: removing module "stream_out_standard"
main debug: destroying chain done
main debug: removing module "stream_out_duplicate"
main debug: destroying chain done
main debug: removing module "stream_out_transcode"
main debug: destroying chain done
Re: Trying to capture a Radio Stream
Posted: 12 Feb 2009 20:19
by Arite
Ok, maybe this will shed some light on things ...do I need to have "dump raw file checked?"
No, that will just dump the stream as-is (i.e. will not transcode it).
The messages output looks OK - so did VLC just generate an empty file called "WCCPtest7.mp3" (presumably that is in your home directory?)? Try entering the full path to it - i.e. /home/(your-username)/WCCPtest7.mp3 or something similar (where (your-username) is your username).
Arite.
Re: Trying to capture a Radio Stream
Posted: 13 Feb 2009 01:39
by RatPackSopra
Well I had a brain cloud ...I had been saving my test streams into a specific folder, but some how had forgotten to run the last 3 attempts into that same folder. LOL
It looks like it is working fine and thank you!
Now ...is there a way to start the recording by schedule? I'm usually away every day at the time a certain show starts and I'd like to set it up to start recording while I'm away.
Re: Trying to capture a Radio Stream
Posted: 13 Feb 2009 02:05
by Arite
Well I had a brain cloud ...I had been saving my test streams into a specific folder, but some how had forgotten to run the last 3 attempts into that same folder. LOL
It looks like it is working fine and thank you!
Ah OK - glad it is working
.
Now ...is there a way to start the recording by schedule? I'm usually away every day at the time a certain show starts and I'd like to set it up to start recording while I'm away.
Yes, you can set up scheduled streams etc. using VLM:
http://wiki.videolan.org/VLM
See some of the examples in the wiki. You can use e.g. the HTTP interface or the telnet one.
Arite.
Re: Trying to capture a Radio Stream
Posted: 13 Feb 2009 05:31
by RatPackSopra
Does anyone have any opinions on whether it is easier to set up timed recordings with HTTP or Telnet?
Re: Trying to capture a Radio Stream
Posted: 13 Feb 2009 15:46
by Arite
The telnet interface with a text file is good, however you might fine the Web interface easier to handle.
To use the telnet interface, for example open a text editor and do something like this (this is based on the example in the wiki):
Code: Select all
new Test broadcast enabled
setup Test input mms://208.43.60.68/wccp-fm
setup Test output #transcode{acodec=mp3,ab=32,channels=2}:duplicate{dst=std{access=file,mux=raw,dst=/home/(YOUR_USERNAME)/out.mp3}}
new Sched schedule enabled
setup Sched date 2009/02/13-14:34:00
setup Sched append control Test play
Where (YOUR_USERNAME) is your username. Save it as e.g. "test-schedule.conf"
Then open up a terminal and enter:
Code: Select all
vlc -I telnet --vlm-conf test-schedule.conf
Where test-schedule.conf is the path/name of you *.conf file, and 2009/02/13-14:34:00 is the time you want to start the stream.
Then at the specified time and date VLC should start steaming 32kbps MP3 to ~/out.mp3. Note I used the actual mms:// stream that VLC derived from the ASP file as just linking to the ASP file didn't work.
You should be able to expand on that to e.g. stop, set up repeat schedules etc. at a certain times. Look at the wiki for more info.
Arite.
Re: Trying to capture a Radio Stream
Posted: 15 Feb 2009 22:50
by webster.net.ctp
Arite,
I've had a similar issue as this one. Because of your reposonses on this thread I've finally realized to un-check the "dump raw input" box to save the stream. (When it WAS checked, the file a specified to convert/save to never showed up in windows explorer).
THANK YOU!!!
]\
Re: Trying to capture a Radio Stream
Posted: 17 Feb 2009 22:04
by webster.net.ctp
This is probably a newb question, but even after looking thru this thread I'm having trouble figuring out how to properly capture this radio stream:
http://94.23.8.96:8030
CODEC DETAILS:
Codec: mp4a
Type: Audio
Channels: 2
Sample rate: 44100
AAC extension: SBR
(Stream bitrate 62-64kb/s)
I use Convert/Save with profile MPEG-4 audio AAC with Codec as MPEG 4 Audio (AAC) and Bitrate = 64kb/s, but the sound is all distorted.
Play locally is checked.
When use Convert/Save with the MP3 profile it works as expected and the sound quality is fine.
It's probably a simple setting I'm missing here. Thanks for any help in advance!
Re: Trying to capture a Radio Stream
Posted: 18 Feb 2009 01:10
by Arite
Bear in mind that LC-AAC at 64kbps is significantly lower quality than HE-AAC (AAC with SBR).
Try adding:
To the transcode part of sout string.
For example:
Code: Select all
:sout=#transcode{acodec=mp4a,ab=64,channels=2,samplerate=44100}:duplicate{dst=std{access=file,mux=mp4,dst=OUTPUT}}
Where OUTPUT is you destination.
Arite.
Re: Trying to capture a Radio Stream
Posted: 18 Feb 2009 03:54
by webster.net.ctp
Arite, that one fix solved my entire (AAC) sound quality problem; like magic. Very helpful.
Re: Trying to capture a Radio Stream
Posted: 18 Feb 2009 05:02
by Arite
Glad it worked
.
Arite.
Re: Trying to capture a Radio Stream
Posted: 29 Mar 2009 23:04
by kwanbis
I had a similar problem.
In my case, the problem was that i was trying to save to a folder name "$music", and somehow, vlc doesn't likes the $ in it.
I took it out, and all is fine now.
Anyway, is it possible to tell VLC to create MP3s from radio streams, but to cut them each our?
I want it to automatically record each day, from 10 AM to 12PM, and to generate something like
YYYYMMDD-10-11
YYYYMMDD-11-12
Is it possible?
Re: Trying to capture a Radio Stream
Posted: 12 Apr 2009 22:20
by olrt
Arite,
I've had a similar issue as this one. Because of your reposonses on this thread I've finally realized to un-check the "dump raw input" box to save the stream. (When it WAS checked, the file a specified to convert/save to never showed up in windows explorer).
THANK YOU!!!
]\
I observed the same behaviour.
Is it a bug ? (it seems it worked with previous versions).
Or is it a voluntary limitation so that to force downgrade of audio quality by transcoding ?
Re: Trying to capture a Radio Stream
Posted: 12 Apr 2009 22:27
by olrt
Forget about my post. There's no downgrade of audio quality since there no need to transcode. It's just necessary to specify "raw" for the muxer.
Re: Trying to capture a Radio Stream
Posted: 12 Apr 2009 22:46
by olrt
Now another bug in 0.9.9
I saved the playlist in a file (.m3u).
I quit VLC then reopen the playlist.
The problem is that the streaming to disk doesn't work.
Quite annoying to repeat the convert procedure each time one want to record a Web radio stream
Re: Trying to capture a Radio Stream
Posted: 12 Apr 2009 22:51
by olrt
Same scenario and same bug with saving the playlist in a .xspf file.
Will bug be fixed in 1.0.0 ?