ok Tanu, first of all thx a lot for your reply. I used pactl list to localize the correct input device, as well for the output, but I'm not shure is the right one. this is the output device I choose: Sink #0 State: SUSPENDED Name: alsa_output.pci-0000_00_07.0.analog-stereo Description: Internal Audio Analog Stereo Driver: module-alsa-card.c Sample Specification: s16le 2ch 44100Hz Channel Map: front-left,front-right Owner Module: 6 Mute: no Volume: 0: 88% 1: 88% 0: -3.33 dB 1: -3.33 dB balance 0.00 Base Volume: 100% 0.00 dB Monitor Source: alsa_output.pci-0000_00_07.0.analog-stereo.monitor Latency: 0 usec, configured 0 usec Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY Properties: alsa.resolution_bits = "16" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "ALC888 Analog" alsa.id = "ALC888 Analog" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "HDA NVidia" alsa.long_card_name = "HDA NVidia at 0xf9e78000 irq 21" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:07.0" sysfs.path = "/devices/pci0000:00/0000:00:07.0/sound/card0" device.bus = "pci" device.vendor.id = "10de" device.vendor.name = "nVidia Corporation" device.product.id = "0774" device.product.name = "MCP72XE/MCP72P/MCP78U/MCP78S High Definition Audio" device.form_factor = "internal" device.string = "front:0" device.buffering.buffer_size = "65536" device.buffering.fragment_size = "32768" device.access_mode = "mmap+timer" device.profile.name = "analog-stereo" device.profile.description = "Analog Stereo" device.description = "Internal Audio Analog Stereo" alsa.mixer_name = "Nvidia MCP78 HDMI" alsa.components = "HDA:10ec0888,14627511,00100001 HDA:10de0002,10de0101,00100000" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Ports: analog-output: Analog Output (priority. 10000) analog-output-headphones: Analog Headphones (priority. 9000) Active Port: analog-output Anyway with this command pactl load-module module-loopback source=alsa_input.pci-0000_01_08.0.analog-stereo sink=alsa_output.pci-0000_00_07.0.analog-stereo rate=32000 format=s16le I can ear the tv audio from the standard output. That is fantastic, because I tried as well with the BT headset and it works. Sadly the audio has a very poor quality: 1) the audio is pulsing, it seems that it pump up and down the master volume 2) the rate is wrong, because it seems an old 33 rpm listening at 45. When I use sox to redirect audio it needs to set the rate to 32000, but in this case has no effect. I use s16le format because I know that the format used from the tv card is Signed Little Endian 16 bit. Finally I tried to change the latency to 1 and as well to 2000 but nothing change, or very little changes. I need different settings or may be I need to change the sink device? > Where did you get that "pcm:0" string? I haven't used tvtime, but > looking at its man page it looks like the part before the colon should > be a path to an OSS mixer device, e.g. /dev/mixer, and the part after > the colon should be one of a fixed set of strings, one of which is > "line" I'm not very experienced in audio devices, so I just read the tvtime man, but probably I was not accurate. Anyway now tvtime is capable to manage the mixer, because when I switch on tvtime I listen the audio, when I switch it off the audio from tv card is muted. > Does Skype use pulseaudio for input? Are you running the newest > release of Skype that I hear finally works decently with pulseaudio? Yes, I'm using the new Beta from Skype, and now the audio if perfectly integrated in Pulseaudio. If it find Pulseaudio 0.9.15 or later it locks the audio device management and leave everything to Pulse. This is perfect because there only one location where to manage the audio device. Furthermore, the cpu is not longer stressed when skype makes a phone call.
