This project started attempting to capture the output of window.speechSynthesis.speak(). Consider https://stackoverflow.com/a/43553706. If run $ pacmd list-sink-inputs will get index: 219 driver: <protocol-native.c> flags: state: RUNNING sink: 4 <alsa_output.pci-0000_00_14.2.analog-stereo> volume: mono: 64460 / 98% / -0.43 dB balance 0.00 muted: no current latency: 0.00 ms requested latency: 0.59 ms sample spec: s16le 1ch 22050Hz channel map: mono Mono resample method: speex-float-1 module: 14 client: 192 <speech-dispatcher-espeak-ng> properties: media.name = "playback" application.name = "speech-dispatcher-espeak-ng" native-protocol.peer = "UNIX socket client" native-protocol.version = "33" application.process.id = "45464" application.process.user = "ubuntu-studio" application.process.host = "ubuntu-studio" application.process.binary = "sd_espeak-ng" application.language = "C" window.x11.display = ":0.0" application.process.machine_id = "<id>" application.process.session_id = "<id>" module-stream-restore.id = "sink-input-by-application-name:speech-dispatcher-espeak-ng" $ pacmd list-sources 2 source(s) available. * index: 4 name: <alsa_input.usb-046d_0804_0B17FC60-02.mono-fallback> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: RUNNING suspend cause: (none) priority: 9040 volume: mono: 0 / 0% / -inf dB balance 0.00 base volume: 20724 / 32% / -30.00 dB volume steps: 65537 muted: yes current latency: 4.60 ms max rewind: 0 KiB sample spec: s16le 1ch 48000Hz channel map: mono Mono used by: 1 linked by: 1 configured latency: 40.00 ms; range is 0.50 .. 2000.00 ms card: 1 <alsa_card.usb-046d_0804_0B17FC60-02> module: 8 properties: alsa.resolution_bits = "16" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "USB Audio" alsa.id = "USB Audio" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "1" alsa.card_name = "USB Device 0x46d:0x804" alsa.long_card_name = "USB Device 0x46d:0x804 at usb-0000:00:12.0-1.4, high speed" alsa.driver_name = "snd_usb_audio" device.bus_path = "pci-0000:00:12.0-usb-0:1.4:1.2" sysfs.path = "/devices/pci0000:00/0000:00:12.0/usb1/1-1/1-1.4/1-1.4:1.2/sound/card1" udev.id = "usb-046d_0804_0B17FC60-02" device.bus = "usb" device.vendor.id = "046d" device.vendor.name = "<vendor>" device.product.id = "0804" device.product.name = "<name>" device.serial = "046d_0804_0B17FC60" device.form_factor = "webcam" device.string = "hw:1" device.buffering.buffer_size = "192000" device.buffering.fragment_size = "96000" device.access_mode = "mmap+timer" device.profile.name = "mono-fallback" device.profile.description = "Mono" device.description = "<description>" module-udev-detect.discovered = "1" device.icon_name = "camera-web-usb" ports: analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown) properties: device.icon_name = "audio-input-microphone" active port: <analog-input-mic> One part of what am trying to do is create a virtual microphone that takes input from speech-dispatcher-espeak-ng persistently or route output from speech-dispatcher-espeak-ng to the microphone, so that when capture the microphone the output will be from speech-dispatcher. How to achieve that? On Tue, Sep 8, 2020 at 9:24 PM Sean Greenslade <sean@xxxxxxxxxxxxxxxxxx> wrote: > > On Mon, Sep 07, 2020 at 05:03:59PM -0700, guest271314 wrote: > > > I doubt that will be possible. Pavucontrol makes use of the native > > > pulseaudio APIs, which are not exposed to javascript. > > > > If mpv can be embedded in an HTML document > > https://github.com/Kagami/mpv.js it should be possible to embed > > pavucontrol or pavucontrol-qt > > (https://doc.qt.io/qt-5/qtwebchannel-javascript.html ; > > https://medium.com/@petar.koretic/why-you-should-use-qt-qml-for-you-next-cross-platform-application-part-1-desktop-5e6d8856b7b4) > > in particular in a browser; for example using WebAssembly; WASI; > > Native Messaging; at least the ability to control Recording tab (-t 2) > > from JavaScript or an HTML form. > > Electron is a whole separate discussion. Electron applications have > different levels of access to the system than the vanilla chromium > browser. > > > > After reading through those bug reports and related issue links, it's > > > pretty clear that this is not a use case that they are particularly > > > interested in supporting. > > > > The majority of own repositories at github are dedicated to fixing > > WontFix; supporting SSML parsing; variable width and height video > > capture with ability to merge multiple tracks into a single media > > container, or stream media without a container; streaming audio > > potentially infinite input streams, that is, a dynamic Internet radio > > station; capturing speech synthesis output from Web Speech API, which > > is the origin of this use case of capturing system audio output. > > > > > May I perhaps suggest using a different > > > browser? Firefox had no problem with monitor inputs last time I > > > checked. > > > > Interestingly, am completing testing of another workaround where since > > Firefox does capture monitor devices, a new, dedicated instance of > > Nightly is started prior to launching Chromium, the MediaStreamTrack > > and MediaStream therefrom are added to a WebRTC RTCPeerConnection, and > > currently using clipboard for signaling which is not ideal though is > > one way to exchange text data between different browsers, accessing > > the monitor device at Chromium instance generated at Nightly > > https://gist.github.com/guest271314/04a539c00926e15905b86d05138c113c. > > That approach avoids writing and reading raw PCM to memory. > > With the amount of work you're seeming to have put into workarounds, > maybe you would be better off writing a proper native application? > > > > No idea, I've never done it myself. The example listed in the online > > > docs shows a simple stereo swap, so you could presumably adapt it by > > > switching the channels to be non-swapped (and of course substitute your > > > specific master source name). > > > > Not sure precisely how to begin. > > > > Am still trying to gather the specific commands in code that > > pavucontrol uses when setting the stream at the UI. Am not certain > > what to pass to pactl move-source-output at what time > > https://gitlab.freedesktop.org/pulseaudio/pavucontrol/-/issues/91#note_590795. > > If you read the documentation (e.g. man pactl), it seems pretty > straightforward: > > > move-source-output ID SOURCE > > Move the specified recording stream (identified by its numerical index) to the specified source (identified by its symbolic name or numerical index). > > So you need to find the specific source output index (ID) and your new > target source (SOURCE). I started an audacity session recording my > microphone, and ran the "pactl list source-outputs" command. This gave > me the index. I then ran the "pactl list sources" to find my target > source name. The switch command (for my setup) then looked like this: > > > pactl move-source-output 203 pulse_send_nofx.monitor > > --Sean > > _______________________________________________ > pulseaudio-discuss mailing list > pulseaudio-discuss@xxxxxxxxxxxxxxxxxxxxx > https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss _______________________________________________ pulseaudio-discuss mailing list pulseaudio-discuss@xxxxxxxxxxxxxxxxxxxxx https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
