mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1450851095
Both the source and the sink elements were broken in a number of ways: * prepare() was assuming that the format was always S16LE 2ch 44.1KHz. We now probe the preferred format with GetMixFormat(). * Device initialization was done with the wrong buffer size (buffer_time is in microseconds, not nanoseconds). * sink_write() and src_read() were just plain wrong and would never write or read anything useful. * Some functions in prepare() were always returning FALSE which meant trying to use the elements would *always* fail. * get_caps() and delay() were not implemented at all. TODO: support for >2 channels TODO: pro-audio low-latency TODO: SPDIF and other encoded passthroughs Three new properties are now implemented: role, mute, and device. * 'role' designates the stream role of the initialized device, see: https://msdn.microsoft.com/en-us/library/windows/desktop/dd370842(v=vs.85).aspx * 'device' is a system-wide GUIDesque string for a specific device. * 'mute' is a sink property and simply mutes it. On my Windows 8.1 system, the lowest latency that works is: wasapisrc buffer-time=20000 wasapisink buffer-time=10000 aka, 20ms and 10ms respectively. These values are close to the lowest possible with the IAudioClient interface. Further improvements require porting to IAudioClient2 or IAudioClient3. https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/low-latency-audio |
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| acmenc | ||
| acmmp3dec | ||
| androidmedia | ||
| applemedia | ||
| bluez | ||
| d3dvideosink | ||
| decklink | ||
| directsound | ||
| dshowdecwrapper | ||
| dshowsrcwrapper | ||
| dshowvideosink | ||
| dvb | ||
| fbdev | ||
| ipcpipeline | ||
| kms | ||
| msdk | ||
| nvdec | ||
| nvenc | ||
| opensles | ||
| shm | ||
| tinyalsa | ||
| uvch264 | ||
| vcd | ||
| vdpau | ||
| wasapi | ||
| winks | ||
| winscreencap | ||
| Makefile.am | ||
| meson.build | ||