riscv/qemu - qemu - Gitea: Git with a cup of tea

Author	SHA1	Message	Date
Kővágó, Zoltán	ed2a4a7941	audio: proper support for float samples in mixeng This adds proper support for float samples in mixeng by adding a new audio format for it. Limitations: only native endianness is supported. None of the virtual sound cards support float samples (it looks like most of them only support 8 and 16 bit, only hda supports 32 bit), it is only used for the audio backends (i.e. host side). Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Acked-by: Markus Armbruster <armbru@redhat.com> Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>	6 years ago
Kővágó, Zoltán	8efac073a5	audio: add mixing-engine option (documentation) This will allow us to disable mixeng when we use a decent backend. Disabling mixeng have a few advantages: * we no longer convert the audio output from one format to another, when the underlying audio system would just convert it to a third format. We no longer convert, only the underlying system, when needed. * the underlying system probably has better resampling and sample format converting methods anyway... * we may support formats that the mixeng currently does not support (S24 or float samples, more than two channels) * when using an audio server (like pulseaudio) different sound card outputs will show up as separate streams, even if we use only one backend Disadvantages: * audio capturing no longer works (wavcapture, and vnc audio extension) * some backends only support a single playback stream or very picky about the audio format. In this case we can't disable mixeng. Originally thw two main use cases of the disabled option was: using unsupported audio formats (5.1 and 7.1 audio) and having different pulseaudio streams per audio frontend. Since we can have multiple -audiodevs, the latter is not that important, so currently you only need this option if you want to use 5.1 or 7.1 audio (implemented in a later patch), otherwise it's probably better to stick to the old and tried mixeng, since it's less picky about the backends. The ideal solution would be to port as much as possible to gstreamer, but this is currently out of scope: https://wiki.qemu.org/Internships/ProjectIdeas/AudioGStreamer Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 5765186a7aadd51a72bc7d3e804307f0ee8a34ce.1570996490.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>	7 years ago
Kővágó, Zoltán	f47dffe8d1	audio: paaudio: ability to specify stream name This can be used to identify stream in tools like pavucontrol when one creates multiple -audiodevs or runs multiple qemu instances. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Acked-by: Markus Armbruster <armbru@redhat.com> Message-id: 2d6e337c474ac84172d0809e6959c26b21d48120.1568157545.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>	7 years ago
Martin Schrodt	f614277765	audio/paaudio: prolong and make latency configurable The latency of a connection to the PulseAudio server is determined by the tlength parameter. This was hardcoded to 10ms, which is a bit too tight on my machine, causing audio on host and guest to malfunction. A setting of 15ms works fine here. To allow tweaking, I also made the setting configurable via the new -audiodev config. This allows to squeeze out better timings in scenarios where the emulation allows it. I also removed setting of the minreq parameter to (seemingly arbitrary) half the latency, since it showed worse audio quality during my tests. Allowing PulseAudio to request smaller chunks helped. Signed-off-by: Martin Schrodt <martin@schrodt.org> Message-id: 20190315084653.120020-3-martin@schrodt.org Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>	7 years ago
Kővágó, Zoltán	8c3a7d0087	qapi: qapi for audio backends This patch adds structures into qapi to replace the existing configuration structures used by audio backends currently. This qapi will be the base of the -audiodev command line parameter (that replaces the old environment variables based config). This is not a 1:1 translation of the old options, I've tried to make them much more consistent (e.g. almost every backend had an option to specify buffer size, but the name was different for every backend, and some backends required usecs, while some other required frames, samples or bytes). Also tried to reduce the number of abbreviations used by the config keys. Some of the more important changes: * use `in` and `out` instead of `ADC` and `DAC`, as the former is more user friendly imho * moved buffer settings into the global setting area (so it's the same for all backends that support it. Backends that can't change buffer size will simply ignore them). Also using usecs, as it's probably more user friendly than samples or bytes. * try-poll is now an alsa backend specific option (as all other backends currently ignore it) Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Reviewed-by: Markus Armbruster <armbru@redhat.com> Message-id: 5461b514dbf3e0bc31b0abb6498a9b3a008c271e.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>	7 years ago