Como tocar audio em fila no SDL MIxer e C++?

Tenho 4 audios diferentes; quero tocá-los em fila, apertar um botão e tocar o 1; quando acabar de tocar automaticamente ele começar a tocar o 2, depois o 3 e depois o 4…. é possível fazer isso no SDL Mixer?

Eu só consegui através do while, porém, quando entra nele tudo pára, ele interrompe as outras funções… também tentei através das tarefas do Thread.Join…. mas ai é que está, o Join não sabe que a tarefa 1 terminou…. tem que haver um meio do C++ entender que a tarefa foi finalizada; mas acho que ele não reconhece porque o método é de uma biblioteca externa….

void tarefa1() { Mix_PlayChannel(1, som_1, 0);

}

void tarefa2() { Mix_PlayChannel(1, som_2, 0); }

void tocar() {

thread first(tarefa1); thread second(tarefa2);  first.join();    second.join();  

}

Ubuntu 19.04 Surface Pro 4 Core M Mini-displayport audio

I just recently installed 19.04 on my core M surface pro 4 and I can’t seem to get audio to send via the mini-displayport to my monitor. Any ideas? I tried to install the pulse audio volume control but that does not show the monitor as an audio output, just the headphones and internal speakers.

I am using a mini-displayport to hdmi cable that has worked in the past for sending audio and have verified it works with my laptop and the same monitor.

Sound problems on Ubuntu 18.10, Thinkpad X1 Carbon 5th, Intel Corporation Sunrise Point-LP HD Audio, CX8200

System configuration:

  • Thinkpad X1 Carbon 5th Gen.
  • Ubuntu 18.10
  • Sound card: Intel Corporation Sunrise Point-LP HD Audio, CX8200
  • Kernel: 4.18.0-15-generic

The problem:

When playing audio (any source, Youtube, Spotify etc…) , audio sounds well at first (can be for many minutes, sometimes an hour) and all of a sudden the sound becomes metallic, distant, garbled and distorted. This happens on my headphones connected directly to the laptop or playing through the Thinkpad USB-C Dock, playing directly through the speakers is not an option since I’m in an open space office.

I tried the following without success:

  • Modify default.pa/system.pa with: load-module module-udev-detect tsched=0
  • Tried to play directly on Alsa without pulseaudio
  • Tried to play with the volume up and down etc … (sometimes it helps but then get screwed up again after a while)
  • Tried to modify the sample rates in daemon.conf to other values.

Restart helps but again the sound gets distorted after few minutes. It makes my audio listening completely useless on this hardware. And generally speaking, the whole audio experience as compared with Windows is much much less robust 🙁

Some queries on my system:

root:~# uname -r 4.18.0-15-generic  root:~# dmesg | grep audio [    3.250359] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [    3.351782] snd_hda_codec_conexant hdaudioC0D0: CX8200: BIOS auto-probing. [    3.352919] snd_hda_codec_conexant hdaudioC0D0: autoconfig for CX8200: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [    3.352921] snd_hda_codec_conexant hdaudioC0D0:    speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [    3.352923] snd_hda_codec_conexant hdaudioC0D0:    hp_outs=1 (0x16/0x0/0x0/0x0/0x0) [    3.352924] snd_hda_codec_conexant hdaudioC0D0:    mono: mono_out=0x0 [    3.352925] snd_hda_codec_conexant hdaudioC0D0:    inputs: [    3.352927] snd_hda_codec_conexant hdaudioC0D0:      Internal Mic=0x1a [    3.352929] snd_hda_codec_conexant hdaudioC0D0:      Mic=0x19 [    3.356819] snd_hda_codec_conexant hdaudioC0D0: Enable sync_write for stable communication [    5.292555] usbcore: registered new interface driver snd-usb-audio  root:~# lspci -v | grep -i audio 00:1f.3 Audio device: Intel Corporation Sunrise Point-LP HD Audio (rev 21)     Subsystem: Lenovo Sunrise Point-LP HD Audio      root:~# pacmd list-sinks 2 sink(s) available.   * index: 0     name: <alsa_output.usb-Lenovo_ThinkPad_USB-C_Dock_Audio_000000000000-00.analog-stereo>     driver: <module-alsa-card.c>     flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL LATENCY      state: IDLE     suspend cause: (none)     priority: 9049     volume: front-left: 11106 /  17%,   front-right: 11106 /  17%             balance 0.00     base volume: 65536 / 100%     volume steps: 51     muted: no     current latency: 102.52 ms     max request: 18 KiB     max rewind: 18 KiB     monitor source: 0     sample spec: s16le 2ch 48000Hz     channel map: front-left,front-right                  Stereo     used by: 0     linked by: 1     fixed latency: 100.00 ms     card: 0 <alsa_card.usb-Lenovo_ThinkPad_USB-C_Dock_Audio_000000000000-00>     module: 7     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 = "ThinkPad USB-C Dock Audio"         alsa.long_card_name = "Lenovo ThinkPad USB-C Dock Audio at usb-0000:3c:00.0-1.4.2.3, full speed"         alsa.driver_name = "snd_usb_audio"         device.bus_path = "pci-0000:3c:00.0-usb-0:1.4.2.3:1.0"         sysfs.path = "/devices/pci0000:00/0000:00:1d.0/0000:06:00.0/0000:07:02.0/0000:3c:00.0/usb3/3-1/3-1.4/3-1.4.2/3-1.4.2.3/3-1.4.2.3:1.0/sound/card1"         udev.id = "usb-Lenovo_ThinkPad_USB-C_Dock_Audio_000000000000-00"         device.bus = "usb"         device.vendor.id = "17ef"         device.vendor.name = "Lenovo"         device.product.id = "3063"         device.product.name = "ThinkPad USB-C Dock Audio"         device.serial = "Lenovo_ThinkPad_USB-C_Dock_Audio_000000000000"         device.string = "front:1"         device.buffering.buffer_size = "19200"         device.buffering.fragment_size = "4800"         device.access_mode = "mmap"         device.profile.name = "analog-stereo"         device.profile.description = "Analog Stereo"         device.description = "ThinkPad USB-C Dock Audio Analog Stereo"         alsa.mixer_name = "USB Mixer"         alsa.components = "USB17ef:3063"         module-udev-detect.discovered = "1"         device.icon_name = "audio-card-usb"     ports:         analog-output: Analog Output (priority 9900, latency offset 0 usec, available: unknown)             properties:      active port: <analog-output>     index: 1     name: <alsa_output.pci-0000_00_1f.3.analog-stereo>     driver: <module-alsa-card.c>     flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL LATENCY      state: RUNNING     suspend cause: (none)     priority: 9039     volume: front-left: 63624 /  97%,   front-right: 63624 /  97%             balance 0.00     base volume: 65536 / 100%     volume steps: 75     muted: no     current latency: 100.90 ms     max request: 18 KiB     max rewind: 18 KiB     monitor source: 2     sample spec: s16le 2ch 48000Hz     channel map: front-left,front-right                  Stereo     used by: 1     linked by: 3     fixed latency: 100.00 ms     card: 1 <alsa_card.pci-0000_00_1f.3>     module: 8     properties:         alsa.resolution_bits = "16"         device.api = "alsa"         device.class = "sound"         alsa.class = "generic"         alsa.subclass = "generic-mix"         alsa.name = "CX8200 Analog"         alsa.id = "CX8200 Analog"         alsa.subdevice = "0"         alsa.subdevice_name = "subdevice #0"         alsa.device = "0"         alsa.card = "0"         alsa.card_name = "HDA Intel PCH"         alsa.long_card_name = "HDA Intel PCH at 0xec340000 irq 134"         alsa.driver_name = "snd_hda_intel"         device.bus_path = "pci-0000:00:1f.3"         sysfs.path = "/devices/pci0000:00/0000:00:1f.3/sound/card0"         device.bus = "pci"         device.vendor.id = "8086"         device.vendor.name = "Intel Corporation"         device.product.id = "9d71"         device.product.name = "Sunrise Point-LP HD Audio"         device.form_factor = "internal"         device.string = "front:0"         device.buffering.buffer_size = "19200"         device.buffering.fragment_size = "4800"         device.access_mode = "mmap"         device.profile.name = "analog-stereo"         device.profile.description = "Analog Stereo"         device.description = "Built-in Audio Analog Stereo"         alsa.mixer_name = "Conexant CX8200"         alsa.components = "HDA:14f12008,17aa224f,00100103 HDA:8086280b,80860101,00100000"         module-udev-detect.discovered = "1"         device.icon_name = "audio-card-pci"     ports:         analog-output-speaker: Speakers (priority 10000, latency offset 0 usec, available: no)             properties:                 device.icon_name = "audio-speakers"         analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: yes)             properties:                 device.icon_name = "audio-headphones"     active port: <analog-output-headphones> 

I tried to look all over the internet but nothing.

Help!!! Anyone got a clue ?

convert youtube audio to wav for linux without ffmpeg or avconv

Is there a simple command-line utility for unix available to use to convert an .m4a audio file or .webm file (that I can generate from youtube-dl) to a wave (.wav) file?

I tried ffmpeg and it gives me errors when I tried converting to any output format and it also creates zero-byte files.

I also don’t have avconv but when I looked at its documentation online I think it wants to install the same libraries ffmpeg already installed.

But rather than installing an elaborate package (as I am low on disk space), isn’t there a simple utility that can do this task for unix?

Low USB headset output quality when recording audio

I dumped the ALSA soundcard output, there was no noticeable drop in quality. after the recording stopped. During recording, audio output is choppy and low volume. This is my pulseaudio debug-level logs:

I: [pulseaudio] client.c: Created 1 "Native client (UNIX socket client)" D: [pulseaudio] protocol-native.c: Protocol version: remote 32, local 32 I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: [pulseaudio] protocol-native.c: SHM possible: yes D: [pulseaudio] protocol-native.c: Negotiated SHM: yes D: [pulseaudio] protocol-native.c: Memfd possible: yes D: [pulseaudio] protocol-native.c: Negotiated SHM type: shared memfd D: [pulseaudio] memblock.c: Using shared memfd memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: [pulseaudio] srbchannel.c: SHM block is 65472 bytes, ringbuffer capacity is 2 * 32712 bytes D: [pulseaudio] protocol-native.c: Enabling srbchannel... D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for pavucontrol D: [pulseaudio] module-augment-properties.c: Found /usr/share/applications/pavucontrol.desktop. D: [pulseaudio] conf-parser.c: Parsing configuration file '/usr/share/applications/pavucontrol.desktop' D: [pulseaudio] protocol-native.c: Client enabled srbchannel. D: [pulseaudio] module-stream-restore.c: Not restoring device for stream source-output-by-application-id:org.PulseAudio.pavucontrol, because already set D: [pulseaudio] module-intended-roles.c: Not setting device for stream Peak detect, because already set. D: [pulseaudio] source-output.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "25"  format.channels = "1"  format.channel_map = "\"mono\"" I: [pulseaudio] source-output.c: Trying to change sample rate D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.usb-1395_PXC_550-00.iec958-stereo becomes busy, resuming. D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.usb-1395_PXC_550-00.iec958-stereo is 0x0000, resuming D: [pulseaudio] reserve-wrap.c: Successfully acquired reservation lock on device 'Audio1' I: [alsa-sink-USB Audio] alsa-sink.c: Trying resume... I: [alsa-sink-USB Audio] alsa-util.c: Cannot disable ALSA period wakeups D: [alsa-sink-USB Audio] alsa-util.c: Maximum hw buffer size is 5461 ms D: [alsa-sink-USB Audio] alsa-util.c: Set buffer size first (to 96000 samples), period size second (to 48000 samples). I: [alsa-sink-USB Audio] alsa-util.c: ALSA period wakeups were not disabled D: [alsa-sink-USB Audio] alsa-sink.c: hwbuf_unused=0 D: [alsa-sink-USB Audio] alsa-sink.c: setting avail_min=95119 I: [alsa-sink-USB Audio] alsa-sink.c: Time scheduling watermark is 18.38ms I: [alsa-sink-USB Audio] alsa-sink.c: Resumed successfully... D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.usb-1395_PXC_550-00.iec958-stereo becomes idle, timeout in 5 seconds. I: [alsa-sink-USB Audio] alsa-sink.c: Starting playback. D: [alsa-sink-USB Audio] ratelimit.c: 702 events suppressed D: [alsa-sink-USB Audio] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink-USB Audio] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.usb-1395_PXC_550-00.iec958-stereo becomes idle, timeout in 5 seconds. D: [pulseaudio] resampler.c: Resampler: D: [pulseaudio] resampler.c:   rate 48000 -> 25 (method peaks) D: [pulseaudio] resampler.c:   format s16le -> float32le (intermediate s16le) D: [pulseaudio] resampler.c:   channels 2 -> 1 (resampling 1) D: [pulseaudio] resampler.c: Channel matrix: D: [pulseaudio] resampler.c:        I00   I01  D: [pulseaudio] resampler.c:     +------------ D: [pulseaudio] resampler.c: O00 | 0.500 0.500 I: [pulseaudio] remap.c: Using stereo to mono remapping D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: [pulseaudio] source-output.c: Created output 0 "Peak detect" on alsa_output.usb-1395_PXC_550-00.iec958-stereo.monitor with sample spec float32le 1ch 25Hz and channel map mono I: [pulseaudio] source-output.c:     media.name = "Peak detect" I: [pulseaudio] source-output.c:     application.name = "PulseAudio Volume Control" I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client" I: [pulseaudio] source-output.c:     native-protocol.version = "32" I: [pulseaudio] source-output.c:     application.id = "org.PulseAudio.pavucontrol" I: [pulseaudio] source-output.c:     application.icon_name = "audio-card" I: [pulseaudio] source-output.c:     application.version = "3.0" I: [pulseaudio] source-output.c:     application.process.id = "31855" I: [pulseaudio] source-output.c:     application.process.user = "p" I: [pulseaudio] source-output.c:     application.process.host = "desktop" I: [pulseaudio] source-output.c:     application.process.binary = "pavucontrol" I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8" I: [pulseaudio] source-output.c:     window.x11.display = ":1" I: [pulseaudio] source-output.c:     application.process.machine_id = "6eacec97385842c2a8c31dfb7da6426d" I: [pulseaudio] source-output.c:     application.process.session_id = "1" I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol" D: [alsa-sink-USB Audio] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink-USB Audio] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0 D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0 I: [pulseaudio] protocol-native.c: Final latency 80.00 ms = 40.00 ms + 40.00 ms D: [alsa-sink-USB Audio] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink-USB Audio] alsa-sink.c: Latency set to 40.00ms D: [alsa-sink-USB Audio] alsa-sink.c: hwbuf_unused=376320 D: [alsa-sink-USB Audio] alsa-sink.c: setting avail_min=95119 D: [alsa-sink-USB Audio] alsa-sink.c: Requesting rewind due to latency change. D: [alsa-sink-USB Audio] alsa-sink.c: Latency set to 40.00ms D: [alsa-sink-USB Audio] alsa-sink.c: hwbuf_unused=376320 D: [alsa-sink-USB Audio] alsa-sink.c: setting avail_min=95119 D: [alsa-sink-USB Audio] alsa-sink.c: Requested to rewind 384000 bytes. D: [alsa-sink-USB Audio] alsa-sink.c: Limited to 383744 bytes. D: [alsa-sink-USB Audio] alsa-sink.c: before: 95936 D: [alsa-sink-USB Audio] alsa-sink.c: after: 95936 D: [alsa-sink-USB Audio] alsa-sink.c: Rewound 383744 bytes. D: [alsa-sink-USB Audio] sink.c: Processing rewind... D: [alsa-sink-USB Audio] source.c: Processing rewind... D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] module-stream-restore.c: Not restoring device for stream source-output-by-application-id:org.PulseAudio.pavucontrol, because already set D: [pulseaudio] module-intended-roles.c: Not setting device for stream Peak detect, because already set. D: [pulseaudio] source-output.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "25"  format.channels = "1"  format.channel_map = "\"mono\"" I: [pulseaudio] source-output.c: Trying to change sample rate D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.usb-1395_PXC_550-00.analog-mono becomes busy, resuming. D: [pulseaudio] source.c: Suspend cause of source alsa_input.usb-1395_PXC_550-00.analog-mono is 0x0000, resuming I: [alsa-source-USB Audio] alsa-source.c: Trying resume... I: [alsa-source-USB Audio] alsa-util.c: Cannot disable ALSA period wakeups D: [alsa-source-USB Audio] alsa-util.c: Maximum hw buffer size is 32768 ms D: [alsa-source-USB Audio] alsa-util.c: Set buffer size first (to 32000 samples), period size second (to 16000 samples). I: [alsa-source-USB Audio] alsa-util.c: ALSA period wakeups were not disabled D: [alsa-source-USB Audio] alsa-source.c: hwbuf_unused=0 D: [alsa-source-USB Audio] alsa-source.c: setting avail_min=30237 I: [alsa-source-USB Audio] alsa-source.c: Time scheduling watermark is 110.25ms I: [alsa-source-USB Audio] alsa-source.c: Resumed successfully... I: [alsa-source-USB Audio] alsa-source.c: Starting capture. D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.usb-1395_PXC_550-00.analog-mono becomes idle, timeout in 5 seconds. D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.usb-1395_PXC_550-00.analog-mono becomes idle, timeout in 5 seconds. D: [pulseaudio] resampler.c: Resampler: D: [pulseaudio] resampler.c:   rate 16000 -> 25 (method peaks) D: [pulseaudio] resampler.c:   format s16le -> float32le (intermediate s16le) D: [pulseaudio] resampler.c:   channels 1 -> 1 (resampling 1) D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=2, prebuf=0, minreq=1 maxrewind=0 D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=2, prebuf=0, minreq=2 maxrewind=0 I: [pulseaudio] source-output.c: Created output 1 "Peak detect" on alsa_input.usb-1395_PXC_550-00.analog-mono with sample spec float32le 1ch 25Hz and channel map mono I: [pulseaudio] source-output.c:     media.name = "Peak detect" I: [pulseaudio] source-output.c:     application.name = "PulseAudio Volume Control" I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client" I: [pulseaudio] source-output.c:     native-protocol.version = "32" I: [pulseaudio] source-output.c:     application.id = "org.PulseAudio.pavucontrol" I: [pulseaudio] source-output.c:     application.icon_name = "audio-card" I: [pulseaudio] source-output.c:     application.version = "3.0" I: [pulseaudio] source-output.c:     application.process.id = "31855" I: [pulseaudio] source-output.c:     application.process.user = "p" I: [pulseaudio] source-output.c:     application.process.host = "desktop" I: [pulseaudio] source-output.c:     application.process.binary = "pavucontrol" I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8" I: [pulseaudio] source-output.c:     window.x11.display = ":1" I: [pulseaudio] source-output.c:     application.process.machine_id = "6eacec97385842c2a8c31dfb7da6426d" I: [pulseaudio] source-output.c:     application.process.session_id = "1" I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol" D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0 D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0 I: [pulseaudio] protocol-native.c: Final latency 80.00 ms = 40.00 ms + 40.00 ms D: [alsa-source-USB Audio] alsa-source.c: latency set to 40.00ms D: [alsa-source-USB Audio] alsa-source.c: hwbuf_unused=62720 D: [alsa-source-USB Audio] alsa-source.c: setting avail_min=161 D: [alsa-source-USB Audio] alsa-source.c: latency set to 40.00ms D: [alsa-source-USB Audio] alsa-source.c: hwbuf_unused=62720 D: [alsa-source-USB Audio] alsa-source.c: setting avail_min=161 D: [alsa-source-USB Audio] alsa-source.c: Requested volume: mono: 65536 / 100% / 0.00 dB D: [alsa-source-USB Audio] alsa-source.c: Got hardware volume: mono: 65536 / 100% / 0.00 dB D: [alsa-source-USB Audio] alsa-source.c: Calculated software volume: mono: 65536 / 100% / 0.00 dB (accurate-enough=yes) D: [alsa-source-USB Audio] source.c: Volume not changing D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-USB Audio] alsa-sink.c: Wakeup from ALSA! E: [pulseaudio] bluez5-util.c: GetManagedObjects() failed: org.freedesktop.DBus.Error.NoReply: Did not receive a reply. Possible causes include: the remote application did not send a reply, the message bus security policy blocked the reply, the reply timeout expired, or the network connection was broken.  

I can’t tell if anything relevant is happening here. There is no log output from ALSA in any files from /var/log or dmesg.

My best guess is line interference over the USB connection? The headset has no problems over bluetooth.

FTT audio processing for an audio spectrum audio spectrum

I want to make a fast and responsive audio visualizer (in real time). I want to use the GPU for the Fast Fourier Transform because almost everyone that has Steam installed has an APU/GPU with acceleration for this workload. I mentioned Steam because I want to make the Visualizer for a Programm called Wallpaper Engine. The integration will happen later when the audio spectrum is ready.

I have some Questions before I can start coding:

1. Library to listen to the soundcard

I want that the visualizer to listen actively that is played regardless of where the sound comes from. I am looking for a fast library with low latency. Does somebody know a good library?

2. Cuda FFT

Because I have an Nvidia GPU want to use cuFFT. Later I want to do the same for Intel/AMD Hardware. I have heard that cuFFT is pretty fast, is the processing time much lower than FFT with the CPU?

3. best/fastest solution for displaying graphical the audio spectrum

I have thought about using OpenGL or a Gameengine like Unity. I want to display the audio spectrum in a wave of bars. Are there better solutions for such a use case?

I am looking forward to your suggestions 🙂

Audio issue when HDMI drawing tablet is connected?

I’m running a 13″ Macbook pro 2017 with operating system Mojave 10.14.4.

I have an HDMI connected tablet (Huion Gt-220). When I connect my drawing tablet to my computer, it locks out my sound for the computer.

I’ve tried going into System Preferences and changing the output to internal speakers. I’ve gone into the Audio MIDI setup and tried to configure it from there as well.

In the audio MIDI setup I’ve also tried making a multi output and an aggregated device.

I’ve tried it with and without headphones. I can’t get it to work for the life of me. Does anyone else have any ideas on how to fix this? I’m dying here because I’ll need to watch/listen to classes at the same time as I work and I can’t seem to set it up.

Audio not working on Windows 10 installed on external SSD running on iMac

I installed Windows 10 (April 18 update) on external SSD drive that can be selected on the iMac’s startup. Run the Boot Camp Windows Support Software and everything is fine, Apple mouse & keyboard, except audio is missing. Windows 10 Device Manager app does not show any problems with the drivers such as yellow triangles or red crosses. In fact, when a video is played the green volumen bars in the audio device driver properties move but there is no sound at all.

  • iMac is the latest 2019 model.
  • Windows 10 is up to date according to Windows update app (had to use April 18 update as start version since the installation does not work with October 18 version).
  • Drivers are update to date according to Windows.
  • Installed Windows 10 on external SDD following these instructions: https://blog.macsales.com/40947-tech-tip-how-to-use-boot-camp-on-an-external-drive, https://medium.com/@svenkirsime/install-windows-on-the-external-ssd-hdd-for-your-mac-5d29eefe5d1, https://www.youtube.com/watch?v=910Y1hLreRc&t=117s.