Closed dttlgotv closed 6 years ago
What is your registry values in HKEY_CURRENT_USER\ SOFTWARE\virtual_audio_capture"
On Tue, May 5, 2015 at 11:54 PM, dttlgotv notifications@github.com wrote:
I used ffmpeg command line to record the speaker audio with it well. But after I write ffmpeg codes to record the speaker sound, it listen badly.
So can you give me some hint? What is the endian used in it?
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15.
I just call "regsvr32 %~dp0\audio_sniffer.dll". And ffmpeg command can run well, the recorded speaker sound is well. But I write some ffmpeg codes to use it, the sound is bad.
the values is below default: not set last_output :v. 0.4.6 total reads 2868 total discontinuity blips 0, total overflows 0
Does it record microphone ok? what os?
On Wed, May 6, 2015 at 5:05 AM, dttlgotv notifications@github.com wrote:
I just call "regsvr32 %~dp0\audio_sniffer.dll". And ffmpeg command can run well, the recorded speaker sound is well. But I write some ffmpeg codes to use it, the sound is bad.
the values is below default: not set last_output :v. 0.4.6 total reads 2868 total discontinuity blips 0, total overflows 0
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-99419139 .
Yes. It record microphone well.
My os is windows 7.
ffmpeg records it ok? could you send me the console output of a working good ffmpeg run please?
On Wed, May 6, 2015 at 7:52 PM, dttlgotv notifications@github.com wrote:
Yes. It record microphone well.
My os is windows 7.
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-99677349 .
C:\Users\guoxinghua>ffmpeg -f dshow -i audio="virtual-audio-capturer" -acodec li bopus gxh.opus ffmpeg version 2.5.2 Copyright (c) 2000-2014 the FFmpeg developers built on Dec 30 2014 17:18:09 with gcc 4.9.2 (GCC) configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --ena ble-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --e nable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-lib gsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencor e-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enabl e-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-l ibtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable- libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-l ibwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --ena ble-lzma --enable-decklink --enable-zlib libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.102 / 56. 15.102 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, dshow, from 'audio=virtual-audio-capturer': Duration: N/A, start: 16258.072000, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s File 'gxh.opus' already exists. Overwrite ? [y/N] y [libopus @ 00670940] No bit rate set. Defaulting to 96000 bps. Output #0, opus, to 'gxh.opus': Metadata: encoder : Lavf56.15.102 Stream #0:0: Audio: opus (libopus), 48000 Hz, stereo, s16, 96 kb/s Metadata: encoder : Lavc56.13.100 libopus Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus)) Press [q] to stop, [?] for help size= 141kB time=00:00:14.69 bitrate= 78.8kbits/s
expert: any response?
OK registry values after a run with ffmpeg and then run with your own?
On Tue, May 12, 2015 at 6:05 AM, dttlgotv notifications@github.com wrote:
expert: any response?
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-101252782 .
yes.
(what are they both times) Also when you capture the microphone vs. the virtual audio, what is the HZ for each (like 44100hz)?
On Wed, May 13, 2015 at 7:58 PM, dttlgotv notifications@github.com wrote:
yes.
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-101880245 .
same hz, 48000
shoot I don't know then, sorry about that! If you figure it out let me know. How does the audio sound again? patchy?
On Thu, May 14, 2015 at 8:12 PM, dttlgotv notifications@github.com wrote:
same hz, 48000
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-102223749 .
Sometimes there is some noise in the recorded audio. I am sad that the ffmpeg command line can work well use same capture sample rate.
Any advice?
It seems that (32000 and 48000 sample rate, 16bits) are well in my app. But (44100, is bad, the captured audio has noise. Any captured audio from 24bits out devices are bad..
Any ffmpeg function needed to call?
full command line and uncut console output please?
On Fri, Jun 5, 2015 at 4:24 AM, dttlgotv notifications@github.com wrote:
It seems that 32000 and 48000 are well in my app. But 44100 is bad, the captured audio has noise.
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-109251994 .
Sorry for late response.
I clarify the question again. I use this dll to capture audio , and the result is below:
These results are got by my app. I do not know what is different about 44100 with other. Do you know?
And in the other side, I use ffmepg command line to test, all are good.
Please give me some advice. Thanks a lot.
is your "sound system" 48000 ? (like normal playback?)
On Tue, Jun 9, 2015 at 10:09 PM, dttlgotv notifications@github.com wrote:
Sorry for late response.
I clarify the question again. I use this dll to capture audio , and the result is below:
- sound card out parameter, sample rate 44100, channels 2, the audio got from av_read_frame has noise. it is bad
- sound card out parameter, sample rate 48000, channels 2, the audio got from av_read_frame has no noise, it is well.
- sound card out parameter, sample rate 96000, channels 2, the audio got from av_read_frame has no noise, it is well.
These results are got by my app. I do not know what is different about 44100 with other. Do you know?
And in the other side, I use ffmepg command line to test, all are good.
Please give me some advice. Thanks a lot.
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-110582865 .
sound system out sample rate can be set. I use 44100, av_read_frame get noise. And I use 48000, av_read_frame get no noise. It can be seen in one computer. But in the other computer, this issue can not be seen.
I want to know there is any trick about using this component using av_read_frame function.
Hope your answer.
OK does it only get "noise" if the two don't match?
On Fri, Jun 12, 2015 at 8:04 PM, dttlgotv notifications@github.com wrote:
sound system out sample rate can be set. I use 44100, av_read_frame get noise. And I use 48000, av_read_frame get no noise. It can be seen in one computer. But in the other computer, this issue can not be seen.
I want to know there is any trick about using this component using av_read_frame function.
Hope your answer.
— Reply to this email directly or view it on GitHub https://github.com/rdp/virtual-audio-capture-grabber-device/issues/15#issuecomment-111654973 .
Try with the latest, reopen if still getting this prob.
I used ffmpeg command line to record the speaker audio well. command: >ffmpeg -f dshow -i audio="virtual-audio-capturer" gxh.wav
But after I write ffmpeg codes to record the speaker sound, it listen very badly. avformat_open_input and av_read_frame can be called well.
So can you give me some hint? What is the endian used in it?