I'm trying to record audio from a microphone but I got below error. I tried plughw:0,4 to get automatic sample format conversion and also given manually as arecord -D hw:0,4 -d 5 -f S32_LE -r 48000 test.wav, but I get same error.
When I run alsamixer I get this error as ALSA lib ../../alsa-plugins-1.2.1/pulse/pulse.c:242:(pulse_connect) PulseAudio: Unable to connect: Connection refused cannot open mixer: Connection refused. When I manually start pulseaudio --start alsamixer runs, but when I do some microphone related operation alsamixer repeats same error again.
arecord:
Available formats:
- S16_LE
- S24_LE
- S32_LE
Recording WAVE 'test.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Mono
arecord: set_params:1405: Unable to install hw params:
ACCESS: RW_INTERLEAVED
FORMAT: S32_LE
SUBFORMAT: STD
SAMPLE_BITS: 32
FRAME_BITS: 32
CHANNELS: 1
RATE: 48000
PERIOD_TIME: 125000
PERIOD_SIZE: 6000
PERIOD_BYTES: 24000
PERIODS: 4
BUFFER_TIME: 500000
BUFFER_SIZE: 4000
BUFFER_BYTES: 96000
TICK_TIME: 0
Related
I am using arecord to record audio using a USB microphone that allows recording up to 384k sampling rate. With Audacity I can easily record with other lower sampling rates (48k, 44.1k ..) but when I use:
arecord -vD hw:4,0 -f S16_LE -d 120 -r 48000 -c 1 sample.wav
It displays a warning message:
Warning: rate is not accurate (requested = 48000Hz, got = 384000Hz)
And records with 384000Hz.
I can't get to fix this.
In raspberry pi I've following i2s microphone breakout board and use it like the guide suggested. When I try record audio from it using ffmpeg to the file with ffmpeg -f alsa -i dmic_sv out.wav command. I'll receive following error
[alsa # 0x22e21c0] cannot set sample format 0x10000 2 (Invalid argument)
dmic_sv: Input/output error
When I specify the used codec explicitly with -acodec it works fine:
ffmpeg -f alsa -acodec pcm_s32le -i dmic_sv out.wav
And from the output ffmpeg will reencode to pcm_s16le
Input #0, alsa, from 'dmic_sv':
Duration: N/A, start: 1597597938.887969, bitrate: 3072 kb/s
Stream #0:0: Audio: pcm_s32le, 48000 Hz, stereo, s32, 3072 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s32le (native) -> pcm_s16le (native))
How I can tell ffmpeg to use signed 32-bit little endian by default without specifying it explicitly? And where ffmpeg gets this default and can I configure it somehow?
I figured this one out myself by reading ffmpeg source code. It seems when codec is not specified and alsa device is used. FFmpeg will default to pcm 16-bit samples instead. Code to set the default here and the default macro here.
Trying to create a simple command line player for .dsf (DSD audio) files, and output to an alsa device that supports up to 24-bit 192 kHz sample rate. The following command almost works and it does play the track. Examining the bold text below, the dsf input file is converted to 24-bit/192 kHz, but the output is then truncated to 16-bit 192 kHz (pcm_s16le i.e, 16 bit little endian).
ffmpeg -i '01 - Sweet Georgia Brown.dsf' -f alsa hw:0,0
After displaying the ffmpeg banner and song metadata (tags), here is the result, bold is my emphasis:
Duration: 00:05:14.83, start: 0.000000, bitrate: 9234 kb/s
Stream #0:0: Audio: flac, 192000 Hz, stereo, s32 (24 bit)
Stream mapping:
Stream #0:0 -> #0:0 (flac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, alsa, to 'hw:0,0':
Since I can play this and many other tracks at full resolution using another player (foobar2000) it seems there might be an option in the encoder which is part of FFmpeg: Lavf57.83.100 I can find no information in any of the FFmpeg documentation that helps. Tried finding options in FFplay and even guessing using other FFmpeg options like this example.
ffmpeg -sample_fmt s24 -i '01 - Sweet Georgia Brown.dsf' -f alsa hw:0,0 ***** same results.
I'm stuck. Any suggestions?
Environment: Linux Mint 19.2, 64-bit, ASUS Xonar STXii sound card.
Each output format or device has a default encoder registered for each media type it accepts. ALSA accepts audio and its default encoder is 16-bit signed PCM.
You can change the encoder by specifying one.
ffmpeg -i '01 - Sweet Georgia Brown.dsf' -c:a pcm_s24le -f alsa hw:0,0
If I run this command line
ffmpeg -ss 0 -t 3600 -i file1.mp3 -ss 0 -t 20 -i file2.mp3 -filter_complex "[0][1]concat=n=2:v=0:a=1" -ac 2 -f wav - > test.wav
I'm basically putting the stout inside a container wav (test.wav) but the duration is always wrong. The output file should be 01:00:20.00 but if I play it on VLC (or any player audio) it shows 06:12:49.00 and even if I change the start_times, the durations and number of files, I still get that timecode out. The even weirder thing is that ffprobe shows the duration as it should be. Can somebody please help me on this?
UPDATE:
[wav # 0000000000cf3680] Ignoring maximum wav data size, file may be invalid
[wav # 0000000000cf3680] Estimating duration from bitrate, this may be inaccurate
Input #0, wav, from 'test.wav':
Metadata:
encoder : Lavf57.72.101
timecode : 01:00:20.00
Duration: 01:00:20.00, bitrate: 1536 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s
That is what the ffprobe on the output shows..the duration is correct here but not on any audio player
I want to convert a MOV from my Casio cam to mp4 using transcode. Why transcode? Because I also want to deshake the video in the same step.
When I use
ffmpeg -i in.MOV out.mp4
it works. When using
transcode -J stabilize -i in.MOV
or
transcode -J transform -i in.MOV -y ffmpeg -F mpeg4 -o out.mp4
I get hundreds of these errors:
[ffmpeg_audio] Error: avcodec_open2 failed
[adpcm_ima_wav # 0x1f7f180] Only 4-bit ADPCM IMA WAV files are supported
This looks to me as if transcode uses ffmpeg internally.
I could use ffmpeg to make it mp4 first and then use transcode to stabilize the video, but then it would be re-encoded twice which I would like to avoid.
This is what mplayer says about my MOV file:
MPlayer2 2.0-701-gd4c5b7f-2ubuntu2 (C) 2000-2012 MPlayer Team
Cannot open file '/home/koem/.mplayer/input.conf': No such file or directory
Failed to open /home/koem/.mplayer/input.conf.
Cannot open file '/etc/mplayer/input.conf': No such file or directory
Failed to open /etc/mplayer/input.conf.
Playing 1-original.MOV.
Detected file format: QuickTime / MOV (libavformat)
[lavf] stream 0: video (h264), -vid 0
[lavf] stream 1: audio (adpcm_ima_wav), -aid 0, -alang eng
Clip info:
major_brand: qt
minor_version: 537921536
compatible_brands: qt caqv
creation_time: 2017-01-02 23:31:38
Load subtitles in .
Failed to open VDPAU backend libvdpau_i965.so: cannot open shared object file: No such file or directory
[vdpau] Error when calling vdp_device_create_x11: 1
[ass] auto-open
Selected video codec: H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 [libavcodec]
Selected audio codec: ADPCM IMA WAV [libavcodec]
AUDIO: 44100 Hz, 2 ch, s16le, 352.8 kbit/25.00% (ratio: 44100->176400)
AO: [pulse] 44100Hz 2ch s16le (2 bytes per sample)
Starting playback...
VIDEO: 1920x1080 29.970 fps 15940.0 kbps (1992.5 kB/s)
VO: [xv] 1920x1080 => 1920x1080 Planar YV12
Colorspace details not fully supported by selected vo.
A: 1.1 V: 1.1 A-V: -0.000 ct: 0.000 0/ 0 16% 8% 1.6% 0 0
Exiting... (Quit)
How can I make it work with transcode without using ffmpeg first?
FFmpeg has a deshake as well as a stabilization filter. Get a new binary if yours doesn't.
To continue with your existing binaries, run
ffmpeg -i in.MOV -vcodec copy out.mp4
This will skip video re-encoding.