Using windows based ffmpeg to convert MP3 to AMR.
For some reason it fails with error as given below.
Don't know how to given the correct parameters for AMR.
C:\Program Files (x86)\AMR to MP3 Converter>ffmpeg -i mfile.mp3 -ar 8000 -ab 12.2k audio.amr
FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers
built on Jan 18 2011 04:07:05 with gcc 4.4.2
configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb
is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg
--enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb
--enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect -
-enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss
l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads --
cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack
libavutil 50.36. 0 / 50.36. 0
libavcore 0.16. 1 / 0.16. 1
libavcodec 52.108. 0 / 52.108. 0
libavformat 52.93. 0 / 52.93. 0
libavdevice 52. 2. 3 / 52. 2. 3
libavfilter 1.74. 0 / 1.74. 0
libswscale 0.12. 0 / 0.12. 0
[mp3 # 003abeb0] max_analyze_duration reached
[mp3 # 003abeb0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'mfile.mp3':
Metadata:
title : SPL_TRACK
artist : Krishnan
album : Lord
genre : Lord
track : 5
Duration: 00:30:06.00, start: 0.000000, bitrate: 127 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, 2 channels, s16, 128 kb/s
[libopencore_amrnb # 017e0d20] Only mono supported
Output #0, amr, to 'audio.amr':
Stream #0.0: Audio: libopencore_amrnb, 8000 Hz, 2 channels, s16, 12 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Error while opening encoder for output stream #0.0 - maybe incorrect parameters
such as bit_rate, rate, width or height
As the error msg says, Only mono supported , so add -ac 1.
In any case, your ffmpeg is prehistoric. Get a new binary from https://ffmpeg.zeranoe.com/builds/
Related
I want to extract a simplified mono low bitrate low frequency and basically low sized audio from any mp4 file. I use the following FFMPEG command but the resulting audio somehow has 2 channels and its bitrate is the same as the original audio from mp4.
ffmpeg.exe -report -y -i "{mp4}" -vn -acodec libmp3lame -ac 1 -ab 64k -ar 24000 -f mp3 output "{mp3}"
Here is my report:
ffmpeg started on 2021-07-17 at 18:30:20
Report written to "ffmpeg-20210717-183020.log"
Log level: 48
Command line:
"C:\\ffmpeg\\bin\\ffmpeg.exe" -report -y -i "C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Video1.mp4" -vn -acodec libmp3lame -ac 1 -ab 64k -f mp3 output "C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Audio.mp3"
ffmpeg version 4.3.1-2021-01-01-full_build-www.gyan.dev Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 10.2.0 (Rev5, Built by MSYS2 project)
configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-libsnappy --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libzvbi --enable-librav1e --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-l libavutil 56. 51.100 / 56. 51.100
libavcodec 58. 91.100 / 58. 91.100
libavformat 58. 45.100 / 58. 45.100
libavdevice 58. 10.100 / 58. 10.100
libavfilter 7. 85.100 / 7. 85.100
libswscale 5. 7.100 / 5. 7.100
libswresample 3. 7.100 / 3. 7.100
libpostproc 55. 7.100 / 55. 7.100
Splitting the commandline.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-i' ... matched as input url with argument 'C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Video1.mp4'.
Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'libmp3lame'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
Reading option '-ab' ... matched as option 'ab' (audio bitrate (please use -b:a)) with argument '64k'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
Reading option 'output' ... matched as output url.
Reading option 'C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Audio.mp3' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Video1.mp4.
Successfully parsed a group of options.
Opening an input file: C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Video1.mp4.
[NULL # 000002e7d956ed40] Opening 'C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Video1.mp4' for reading
[file # 000002e7d956fd80] Setting default whitelist 'file,crypto,data'
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] ISO: File Type Major Brand: mp42
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] Setting codecpar->delay to 1 for stream st: 0
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] Before avformat_find_stream_info() pos: 139797 bytes read:65536 seeks:1 nb_streams:2
[h264 # 000002e7d9570c00] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 # 000002e7d9570c00] nal_unit_type: 8(PPS), nal_ref_idc: 3
[h264 # 000002e7d9570c00] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 # 000002e7d9570c00] Format yuv420p chosen by get_format().
[h264 # 000002e7d9570c00] Reinit context to 1920x1088, pix_fmt: yuv420p
[h264 # 000002e7d9570c00] no picture
[h264 # 000002e7d9570c00] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 2
[h264 # 000002e7d9570c00] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 0
[h264 # 000002e7d9570c00] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 0
[h264 # 000002e7d9570c00] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 0
[h264 # 000002e7d9570c00] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 2
[h264 # 000002e7d9570c00] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 0
[h264 # 000002e7d9570c00] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 0
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] All info found
[mov,mp4,m4a,3gp,3g2,mj2 # 000002e7d956ed40] After avformat_find_stream_info() pos: 158020 bytes read:65536 seeks:1 frames:11
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Video1.mp4':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: mp42mp41
creation_time : 2020-11-24T06:56:49.000000Z
Duration: 00:00:56.45, start: 0.000000, bitrate: 893 kb/s
Stream #0:0(eng), 10, 1/30000: Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080, 749 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
Metadata:
creation_time : 2020-11-24T06:56:49.000000Z
handler_name : Mainconcept Video Media Handler
encoder : AVC Coding
Stream #0:1(eng), 1, 1/44100: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)
Metadata:
creation_time : 2020-11-24T06:56:49.000000Z
handler_name : #Mainconcept MP4 Sound Media Handler
Successfully opened the file.
Parsing a group of options: output url output.
Applying option vn (disable video) with argument 1.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument libmp3lame.
Applying option ac (set number of audio channels) with argument 1.
Applying option ab (audio bitrate (please use -b:a)) with argument 64k.
Applying option f (force format) with argument mp3.
Successfully parsed a group of options.
Opening an output file: output.
[file # 000002e7d958e340] Setting default whitelist 'file,crypto,data'
Successfully opened the file.
Parsing a group of options: output url C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Audio.mp3.
Successfully parsed a group of options.
Opening an output file: C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Audio.mp3.
[file # 000002e7da1bc480] Setting default whitelist 'file,crypto,data'
Successfully opened the file.
Stream mapping:
Stream #0:1 -> #0:0 (aac (native) -> mp3 (libmp3lame))
Stream #0:1 -> #1:0 (aac (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
detected 8 logical cores
[graph_0_in_0_1 # 000002e7dac2b7c0] Setting 'time_base' to value '1/44100'
[graph_0_in_0_1 # 000002e7dac2b7c0] Setting 'sample_rate' to value '44100'
[graph_0_in_0_1 # 000002e7dac2b7c0] Setting 'sample_fmt' to value 'fltp'
[graph_0_in_0_1 # 000002e7dac2b7c0] Setting 'channel_layout' to value '0x3'
[graph_0_in_0_1 # 000002e7dac2b7c0] tb:1/44100 samplefmt:fltp samplerate:44100 chlayout:0x3
[format_out_0_0 # 000002e7dac2b8c0] Setting 'sample_fmts' to value 's32p|fltp|s16p'
[format_out_0_0 # 000002e7dac2b8c0] Setting 'sample_rates' to value '44100|48000|32000|22050|24000|16000|11025|12000|8000'
[format_out_0_0 # 000002e7dac2b8c0] Setting 'channel_layouts' to value '0x4'
[format_out_0_0 # 000002e7dac2b8c0] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph # 000002e7da1bdfc0] query_formats: 4 queried, 7 merged, 3 already done, 0 delayed
[auto_resampler_0 # 000002e7d958edc0] [SWR # 000002e7d95fd240] Using fltp internally between filters
[auto_resampler_0 # 000002e7d958edc0] [SWR # 000002e7d95fd240] Matrix coefficients:
[auto_resampler_0 # 000002e7d958edc0] [SWR # 000002e7d95fd240] FC: FL:0.707107 FR:0.707107
[auto_resampler_0 # 000002e7d958edc0] ch:2 chl:stereo fmt:fltp r:44100Hz -> ch:1 chl:mono fmt:fltp r:44100Hz
Output #0, mp3, to 'output':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: mp42mp41
TSSE : Lavf58.45.100
Stream #0:0(eng), 0, 1/44100: Audio: mp3 (libmp3lame), 44100 Hz, mono, fltp, 64 kb/s (default)
Metadata:
creation_time : 2020-11-24T06:56:49.000000Z
handler_name : #Mainconcept MP4 Sound Media Handler
encoder : Lavc58.91.100 libmp3lame
[graph_1_in_0_1 # 000002e7d9a5bf00] Setting 'time_base' to value '1/44100'
[graph_1_in_0_1 # 000002e7d9a5bf00] Setting 'sample_rate' to value '44100'
[graph_1_in_0_1 # 000002e7d9a5bf00] Setting 'sample_fmt' to value 'fltp'
[graph_1_in_0_1 # 000002e7d9a5bf00] Setting 'channel_layout' to value '0x3'
[graph_1_in_0_1 # 000002e7d9a5bf00] tb:1/44100 samplefmt:fltp samplerate:44100 chlayout:0x3
[format_out_1_0 # 000002e7da1b8580] Setting 'sample_fmts' to value 's32p|fltp|s16p'
[format_out_1_0 # 000002e7da1b8580] Setting 'sample_rates' to value '44100|48000|32000|22050|24000|16000|11025|12000|8000'
[format_out_1_0 # 000002e7da1b8580] Setting 'channel_layouts' to value '0x4|0x3'
[AVFilterGraph # 000002e7da1be5c0] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Output #1, mp3, to 'C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Audio.mp3':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: mp42mp41
TSSE : Lavf58.45.100
Stream #1:0(eng), 0, 1/44100: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp (default)
Metadata:
creation_time : 2020-11-24T06:56:49.000000Z
handler_name : #Mainconcept MP4 Sound Media Handler
encoder : Lavc58.91.100 libmp3lame
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
size= 129kB time=00:00:16.40 bitrate= 64.2kbits/s speed=32.8x
size= 254kB time=00:00:32.49 bitrate= 64.1kbits/s speed=32.5x
size= 256kB time=00:00:47.83 bitrate= 43.8kbits/s speed=31.8x
[out_0_0 # 000002e7d95b3d40] EOF on sink link out_0_0:default.
[out_1_0 # 000002e7d9bae780] EOF on sink link out_1_0:default.
No more output streams to write to, finishing.
[libmp3lame # 000002e7d958dbc0] Trying to remove 175 more samples than there are in the queue
[libmp3lame # 000002e7da1bb940] Trying to remove 175 more samples than there are in the queue
size= 442kB time=00:00:56.45 bitrate= 64.1kbits/s speed= 33x
video:0kB audio:1324kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Video1.mp4):
Input stream #0:0 (video): 10 packets read (17858 bytes);
Input stream #0:1 (audio): 2431 packets read (886147 bytes); 2431 frames decoded (2489344 samples);
Total: 2441 packets (904005 bytes) demuxed
Output file #0 (output):
Output stream #0:0 (audio): 2161 frames encoded (2489344 samples); 2162 packets muxed (451813 bytes);
Total: 2162 packets (451813 bytes) muxed
Output file #1 (C:/Users/ScottRobertson/Desktop/VirtualExhibition/Final Booth Data/1-Abidi (gold)/Video/Audio.mp3):
Output stream #1:0 (audio): 2161 frames encoded (2489344 samples); 2162 packets muxed (903627 bytes);
Total: 2162 packets (903627 bytes) muxed
2431 frames successfully decoded, 0 decoding errors
[AVIOContext # 000002e7d958e400] Statistics: 2 seeks, 3 writeouts
[AVIOContext # 000002e7da1bc580] Statistics: 2 seeks, 5 writeouts
[AVIOContext # 000002e7d9578040] Statistics: 2725576 bytes read, 65 seeks
How can I force the desired audio bitrate?
Also, why is there 2 outputs called Output file #0 and Output file #1 and why the #0 doesn't have any path?
Remove the word output in your command.
That is not an option name; ffmpeg treats it as the name of an output target since it does not immediately follow a token of the form -option.
Using ffmpeg, I get this output (2 different examples)
Input #0, au, from 'c:\TestFiles\sample.au':
Duration: 00:02:02.09, start: 0.000000, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16be ([3][0][0][0] / 0x0003), 44100 Hz, stereo, s16, 1411 kb/s
---
Input #0, wav, from 'c:\TestFiles\sample.wav':
Metadata:
encoder : Lavf57.83.100
Duration: 00:02:02.09, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
I understand what the pcm_s16be part means, but what is the information that follows that in parenthesis, e.g., ([3][0][0][0] / 0x0003)? I can't find any documentation that explains it
[3][0][0][0] / 0x0003 <-- these are codec tags.
The hex value on the right is the raw value as stored (usually little-endian); the format on the left is the literary representation. Usually if it's printable characters, that will be shown e.g. mp4a else the decimal number in square brackets.
Some people say that the parameter fake_buffer can remove the noise, but I tried it and there is still noise
modprobe snd-dummy fake_buffer=0
#modprobe snd-dummy fake_buffer=0
# lsmod |grep snd
snd_dummy 24576 0
snd_pcm_oss 61440 0
snd_mixer_oss 28672 1 snd_pcm_oss
snd_pcm 118784 2 snd_pcm_oss,snd_dummy
snd_timer 36864 1 snd_pcm
snd 94208 5 snd_timer,snd_pcm_oss,snd_pcm,snd_dummy,snd_mixer_oss
soundcore 16384 1 snd
# modinfo snd_dummy
filename: /lib/modules/4.14.129-bbrplus/kernel/sound/drivers/snd-dummy.ko
license: GPL
description: Dummy soundcard (/dev/null)
author: Jaroslav Kysela <perex#perex.cz>
depends: snd-pcm,snd
retpoline: Y
intree: Y
name: snd_dummy
vermagic: 4.14.129-bbrplus SMP mod_unload modversions
parm: index:Index value for dummy soundcard. (array of int)
parm: id:ID string for dummy soundcard. (array of charp)
parm: enable:Enable this dummy soundcard. (array of bool)
parm: model:Soundcard model. (array of charp)
parm: pcm_devs:PCM devices # (0-4) for dummy driver. (array of int)
parm: pcm_substreams:PCM substreams # (1-128) for dummy driver. (array of int)
parm: fake_buffer:Fake buffer allocations. (bool)
parm: hrtimer:Use hrtimer as the timer source. (bool)
#aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
dummy
default:CARD=Dummy
Dummy, Dummy PCM
Default Audio Device
sysdefault:CARD=Dummy
Dummy, Dummy PCM
Default Audio Device
dmix:CARD=Dummy,DEV=0
Dummy, Dummy PCM
Direct sample mixing device
dsnoop:CARD=Dummy,DEV=0
Dummy, Dummy PCM
Direct sample snooping device
hw:CARD=Dummy,DEV=0
Dummy, Dummy PCM
Direct hardware device without any conversions
plughw:CARD=Dummy,DEV=0
Dummy, Dummy PCM
Hardware device with all software conversions
# ffmpeg -f alsa -i hw:0 -t 5 /dev/shm/out.mp3 -y
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'hw:0':
Duration: N/A, start: 1616225206.353453, bitrate: 1536 kb/s
Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to '/dev/shm/out.mp3':
Metadata:
TSSE : Lavf57.83.100
Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
Metadata:
encoder : Lavc57.107.100 libmp3lame
size= 79kB time=00:00:05.01 bitrate= 129.0kbits/s speed= 1x
video:0kB audio:79kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.293899%
i use alsamixer to set all Volume to 0 ,and there is still noise
I searched in google for a long time and did not find other people have this problem , where am I doing it wrong
I've set up a radio for local network, everything (Icecast2, BUTT (Broadcast Using This Tool)) is ready and working except one thing. BUTT intends to capture audio from input device, but that is not what I want, I want to stream desktop audio. I've created a loopback device using modprobe snd-aloop. I actually managed to route the audio from a specific program to that loopback device and I was able to hear it on the web player but the sound is stuttering, speeding up and getting back to normal its hard to describe. It's like decoder cannot catch up. All these are happening in less than a second.
Output from pactl list short sinks
1 alsa_output.pci-0000_00_1f.3.analog-stereo module-alsa-card.c s16le 2ch 48000Hz RUNNING
3 alsa_output.platform-snd_aloop.0.analog-stereo module-alsa-card.c s16le 2ch 44100Hz IDLE
11 alsa_output.pci-0000_01_00.1.hdmi-stereo module-alsa-card.c s16le 2ch 44100Hz IDLE
I have created ~/.asoundrc as most of the tutorial encouraged me to do so, and I believe this problem can be solved by this file but I am not familiar enough with alsa and its features. Sample rate of sound card is 48000 Hz but BUTT is forcing me to select 44100.
Contents of ~/.asoundrc
pcm.!default {
type asym
playback.pcm "LoopAndReal"
#capture.pcm "looprec"
capture.pcm "hw:0,0"
}
pcm.looprec {
type hw
card "Loopback"
device 1
subdevice 0
}
pcm.LoopAndReal {
type plug
slave.pcm mdev
route_policy "duplicate"
}
pcm.mdev {
type multi
slaves.a.pcm pcm.MixReale
slaves.a.channels 2
slaves.b.pcm pcm.MixLoopback
slaves.b.channels 2
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave b
bindings.2.channel 0
bindings.3.slave b
bindings.3.channel 1
}
pcm.MixReale {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
#rate 48000
rate 44100
periods 128
period_time 0
period_size 1024 # must be power of 2
buffer_size 8192
}
}
pcm.MixLoopback {
type dmix
ipc_key 1025
slave {
pcm "hw:Loopback,0,0"
#rate 48000
rate 44100
periods 128
period_time 0
period_size 1024 # must be power of 2
buffer_size 8192
}
}
Thanks in advance for any help.
I've managed to pipe my desktop audio to mic input using purely pulse audio and no alsa, thanks to this thread: https://unix.stackexchange.com/questions/82259/how-to-pipe-audio-output-to-mic-input
I am trying to mux video (H.264) and audio (PCM_S16LE, no compression) into an MPEG transport stream using ffmpeg. The video shows fine. The audio stream, however, does not play. The audio stream, shown by ffprobe is AAC, which is obviously not my intention. So I must be doing something wrong in adding the audio stream. Any idea how I can correct this?
This is my code for adding an audio stream:
void add_audio_stream()
{
CodecID codec_id = CODEC_ID_PCM_S16LE;
AVStream *p_ast = av_new_stream(fc, 1);
if (!p_ast) {
fprintf(stderr, "Could not alloc audio stream\n");
exit(1);
}
ai = p_ast->index;
AVCodecContext *pcc = p_ast->codec;
avcodec_get_context_defaults2( pcc, AVMEDIA_TYPE_AUDIO );
pcc->codec_type = AVMEDIA_TYPE_AUDIO;
pcc->codec_id = codec_id;
pcc->sample_fmt = AV_SAMPLE_FMT_S16;
//pcc->bit_rate = 44100*16*2;
pcc->bit_rate = 0;
pcc->sample_rate = 44100;
pcc->channels = 2;
pcc->time_base = (AVRational){1, 44100};
// some formats want stream headers to be separate
if (fc->oformat->flags & AVFMT_GLOBALHEADER)
{
printf(" **** 1 ****\n");
pcc->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
else
printf(" **** 2 ****\n");
AVCodec *codec;
/* find the audio encoder */
codec = avcodec_find_encoder(pcc->codec_id);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
/* open it */
if (avcodec_open(pcc, codec) < 0)
{
fprintf(stderr, "could not open codec\n");
exit(1);
}
}
Here is the output of ffprobe:
ffprobe version N-32405-g6337de9, Copyright (c) 2007-2011 the FFmpeg developers
built on Sep 8 2011 11:20:12 with gcc 4.4.3
configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-libmp3lame
libavutil 51. 16. 0 / 51. 16. 0
libavcodec 53. 13. 0 / 53. 13. 0
libavformat 53. 12. 0 / 53. 12. 0
libavdevice 53. 3. 0 / 53. 3. 0
libavfilter 2. 39. 0 / 2. 39. 0
libswscale 2. 1. 0 / 2. 1. 0
libpostproc 51. 2. 0 / 51. 2. 0
[mpegts # 0xa96daa0] Continuity Check Failed
[mpegts # 0xa96daa0] Continuity Check Failed
[aac # 0xa974da0] channel element 0.1 is not allocated
[aac # 0xa974da0] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
.
.
lot of gobbly-gook about missing AAC parameters . . .
.
.
[aac # 0xa974da0] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac # 0xa974da0] Error decoding AAC frame header.
[mpegts # 0xa96daa0] max_analyze_duration 5000000 reached at 5429789
[mpegts # 0xa96daa0] Continuity Check Failed
[mpegts # 0xa96daa0] Continuity Check Failed
Input #0, mpegts, from 'test_audio_video.mts':
Duration: 00:00:40.35, start: 0.010000, bitrate: 1907 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0.0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 30 fps, 30 tbr, 90k tbn, 60 tbc
Stream #0.1[0x101]: Audio: aac ([6][0][0][0] / 0x0006), 96000 Hz, 4.0, s16, 9 kb/s
I think i doubt that MEPG2 TS will allow PCM Audio. It can take up MPI, MP2 or AAC. AAC is being taken up more as a default choice rather than identificaiton.
Also, unlike Video, Audio headers are not very descriminative. i.e. doesn't have start codes and stuff, so other than PES header, usually there is no way to find out what type of Audio it is.
Encode audio if possible.
Try Gspot application to cross check.
Looks like MPEG-TS supports AES3-formatted PCM audio as private data, as specified by SMPTE 302M.
There is currently a s302m encoder/decoder in ffmpeg that will allow you to easily accomplish your goal.
some time ago I played with it too. And what I found out is that:
- Blu-ray only support 48000 Sample Rate
- I always use BIG ENDIAN not LITTLE ENDIAN.
I think ffmpeg will use Blu-ray settings for mpeg2_ts.