Transcribe MP3 audio file with Bing Speech API (speech to text) - speech-to-text

I have a long recording (hour+) in the format of MP3. The following is the info i managed to get from FFMPEG about the audio file:
[mp3 # 000001fe666da320] Skipping 0 bytes of junk at 58650.
[mjpeg # 000001fe666effe0] Changing bps to 8
[mp3 # 000001fe666da320] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from '1.mp3':
Duration: 00:57:18.52, start: 0.000000, bitrate: 192 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, mono, s16p, 192 kb/s
Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1300x1370, 90k tbr, 90k tbn, 90k tbc
I would like to use Bing Speech API (Microsoft Oxford - Cognitive Services - Speech API) to transcribe this file (speech to text).
I believe that this is achievable by using something like the code below.
Option 1:
before sending up any audio data, you must first send up an SpeechAudioFormat descriptor to describe the layout and format of your raw audio data via DataRecognitionClient's sendAudioFormat() method. Can you provide a code sample for this option?
Option 2: converting the file to the target's acceptable format. I have done that with FFMPEG and this is what i got:
Duration: 00:57:23.67, bitrate: 256 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, 1 channels, s16, 256 kb/s
As I understand from the documentation, this should be acceptable: The audio must be PCM, mono, 16-bit sample, with sample rate of 8000 Hz or 16000 Hz.
I tried to send the audio to the server but did not get any reply. Am I on the right tracks? What is the maximum buffer size?
Do u see other, maybe easier option to get my audio file transcribed?
private void SendAudioHelper(string wavFileName)
{
using (FileStream fileStream = new FileStream(wavFileName, FileMode.Open, FileAccess.Read))
{
int bytesRead = 0;
byte[] buffer = new byte[1024];
try
{
do
{
// Get more Audio data to send into byte buffer.
bytesRead = fileStream.Read(buffer, 0, buffer.Length);
// Send of audio data to service.
this.dataClient.SendAudio(buffer, bytesRead);
}
while (bytesRead > 0);
}
finally
{
// We are done sending audio. Final recognition results will arrive in OnResponseReceived event call.
this.dataClient.EndAudio();
}
}
}

There is a limit of 15 seconds when you use the REST implementation. SDK has a limit of 2minutes.
Bing Speech team

Related

Writing a raw opus buffer to a file in NodeJS

I'm working on Discord voice and I made voice receiver, but I can't seem to be able to write the audio buffer to a file to be able to play it. Here's my code:
stopRecording() {
this.recording = false;
let output = Buffer.concat(this.recordedBuffers);
fs.writeFileSync('./out.opus', output.toString());
this.recordedBuffers = [];
console.log('done');
};
The file gets created but it is unplayable. Any ideas?
It may be worth confirming that recordedBuffers contains Ogg Opus file data. The first 32 bytes should show file header data indicating this is an Ogg, Webm, or Mp4 container. Once the mime-type is confirmed, the file extension can be changed from .opus if needed:
/* .opus = 'OggS ... Opus'
* .webm = '... webm ...'
* .mp4 = '... mp4 ...'
* .wav = '... WAVE ...'
*/
new TextDecoder().decode(concatenatedBuffers.slice(0, 32))
FFmpeg could also inspect the downloaded file:
$ ffprobe audio.opus
Input #0, ogg, from 'audio.opus':
Duration: 00:06:00.11, start: 0.000000, bitrate: 99 kb/s
Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp

ffmpeg output in multiple files one after another [duplicate]

I need to split a large video file into multiple pieces quickly and without files with errors. The basic idea is, I have a 2GB video file which I want to change to multiple formats. I have read some encoders can not make use of more than 1 core when encoding to a different format. So I had the idea to split the large file (which is very quick) into 4/8 pieces depending on how many cores I have available on individual servers. re-encode each piece to a new format and use these to display video in sequence.
so
testfile.mp4
becomes
testfile_piece_1.mp4
testfile_piece_2.mp4
testfile_piece_3.mp4
testfile_piece_4.mp4
these can then be individually be converted.
The solution should not be format specific. However I have found issues with mp4 files.
I have tried the command below, which works really well and fast but creates files with errors.
ffmpeg -i testfile.mp4 -ss 00:00:00 -t 00:20:00 -c copy testfile_piece_1.mp4
When I play the testfile_piece_1.mp4 on VLC it works fine. An issue arises when converting the split file to a different height and width mp4 file. I would get an error similar to "moov atom not found"
I tried adding -movflags faststart with no luck
I then came across this library https://code.google.com/archive/p/moovrelocator/ which fixed the moov issue but I would then get an error with regards to aac "Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height"
The other way of splitting the files is useless but involved re-encoding the file. not too bad for smaller file size but the 2GB file would probably take days to complete.
Is there a way to split the largr file quickly without producing files with errors? I have been working on it for days with no luck.
Console output for comment - FFmpeg splitting large files
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/usr/share/nginx/html/uploads/testfile01.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf56.36.100
Duration: 00:05:02.08, start: 302.120000, bitrate: 3254 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709) , 1920x1080 [SAR 1:1 DAR 16:9], 3252 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (de fault)
Metadata:
handler_name : VideoHandler
[libx264 # 0x165ffc0] width not divisible by 2 (853x480)
Output #0, mp4, to '/usr/share/nginx/html/uploads/testfile01_480.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf56.36.100
Stream #0:0(und): Video: h264, none, q=2-31, 128 kb/s, SAR 2560:2559 DAR 0:0 , 25 fps (default)
Metadata:
handler_name : VideoHandler
encoder : Lavc56.41.100 libx264
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
console output 2
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/usr/share/nginx/html/uploads/testfile.mp4':
Metadata:
major_brand : dash
minor_version : 0
compatible_brands: iso6avc1mp41
creation_time : 2016-01-24 04:26:37
Duration: 01:15:58.08, start: 0.000000, bitrate: 3163 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 3161 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc (default)
Metadata:
creation_time : 2016-01-24 04:26:37
handler_name : VideoHandler
[segment # 0x1197060] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
Output #0, segment, to '/usr/share/nginx/html/uploads/testfile%02d.mp4':
Metadata:
major_brand : dash
minor_version : 0
compatible_brands: iso6avc1mp41
encoder : Lavf56.36.100
Stream #0:0(und): Video: h264 (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 3161 kb/s, 25 fps, 25 tbr, 12800 tbn, 25 tbc (default)
Metadata:
creation_time : 2016-01-24 04:26:37
handler_name : VideoHandler
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
[mp4 # 0x11512a0] Codec for stream 0 does not use global headers but container format requires global headers
Use the segment muxer to break the input into segments:
ffmpeg -i testfile.mp4 -c copy -f segment -segment_time 1200 testfile_piece_%02d.mp4
This will split the source at keyframes, so segments may not be exactly 1200 seconds long. And the timestamps aren't reset, so some players will fail to play the 2nd and latter segments. If playability is needed, insert -reset_timestamps 1.
After the parallel encoding, you can stitch the generated segments by first creating a text file seg.txt like this
file 'encoded_testfile_piece_00.mp4'
file 'encoded_testfile_piece_01.mp4'
file 'encoded_testfile_piece_02.mp4'
file 'encoded_testfile_piece_03.mp4'
And then running
ffmpeg -f concat -i seg.txt -c copy -fflags +genpts encoded_full.mp4

FFmpeg Live streaming Error: ffmpeg av_interleaved_write_frame broken pipe. Conversion Failed

I am trying to stream video live using ffmpeg. videos stream is coming from a camera over the network and there is bunch error that i got in initial part of doing this but here i am stacked with a error that is coming from ffmpeg where it says av_interleaved_write_frame() broken pipe,then at the end i get a message saying conversion failed ,when i want to capture the stream in any format it works like a charm but when it comes to streaming it gives the error.Why this is happening?
the command i am using to get the streaming is here:
ffmpeg - i rtsp://10.63.191.250/aerostat http://10.20.1.142:8080/feed.ffm
or
ffmpeg - i rtsp://10.63.191.250/aerostat -acodec copy -vcodec copy http://10.20.1.142:8080/feed.ffm
//with both the command i am getting the same error
The ffmpeg out put is Here:
[h264 # 0x3965b40] decode_slice_header error
[h264 # 0x3965b40] no frame!
[h264 # 0x3965b40] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 # 0x3965b40] decode_slice_header error
[h264 # 0x3965b40] no frame!
Input #0, rtsp, from 'rtsp://10.63.191.250/aerostat':
Metadata:
title : \Aerostat
Duration: N/A, start: 63184.792467, bitrate: N/A
Program 1
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 77 kb/s
Stream #0:1: Video: h264 (Baseline) ([27][0][0][0] / 0x001B), yuv420p(progressive), 656x368 [SAR 1:1 DAR 41:23], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
[libvpx # 0x397a360] v1.6.0
Output #0, ffm, to 'http://10.20.1.142:8080/feed.ffm':
Metadata:
title : \Aerostat
creation_time : now
encoder : Lavf57.56.100
Stream #0:0: Audio: vorbis (libvorbis), 22050 Hz, mono, fltp, 64 kb/s
Metadata:
encoder : Lavc57.64.101 libvorbis
Stream #0:1: Video: vp8 (libvpx), yuv420p, 720x576 [SAR 164:115 DAR 41:23], q=10-42, 400 kb/s, 29.97 fps, 1000k tbn, 25 tbc
Metadata:
encoder : Lavc57.64.101 libvpx
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 800000 vbv_delay: -1
Stream mapping:
Stream #0:0 -> #0:0 (aac (native) -> vorbis (libvorbis))
Stream #0:1 -> #0:1 (h264 (native) -> vp8 (libvpx))
Press [q] to stop, [?] for help
av_interleaved_write_frame(): Broken pipe
Last message repeated 6 times
Error writing trailer of http://10.20.1.142:8080/feed.ffm: Broken pipeframe= 6 fps=0.0 q=0.0 Lsize= 52kB time=00:00:00.43 bitrate= 972.0kbits/s dup=5 drop=0 speed=0.778x
video:43kB audio:3kB subtitle:0kB other streams:0kB global headers:4kB muxing overhead: 13.661202%
Conversion failed!
Here is My FFserver.conf file:
HTTPPort 8080 # Port to bind the server to
HTTPBindAddress 0.0.0.0
MaxHTTPConnections 2000
MaxClients 1000
MaxBandwidth 10000 # Maximum bandwidth per client
# set this high enough to exceed stream bitrate
CustomLog -
<Feed feed.ffm> # This is the input feed where FFmpeg will send
File ./feed.ffm # video stream.
FileMaxSize 1G # Maximum file size for buffering video
ACL allow 10.20.0.0 10.20.255.255 # Allowed IPs
ACL allow 127.0.0.1
ACL allow localhost
</Feed>
<Stream live.webm> # Output stream URL definition
Feed feed.ffm # Feed from which to receive video
Format webm
# Audio settings
AudioCodec vorbis
Audi0oBitRate 64 # Audio bitrate
# Video settings
VideoCodec libvpx
VideoSize 720x576 # Video resolution
VideoFrameRate 25 # Video FPS
AVOptionVideo flags +global_header # Parameters passed to encoder
# (same as ffmpeg command-line parameters)
AVOptionVideo cpu-used 0
AVOptionVideo qmin 10
AVOptionVideo qmax 42
AVOptionVideo quality good
AVOptionAudio flags +global_header
PreRoll 15
StartSendOnKey
VideoBitRate 400 # Video bitrate
</Stream>
<Stream status.html> # Server status URL
Format status
# Only allow local people to get the status
ACL allow 127.0.0.1
ACL allow 10.20.0.0 10.20.255.255
</Stream>
<Redirect index.html> # Just an URL redirect for index
# Redirect index.html to the appropriate site
URL http://www.ffmpeg.org/
</Redirect>

fluent ffmpeg size output option not working

Summary: I'm trying to limit output to 3mb, .outputOptions('-fs 3000000') isn't working for me, the file is coming back with a size of 119260428 or 119mb.
Here is the code to try for yourself, all you need is a test.mp3 large enough that the resulting testoutput.ogg is > 3mb:
var ffmpeg = require("fluent-ffmpeg");
var command = ffmpeg();
var convertToOGG = function(){
var fileName = 'test.mp3'
ffmpeg.ffprobe(fileName, function(err, metadata) {
command
.input(fileName)
.inputFormat("mp3")
.audioChannels(1)
.outputOptions('-fs', 3000000)
.output('testoutput.ogg')
.on("progress", function(progress) {
console.log("Processing: " + progress.timemark);
})
.on("error", function(err, stdout, stderr) {
console.log("Cannot process video: " + err.message);
})
.on("end", function(stdout, stderr) {
ffmpeg.ffprobe('testoutput.ogg', function(err,metadata){
if(metadata.format.size >= 3000000){
console.log("didn't work")
}
})
})
.run();
});
};
convertToOGG();
Per the fluent-ffmpeg documentation you should be able to use a ffmpeg command in an output option: outputOption()
This method allows passing any output-related option to ffmpeg. You can call it with a single argument to pass a single option, optionnaly
with a space-separated parameter:
/* Single option */
ffmpeg('/path/to/file.avi').outputOptions('-someOption');
and in FFMPEG's documentation:
-fs limit_size (output) Set the file size limit, expressed in bytes. No further chunk of bytes is written after the limit is exceeded. The
size of the output file is slightly more than the requested file size.
It's giving me no errors, just seemingly ignoring the file size limit of 99mb and outputting a 119.3mb file.
Edit - Looks like -fs 3000000 is working for mp3 to wav, but still wont do mp3 to ogg. This is the output from running the command in terminal:
✗ ffmpeg -i test.mp3 -fs 3000000 testoutput.ogg
ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
libavutil 55. 34.100 / 55. 34.100
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.100 / 57. 56.100
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
[mp3 # 0x7fc6a4000000] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'test.mp3':
Metadata:
lyrics-eng : xxx
title : xxx
artist : xxx
album_artist : xxx
album : xxx
genre : xxx
Duration: 03:27:28.74, start: 0.000000, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, mono, s16p, 128 kb/s
Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 540x360, 90k tbr, 90k tbn, 90k tbc
Metadata:
title : Array
comment : Cover (front)
[swscaler # 0x7fc6a4808800] deprecated pixel format used, make sure you did set range correctly
[ogg # 0x7fc6a3815800] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
Output #0, ogg, to 'testoutput.ogg':
Metadata:
lyrics-eng : xxx
title : xxx
artist : xxx
album_artist : xxx
album : xxx
genre : xxx
encoder : Lavf57.56.100
Stream #0:0: Video: theora (libtheora), yuv444p, 540x360, q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
Metadata:
title : Array
DESCRIPTION : Cover (front)
encoder : Lavc57.64.101 libtheora
lyrics-eng : xxx
artist : xxx
ALBUMARTIST : xxx
album : xxx
genre : xxx
Stream #0:1: Audio: vorbis (libvorbis), 44100 Hz, mono, fltp
Metadata:
encoder : Lavc57.64.101 libvorbis
lyrics-eng : xxx
title : xxx
artist : xxx
ALBUMARTIST : xxx
album : xxx
genre : xxx
Stream mapping:
Stream #0:1 -> #0:0 (mjpeg (native) -> theora (libtheora))
Stream #0:0 -> #0:1 (mp3 (native) -> vorbis (libvorbis))
Press [q] to stop, [?] for help
frame= 1 fps=0.0 q=-0.0 Lsize= 116465kB time=03:27:28.71 bitrate= 76.6kbits/s speed=61.2x
video:9kB audio:114907kB subtitle:0kB other streams:0kB global headers:6kB muxing overhead: 1.347787%
The source mp3 contains a cover image. Attempting to transcode the source file to ogg results in an unusually high framerate for the video stream.
There's a problem with the framerate estimation: 90k fps resembles the stream timebase value:
[ogg # 0x7fc6a3815800] Frame rate very high for a muxer not efficiently supporting it.
[...]
Stream #0:0: Video: theora (libtheora), yuv444p, 540x360, q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
This could be caused by a problem with the source file, a bug in ffmpeg or maybe both.
A quick solution is to ignore the cover image using either -map 0:a or -vn.

Audio is lost on transcoding from DTS to AAC

On a Synology drive, I'm trying to transcode audio in an .MKV file from DTS to AAC using the native ffmpeg encoder. But without any error message (at least I don't recognize any here), the audio stream is just missing in the output files. I tried it on several different files, all with the same result, leading to the conclusion that either I'm unable to see the error or something's wrong with FFMPEG. I used the standard FFMPEG package provided with OPKG.
This is the output I usually get:
CirkosDaten> ffmpeg -i encodertest.mkv -map 0 -c:v copy -c:s copy -c:a aac
-strict experimental output.mkv
ffmpeg version 2.0.2 Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 3 2015 06:25:48 with gcc 4.6.4 (Marvell GCC release
20150204-c4af733b 64K MAXPAGESIZE ALIGN CVE-2015-0235)
configuration: --prefix=/usr/syno --incdir='${prefix}/include/ffmpeg'
--arch=arm --target-os=linux --cross-prefix=/usr/local/arm-marvell-linux-
gnueabi/bin/arm-marvell-linux-gnueabi- --enable-cross-compile --enable-
optimizations --enable-pic --enable-gpl --enable-shared --disable-static
--enable-version3 --enable-nonfree --enable-libfaac --enable-encoders
--enable-pthreads --disable-bzlib --disable-protocol=rtp --disable-
muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-
ffserver --disable-ffplay --disable-devices --disable-bzlib --disable-
altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-
libmp3lame --disable-vaapi --disable-decoder=amrnb --disable-encoder=zmbv
--disable-encoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed
--disable-encoder=eac3 --disable-decoder=dca --disable-decoder=eac3
--disable-decoder=truehd --cc=/usr/local/arm-marvell-linux-gnueabi/bin/arm-
marvell-linux-gnueabi-ccache-gcc
libavutil 52. 38.100 / 52. 38.100
libavcodec 55. 18.102 / 55. 18.102
libavformat 55. 12.100 / 55. 12.100
libavdevice 55. 3.100 / 55. 3.100
libavfilter 3. 79.101 / 3. 79.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
Guessed Channel Layout for Input Stream #0.1 : 5.1
Guessed Channel Layout for Input Stream #0.2 : 5.1
Input #0, matroska,webm, from 'encodertest.mkv':
Metadata:
title : encodingtest
creation_time : 2012-09-23 09:38:19
Duration: 00:01:00.94, start: 0.000000, bitrate: 9624 kb/s
Chapter #0.0: start 0.000000, end 55.347000
Metadata:
title : 00:00:00.000
Chapter #0.1: start 55.347000, end 60.936000
Metadata:
title : 00:04:56.171
Stream #0:0: Video: h264 (High), yuv420p, 1920x816 [SAR 1:1 DAR 40:17], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
Metadata:
title : encodingtest
Stream #0:1(ger): Audio: dts, 48000 Hz, 5.1 (default)
Metadata:
title : DTS
Stream #0:2(eng): Audio: dts, 48000 Hz, 5.1
Metadata:
title : DTS
Stream #0:3(ger): Subtitle: subrip (default)
Metadata:
title : Forced Subs
Output #0, matroska, to 'output.mkv':
Metadata:
title : encodingtest
encoder : Lavf55.12.100
Chapter #0.0: start 0.000000, end 55.347000
Metadata:
title : 00:00:00.000
Chapter #0.1: start 55.347000, end 60.936000
Metadata:
title : 00:04:56.171
Stream #0:0: Video: h264 (H264 / 0x34363248), yuv420p, 1920x816 [SAR 1:1 DAR 40:17], q=2-31, 23.98 fps, 1k tbn, 1k tbc (default)
Metadata:
title : encodingtest
Stream #0:1(ger): Subtitle: subrip (default)
Metadata:
title : Forced Subs
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:3 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 1461 fps=528 q=-1.0 Lsize= 54636kB time=00:01:00.81 bitrate=7360.1kbits/s
video:54624kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.023249%encodingtest
When I additionally set the bitrate, the output says
Codec AVOption b (set bitrate (in bits/s)) specified for output file #0
(output.mkv) has not been used for any stream. The most likely reason is
either wrong type (e.g. a video option with no video streams) or that it is
a private option of some encoder which was not actually used for any stream.
This indicates that somehow the 5.1, 48kHz DTS stream isn't considered as an input. Why?
EDIT: Is it possible that DTS decoding is turned off somewhere here? I reckoned maybe that's why I get no errors. But I didn't see any obvious option for that in the output.
OK, I think I found the error:
--disable-decoder=dca
So this armv7 FFmpeg build i used and got via OPKG is compiled without DCA decoding.
Probably I'll need to recompile this on my Disk Station with all the codecs :-( - or does anybody know whether it's possible to just upgrade via IPKG/OPKG?
Does anyone else think building FFmpeg yourself is an ordeal? :-)
EDIT: I just tried to used the standard 32bit package from the FFmpeg project, and indeed it works on armv7 too. Problem solved.

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