I'm tryin to implement a client/server application based on FFmpeg. Unfortunately RTP_MPEGTS isn't documented in the official FFmpeg Documentation - Formats.
Anyway i found inspiration from this old thread.
Server Side
(1) Capture mic audio as input. (2)Encode it as pcm 8khz mono and (3) send it locally as RTP_MPEGTS format over rtp protocol.
ffmpeg -f avfoundation -i none:2 -ar 8000 -acodec pcm_u8 -ac 1 -f rtp_mpegts rtp://127.0.0.1:41954
This works, but on initiation it alerts "[mpegts # 0x7fda13024600] frame size not set"
Client Side (on the same machine)
(1) Receive rtp audio stream input (2) write it in a file or playback.
ffmpeg -i rtp://127.0.0.1:41954 -vcodec copy -y "output.wav"
I'm using -vcodec copy because i've already verified it in another rtp stream in which -acodec copy didn't work.
This stuck and while closing with Ctrl+C shortcut it prints:
Input #0, rtp, from 'rtp://127.0.0.1:41954':
Duration: N/A, start: 8.956122, bitrate: N/A
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0: Data: bin_data ([6][0][0][0] / 0x0006)
Output #0, wav, to 'output.wav':
Output file #0 does not contain any stream
I don't understand if the client didn't receive any stream, or it cannot write rtp packets into "output.wav" file. (Client or server problem?)
In the old thread is explained a workaround. On server could run 2 ffmpeg instance:
One produces "tmp.ts" file due to mpegts, and the other takes "tmp.ts" as input and streams it over rtp. Is it possibile?
Is there any better way to do implement this client/server with the lowest latency possible?
Thanks for any help provided.
I tested this with an .aac file and it worked:
Streaming:
(notice I use a multicast address.
But if you test the streaming and receiving on the same machine you might use your 127.0.0.1 as loopback address to the local host.)
ffmpeg -f lavfi -i testsrc \
-stream_loop -1 -re -i "music.aac" \
-map 0:v -map 1:a \
-ar 8000 -ac 1 \
-f rtp_mpegts "rtp://239.1.1.9:1234"
You need a video source for the rtp_mpegts muxer. I created one with lavfi.
I used -stream_loop to loop the .aac file forever for my test. You don't need this with a mic as input.
Capture stream:
ffmpeg -y -i "rtp://239.1.1.9:1234" -c:a pcm_u8 "captured_stream.wav"
I use the -c:a pcm_u8 while capturing on purpose, because using it in the Streaming did not work on the capturing side.
The output is a low quality 8bit, 8kHz mono audio file but that was what you've asked for.
Related
I am trying to stream video and audio from a Camera in a browser using Webrtc and Wowza Media Server (4.7.3 version).
The camera stream (h264/aac) is first of all transcoded by using FFMPEG (version N-89681-g2477bfe built with gcc 4.8.5, last available version on ffmpeg website) in VP8/OPUS and then pushed to the Wowza Server.
By using the small Wowza webpage I ask for the Wowza stream to be displayed in the browser (Chrome Version 66.0.3336.5 Build officiel canary 32 bits).
FFMPEG used command :
ffmpeg -rtsp_transport tcp -i rtsp://<camera_stream> -vcodec libvpx -vb 600000 -crf 10 -qmin 0 -qmax 50 -acodec libopus -ab 32000 -ar 48000 -ac 2 -f rtsp rtsp://<IP_Address_Wowza>:<port_no_ssl>/<application_name>/test
When I click on Play stream I have a very bad quality video and audio (jerky video and very bad audio).
If I use this FFMPEG command:
ffmpeg -rtsp_transport tcp -i rtsp://<camera_stream> -vcodec libvpx -vb 600000 -crf 10 -qmin 0 -qmax 50 -acodec copy -f rtsp rtsp://<IP_Address_Wowza>:<port_no_ssl>/<application_name>/test
I will have a good video (flowing, smooth) but no audio (the camera micro is ON).
If libopus is the problem (as this test first shows), I tried libvorbis but with Chrome console I have this error "Failed to set remote offer sdp: Session error code: ERROR_CONTENT". Weird, cause libvorbis is one of the available codecs for Webrtc.
Is someone experiencing the same issue ? Did someone experience the same issue ?
Thanks in advance.
You probably have no audio because opus must have sample rate of 48000
You should add the flag:
"-ar 48000"
to the output settings
I also experienced the "bad quality video and audio issues".
I finally solved the issue by adding:
"-quality realtime" to the output settings .
That work well for me, I hope this will help you.
I am using the following command to stream a video and it's audio to localhost:
ffmpeg -re -i out.mp4 -map 0:0 -vcodec libx264 -f h264 udp://127.0.0.1:1234 -map 0:1 -acodec libfaac -f mp4a udp://127.0.0.1:2020
FFmpeg is not recognising my audio codec and my audio format so I get the following error message:
Error
What audio format and codec do I need to use? The codec information of the video I wish to send is as follows:
Codecs used
When I convert the audio track to mp3 I can run the above command and stream the video and audio properly. However I dont want to convert all my video audio-tracks to mp3.
(I am confused by all the encoders, decoders, codec names in the ffmpeg documentation) Is there a way of finding the right encoder to use with the mp4a audio codec other than reading the whole list of codecs and options?
Thanks.
I am trying to create a rtp stream using ffmpeg. I am taking input from my logitech C920 which has built in h264 encoding support and also has a microphone. I wanted to send both video(h264 either with the built in encoder or ffmpeg's encoder) and audio(any encoding) through RTP and then play the stream using ffplay.
So far I am able to send only the video with the following command:
ffmpeg -i /dev/video0 -r 24 -video_size 320x240 -c:v libx264 -f rtp rtp://127.0.0.1:9999
and also the audio separately using the command:
ffmpeg -f alsa -i plughw:CARD=C920,DEV=0 -acodec libmp3lame -t 20 -f rtp rtp://127.0.0.1:9998
and play the sdp file with:
ffplay -i -protocol_whitelist file,udp,rtp test3.sdp
ffplay -i -protocol_whitelist file,udp,rtp test4.sdp
I'm on Ubuntu 14.04
How can I play the two streams with a single ffplay command as ffplay cannot take two inputs and I can't send two streams using a single RTP stream(or can I?).
Also, how can I use the built in h264 encoder of my webcam?
Thank you!
I'm trying to use ezstream to stream to an icecast server, my problem is while encoding the audio, I decode it from mp3 with madplay and I'm trying to encode it with ffmpeg so the output is aac, someone told me to use adts to be able to stream aac the problem is that the encoding doesn't stream the audio, it shows the timer on the console but it goes from 0:00:00 to 0:00:40 to 0:01:30, etc until the song ends instead of going second by second, this is my config:
<ezstream>
<url>http://localhost:8100/t</url>
<sourcepassword>password</sourcepassword>
<format>MP3</format>
<filename>/home/vybroo/server/audio/play.m3u</filename>
<reencode>
<enable>1</enable>
<encdec>
<format>MP3</format>
<match>.mp3</match>
<decode>madplay -b 16 -R 44100 -S -o raw:- #T#</decode>
<encode>ffmpeg -f s16le -ar 44.1k -ac 2 -i - -b:a 32k -ar 44.1k -f adts -</encode>
</encdec>
</reencode>
</ezstream>
is the enconding config wrong?, what should i change so it streams second by second correctly
I have an avi file that has different levels of audio. Is there a way to decrease and increase appropriately where needed the audio of my file using ffmpeg?
In ffmpeg you can use the volume filter to change the volume of a track. Make sure you download a recent version of the program.
Find out the gain to apply
First you need to analyze the audio stream for the maximum volume to see if normalizing would even pay off:
ffmpeg -i video.avi -af "volumedetect" -f null /dev/null
Replace /dev/null with NUL on Windows. This will output something like the following:
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] mean_volume: -16.0 dB
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] max_volume: -5.0 dB
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] histogram_0db: 87861
As you can see, our maximum volume is -5.0 dB, so we can apply 5 dB gain. If you get a value of 0 dB, then you don't need to normalize the audio.
Apply the volume filter:
Now we apply the volume filter to an audio file. Note that applying the filter means we will have to re-encode the audio stream. What codec you want for audio depends on the original format, of course. Here are some examples:
Plain audio file: Just encode the file with whatever encoder you need:
ffmpeg -i input.wav -af "volume=5dB" output.mp3
Your options are very broad, of course.
AVI format: Usually there's MP3 audio with video that comes in an AVI container:
ffmpeg -i video.avi -af "volume=5dB" -c:v copy -c:a libmp3lame -q:a 2 output.avi
Here we chose quality level 2. Values range from 0–9 and lower means better. Check the MP3 VBR guide for more info on setting the quality. You can also set a fixed bitrate with -b:a 192k, for example.
MP4 format: With an MP4 container, you will typically find AAC audio. We can use ffmpeg's build-in AAC encoder.
ffmpeg -i video.mp4 -af "volume=5dB" -c:v copy -c:a aac -strict experimental -b:a 192k output.mp4
Here you can also use other AAC encoders. Some of them support VBR, too. See this answer and the AAC encoding guide for some tips.
In the above examples, the video stream will be copied over using -c:v copy. If there are subtitles in your input file, or multiple video streams, use the option -map 0 before the output filename.
The author's info is: Jon Skarpeteig in SuperUser
You can use my ffmpeg-normalize script for that.
First, install a recent version of ffmpeg. Then, install via pip install ffmpeg_normalize, then run it on an AVI file:
ffmpeg-normalize input.avi -o output.mkv -c:a aac -b:a 192k
Here, we're choosing to re-encode the audio with AAC at 192 kBit/s, and copy the video stream over to the output. This will perform EBU R128 normalization, but simple peak/RMS normalization is also possible.