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
Related
I need to convert from .mp3 to .gsm (preferably with ffmpeg).
I used it for several different formats but with this isn't as simple as it was with the others.
I don't know what parameters I'm missing.
I tried using ffmpeg with the following comand:
ffmpeg -i ".\example.mp3" ".\example.gsm"
But it shows me the following error:
Sample rate 8000Hz required for GSM, got 44100Hz Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height Conversion failed!
ffmpeg -ar 8000 -ac 1 -i ".\example.mp3" ".\example.gsm"
-ar sample rate
-ac audio channel
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.
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 have converted my .mkv file into an .mp4 by using the command:
sudo avconv -i input.mkv -codec copy output.mp4
I am trying to play the mp4 file in the browser, but the audio is not playing. The video player shows that it is on mute, but the button is disabled so you cannot turn it off of mute.
Other mp4s are working, but they were not converted from .mkv. Any help would be much appreciated.
In an MP4 container, browsers usually support only H.264 video and either AAC or MP3 audio. The output from your avconv command should show the format of your video and audio; look under "Input #0" for the lines that start with "Stream #". If you audio is not already AAC or MP3 you will want to convert it instead of just copying it to the MP4 container. You can copy the video and convert only the audio with a command like this:
avconv -i input.mkv -c:v copy -c:a libmp3lame -q:a 2 output.mp4
The -c:v copy will copy the video, and -c:a libmp3lame will convert the audio to MP3 using the libmp3lame encoder. -q:a 2 sets the audio quality; use a lower number for better quality (and a larger file). You could instead convert to AAC audio if your avconv was configured with non-free codecs enabled and a good quality AAC encoder.
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.