I'm trying to play video in mp4 format but not working.
In console I execute this line and it works:
gst-launch playbin uri=rtmp://localhost:1935/files/video.mp4
But if I change to version 1.0 only works the audio:
gst-launch-1.0 playbin uri=rtmp://localhost:1935/files/video.mp4
in python I have the following code:
self.player = Gst.Pipeline.new("player")
source = Gst.ElementFactory.make("filesrc", "file-source")
demuxer = Gst.ElementFactory.make("mp4mux", "demuxer")
demuxer.connect("pad-added", self.demuxer_callback)
self.video_decoder = Gst.ElementFactory.make("x264enc", "video-decoder")
self.audio_decoder = Gst.ElementFactory.make("vorbisdec", "audio-decoder")
audioconv = Gst.ElementFactory.make("audioconvert", "converter")
audiosink = Gst.ElementFactory.make("autoaudiosink", "audio-output")
videosink = Gst.ElementFactory.make("autovideosink", "video-output")
self.queuea = Gst.ElementFactory.make("queue", "queuea")
self.queuev = Gst.ElementFactory.make("queue", "queuev")
colorspace = Gst.ElementFactory.make("videoconvert", "colorspace")
self.player.add(source)
self.player.add(demuxer)
self.player.add(self.video_decoder)
self.player.add(self.audio_decoder)
self.player.add(audioconv)
self.player.add(audiosink)
self.player.add(videosink)
self.player.add(self.queuea)
self.player.add(self.queuev)
self.player.add(colorspace)
source.link(demuxer)
self.queuev.link(self.video_decoder)
self.video_decoder.link(colorspace)
colorspace.link(videosink)
self.queuea.link(self.audio_decoder)
self.audio_decoder.link(audioconv)
audioconv.link(audiosink)
but I get this error:
Error: Error in the internal data flow. gstbasesrc.c(2865): gst_base_src_loop (): /GstPipeline:player/GstFileSrc:file-source:
streaming task paused, reason not-linked (-1)
What can be happening? think I am no good decoding
You are missing linking the demuxer pads to your queues. Demuxers have 'sometimes' pads so you need to listen to the pad-added signal of them and link in this callback. Remember to check the pad caps once you get them and link to the appropriate branch of your pipeline.
You can read about dynamic pads here: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-pads.html#section-pads-dynamic
You have in your code:
demuxer = Gst.ElementFactory.make("mp4mux", "demuxer")
demuxer.connect("pad-added", self.demuxer_callback)
I hope this is a cut/paste error, as demuxing with a mux will not work. I believe for an .mp4 file, the normal demuxer (if you are choosing one by hand) is qtdemux.
You could also use decodebin to decode the file for you.
Related
In my project, I need to get sound from a radio stream URL and play it. But it needs to be played in a specific output device called "VB-Cable Input (Virtual Audio Cable)".
I couldn't find a way to do it. This is my current code:
import vlc
import time
url = "https://shout25.crossradio.com.br:18156/1"
# Create VLC instance, media player and media
instance = vlc.Instance()
player = instance.media_player_new()
media = instance.media_new(url)
player.set_media(media)
# Get list of output devices
def get_device():
mods = player.audio_output_device_enum()
if mods:
mod = mods
while mod:
mod = mod.contents
# If VB-Cable is found, return it's module and device id
if 'CABLE Input (VB-Audio Virtual Cable)' in str(mod.description):
device = mod.device
module = mod.description
return device,module
mod = mod.next
# Sets the module and device id to VB-Cable
device,module = get_device()
# Sets VB-Cable as output device
player.audio_output_device_set(device,module)
# Trying the function #Random Davis stated, but results still the same
vlc.libvlc_audio_output_device_set(player, module, device)
# It should return VB-Cable as current audio output, but None is returned...
print(player.audio_output_device_get())
# Audio plays in default output device, which is incorrect
player.play()
time.sleep(60)
I searched through StackOverflow and the VLC documentation, but couldn't get the result I expected, I believe I'm using the functions in the wrong way.
After some research, I found this implementation of audio_output_device_set() in GitHub
As #RandomDavis said, the problem was indeed in this function. All I had to do was switching the first argument to None and leave the rest as it is.
This is the correct code:
# Sets VB-Cable as output device
player.audio_output_device_set(None, device)
Still not quite sure why inserting any values other than None makes the function invalid. But the problem is now solved.
I don't know how to do it and whenever i try solutions from other topics about this question i get errors.
Mostly "TypeError: show_progress_bar() missing 1 required positional argument: 'bytes_remaining'".
from pytube import YouTube
#took this def from another topic
def progress_function(stream, chunk, file_handle, bytes_remaining):
percent = round((1-bytes_remaining/video.filesize)*100)
if( percent%10 == 0):
print(percent, 'done...')
url = "Any youtube url"
yt = YouTube(url, on_progress_callback=progress_function)
yt.streams[0].download()
for example when i run this exact code it gives me that error.
I really can't comprehend its logic. I also searched the docs from pytube3 website but i can't solve this. Pls help me. Thanks.
Remove the stream then it will work, recently I tried to develop similar logic facing a similar error.
Here is the code that worked for me:
def progress(chunk, file_handle, bytes_remaining):
global filesize
remaining = (100 * bytes_remaining) / filesize
step = 100 - int(remaining)
print("Completed:", step) # show the percentage of completed download
The filesize can be retrived, once you select which video or audio to download such as
yt = YouTube(str(link), on_progress_callback=progress) # Declare YouTube
yt1 = yt.streams.get_by_itag(int(itag)) # itag is given when you list all the streams of a youtube video
filesize = yt1.filesize
Hope this helps!
I want to stream the audio of my microphone (that is being recorded via pyaudio) via Flask to any client that connects.
This is where the audio comes from:
def getSound(self):
# Current chunk of audio data
data = self.stream.read(self.CHUNK)
self.frames.append(data)
wave = self.save(list(self.frames))
return data
Here's my flask-code:
#app.route('/audiofeed')
def audiofeed():
def gen(microphone):
while True:
sound = microphone.getSound()
#with open('tmp.wav', 'rb') as myfile:
# yield myfile.read()
yield sound
return Response(stream_with_context(gen(Microphone())))
And this is the client:
<audio controls>
<source src="{{ url_for('audiofeed') }}" type="audio/x-wav;codec=pcm">
Your browser does not support the audio element.
</audio>
It does work sometimes, but most of the times I'm getting "[Errno 32] Broken pipe"
When uncommenting that with open("tmp.wav")-part (the self.save() optionally takes all previous frames and saves them in tmp.wav), I kind of get a stream, but all that comes out of the speakers is a "clicking"-noise.
I'm open for any suggestions. How do I get the input of my microphone live-streamed (no pre-recording!) to a webbrowser?
Thanks!
Try This its worked for me. shell cmd "cat" is working perfect see the code
iam using FLASK
import subprocess
import os
import inspect
from flask import Flask
from flask import Response
#app.route('/playaudio')
def playaudio():
sendFileName=""
def generate():
# get_list_all_files_name this function gives all internal files inside the folder
filesAudios=get_list_all_files_name(currentDir+"/streamingAudios/1")
# audioPath is audio file path in system
for audioPath in filesAudios:
data=subprocess.check_output(['cat',audioPath])
yield data
return Response(generate(), mimetype='audio/mp3')
This question was asked long time ago, but since I spent entire day to figure out how to implement the same, I want to give the answer. Maybe it will be helpful for somebody.
"[Errno 32] Broken pipe" error comes from the fact that client can not play audio and closes this stream.
Audio can not be played due to absence of the header in the data stream. You can easily create the header using genHeader(sampleRate, bitsPerSample, channels, samples) function from the code here . This header has to be attached at least to the first chunck of sent data ( chunck=header+data ). Pay attention, that audio can be played ONLY untill client reaches file size in download that you have to specify in the header. So, workaround would be to set in the header some big files size, e.g. 2Gb.
Instead of datasize = len(samples) * channels * bitsPerSample in the header function write datasize = 2000*10**6.
def gen_audio():
CHUNK = 512
sampleRate = 44100
bitsPerSample = 16
channels = 2
wav_header = genHeader(sampleRate, bitsPerSample, channels)
audio = AudioRead()
data = audio.get_audio_chunck()
chunck = wav_header + data
while True:
yield (chunck)
data = audio.get_audio_chunck()
chunck = data
After lots research and tinkering I finally found the solution.
Basically it came down to serving pyaudio.paFloat32 audio data through WebSockets using Flask's SocketIO implementation and receiving/playing the data in JavaScript using HTML5's AudioContext.
As this is requires quite some code, I think it would not be a good idea to post it all here. Instead, feel free to check out the project I'm using it in: simpleCam
The relevant code is in:
- noise_detector.py (recording)
- server.py (WebSocket transfer)
- static/js/player.js (receiving/playing)
Thanks everyone for the support!
VLC 2.2.3, python-vlc 1.1.2.
I have a virtual audio output and am trying to get libVLC to output on it. So far, I can see the virtual output appearing in libVLC, but selecting it makes audio play on the default output (i.e. the speakers).
This is the relevant part of what I have:
self.Instance = vlc.Instance()
self.player = self.Instance.media_player_new()
devices = []
mods = self.player.audio_output_device_enum()
if mods:
mod = mods
while mod:
mod = mod.contents
devices.append(mod.device)
mod = mod.next
vlc.libvlc_audio_output_device_list_release(mods)
# this is the part I change on each run of the code.
self.player.audio_output_device_set(None, devices[0])
I've run the code multiple times, changing the device ID as per the code comment. However, the output device doesn't actually change. This is a headache for two reasons:
1) audio_output_device_set() doesn't return anything. I can't tell if I'm actually accomplishing anything when I run this function.
2) I can't even run audio_output_device_get() to check if the set function is doing anything as this is only for libvlc 3. I would prefer for my program to work with 2.2.3.
So, what I did next was install VLC 3.0 and run the above code with it. Now, audio_output_device_get() works and I can see that the set function is actually changing the output device to the virtual output. But sound STILL plays on the speakers.
What's going on? How do I fix this?
I asked at the VLC forums and got a singularly unhelpful reply telling me to 'check logs and documentation'. That's it. I've been superglued to the rather lacking documentation to get this far. Even though I doubt it can help, I've decided to try logging. I thought it would be as simple as calling libvlc_log_set_file but it needs a libVLC file pointer, and I don't know how to create one with a name and mode as in Python.
tl;dr:
1) How do I successfully change the audio output device?
2) How do I set up maximum verbosity logging?
1) For some reason, I had to pause and unpause before VLC would register my change.
This code fixes things:
[... rest of GUI class ...]
self.p.play()
self.root.after(350, self.DeviceSet)
def DeviceSet(self):
self.p.audio_output_device_set(None, self.audiodevice)
self.p.pause()
self.root.after(10)
self.p.pause()
2) Initialise VLC as follows:
self.instance = vlc.Instance('--verbose 9')
Here is a full example of how to switch to different audio device.
Remember: don't call player.stop() after player.audio_output_device_set(), otherwise the set operation won't work!!
import time
from typing import List
import vlc
def vlc_set_device_test(filename: str):
# creating a vlc instance
vlc_instance: vlc.Instance = vlc.Instance()
player: vlc.MediaPlayer = vlc_instance.media_player_new()
media: vlc.Media = vlc_instance.media_new(filename)
player.set_media(media)
# list devices
device_ids: List[bytes] = []
mods = player.audio_output_device_enum()
if mods:
index = 0
mod = mods
while mod:
mod = mod.contents
desc = mod.description.decode('utf-8', 'ignore')
print(f'index = {index}, desc = {desc}')
device_ids.append(mod.device)
mod = mod.next
index += 1
# free devices
vlc.libvlc_audio_output_device_list_release(mods)
# hard code device
pc_speaker = device_ids[1]
headset = device_ids[3]
# play music to default device
player.play()
time.sleep(3)
# set output device
player.audio_output_device_set(None, headset)
# don't call player.stop()!!
player.pause()
# now music is playing from headset
player.play()
time.sleep(10)
player.stop()
if __name__ == '__main__':
vlc_set_device_test(r'D:\cheer up.mp3')
I have a simple pipeline set up as below with Gstreamer 1.0. When I try to create pull samples from the appsink, the code stalls at "sample = appsink.emit('pull-sample')" . The weird part is that if I remove that line, the code works as expected, continually printing "trying to pull sample". I get the same stall if I try to skip the first 100 samples or so as well as changing the properties on the appsink. Anyone have an idea of what's going on?
gst-launch-1.0 v4l2src device="/dev/video0" ! videorate ! video/x raw,height=480,width=640,framerate=15/1 ! appsink
def createASink():
asink = Gst.ElementFactory.make('appsink', 'asink')
asink.set_property('sync', False)
asink.set_property('emit-signals', True)
asink.set_property('drop', True)
asink.connect('new-sample', new_sample)
return asink
def new_sample(appsink):
print "Trying to pull sample"
sample = appsink.emit('pull-sample')
return False
So this is a hack workaround but it still works. If someone knows a better solution please let me know.
I can use a filesink to output the byte array and then read from that continuously. I use the multiprocessing module in python to run gstreamer and my consumer at the same time.
def consumeStream():
fp = "gst_data"
with open(fp, "r+b") as data_file:
data_file.truncate()
while True:
where = data_file.tell()
line = data_file.readline()
if not line:
time.sleep(.05)
data_file.seek(where)
else:
data_file.truncate()
print "Got data"