HI
I am new to Blackberry.
I am developing an application to get the song name from the live audio stream. I am able to get the mp3 stream bytes from the particular radioserver.To get the song name I add the flag "Icy-metadata:1".So I am getting the header from the stream.To get the mp3 block size I use "Icy-metaInt".How to recognize the metadatablocks with this mp3 block size.I am using the following code.can anyone help me to get it...Here the b[off+k] is the bytes that are from the server...I am converting whole stream in to charArray which is wrong, but how to recognize the metadataHeaders according to the mp3 block size..
b[off+k] = buffers[PlayBuf]PlayByte];
String metaSt = httpConn.getHeaderField("icy-metaint");
metaInt=Integer.parseInt(metaSt);
for (int i=0;i<b[off+k];i++)
{
metadataHeader+=(new String(b)).toCharArray();
System.out.println(metadataHeader);
metadataLength--;
Blackberry has no native regex functionality; I would recommend grabbing the regexp-me library (http://code.google.com/p/regexp-me/) and compiling it into your code. I've used it before and its regex support is pretty good. I think the regex in the code you posted would work just fine.
Related
I want to display a stream, and monitor the video stats of the displayed stream while recording it using libvlc. When I use sout + duplicate to record the stream while displaying it, I can only get the demux_bitrate stat from the displayed stream using libvlc_media_get_stats function. I am looking to get decoded_video, displayed_pictures, etc as well.
I've tried using the duplicate module to try to make this happen but I can't seem to make this work - am not sure if what I want to do is supported. Code below is tweaked from https://wiki.videolan.org/Documentation:Modules/display/ example for transcoding a stream while displaying the original version.
:sout=#duplicate{dst='transcode{vcodec=h264}:std{access=file,mux=ts,dst=c:\junk\test.mp4}',dst=display}
The stream displays, the file is generated, but the only valid stat is demux_bitrate which seems like the stat that would be accessible from the non-display stream instead of the displayed version.
Display and save with Transcoding
:sout=#duplicate{dst=display,dst="transcode{vcodec=h264}:standard{access=file,mux=mp4,dst=c:\junk\test.mp4}"}
Display and save without Transcoding
:sout=#duplicate{dst=display,dst=standard{access=file,mux=mp4,dst=c:\junk\test.mp4}}
I recently figured out that Google's Vision API can accept an external image URL and I was curious if anyone knew if Google's Speech could accept an external video URL such as a YouTube video?
The code I have in my mind would look something like this:
def transcribe_gcs(yotube_url):
"""Asynchronously transcribes the audio file specified by the gcs_uri."""
from google.cloud import speech
from google.cloud.speech import enums
from google.cloud.speech import types
client = speech.SpeechClient()
audio = types.RecognitionAudio(uri=youtube_url) # swapped out gcs_uri with youtube_url
config = types.RecognitionConfig(
encoding=enums.RecognitionConfig.AudioEncoding.FLAC,
# sample_rate_hertz=16000,
language_code='en-US')
operation = client.long_running_recognize(config, audio)
print('Waiting for operation to complete...')
response = operation.result(timeout=90)
# Each result is for a consecutive portion of the audio. Iterate through
# them to get the transcripts for the entire audio file.
for result in response.results:
# The first alternative is the most likely one for this portion.
print(u'Transcript: {}'.format(result.alternatives[0].transcript))
print('Confidence: {}'.format(result.alternatives[0].confidence))
I was curious if anyone knew if Google's Speech could accept an
external video URL such as a YouTube video?
It needs to be a local path to your audio file (less than 1 min audio file) or GCS URI for audio file longer than 1 minute. What you're thinking is not possible, the audio/video file needs to be in GCS.
I think you can achieve this by streaming same video (for example on wowza or on any server of your choice.) and then simply extract audio using lets say ffmpeg and pass this to google. It should work. use StreamingRecognizeRequest instead of RecognitionAudio.
I am using Google cloud speech through Python and finding many transcriptions are inaccurate and missing several words. This is a simple script I'm using to return a transcript of an audio file, in this case 'out307.wav':
client = speech.SpeechClient()
with io.open('out307.wav', 'rb') as audio_file:
content = audio_file.read()
audio = speech.types.RecognitionAudio(content=content)
config = speech.types.RecognitionConfig(
enable_word_time_offsets=True,
language_code='en-US',
audio_channel_count=1)
response = client.recognize(config, audio)
for result in response.results:
alternative = result.alternatives[0]
print(u'Transcript: {}'.format(alternative.transcript))
This returns the following transcript:
to do this the tensions and suspicions except
This is very far off what the actual audio says (I've uploaded it at https://vocaroo.com/i/s1zdZ0SOH1Ki). The audio is a .wav and very clear with no background noise. This is worse than average, as in some cases it will get the transcription fully correct on a 10 second audio file, or it may miss just a couple of words. Is there anything I can do to improve results?
This is weird, I tried your audio file with your code and I get the same result, but, if I change the language_code to "en-UK" I am able to get the full response.
I'm working for Google Cloud and I created for you a public issue here, you can track there the updates.
I'm developing a radio app for BB 5.0 in java. I don't find a way to play the radio from the url stream address that I have. I use multiple formats but nothing works (.pls, .aac, .m3u). I get a RuntimeException every time I try to play the stream. The content is ok, I've checked it.
InputStream stream = Connector.openInputStream(urlPlay);
StreamConnection streamConnection = (StreamConnection) Connector.open(urlPlay, Connector.READ);
InputStream readAhead = streamConnection.openDataInputStream();
byte[] audioData = new byte[500];
readAhead.read(audioData,0,audioData.length);
ByteArrayInputStream in2 = new ByteArrayInputStream(audioData);
player = javax.microedition.media.Manager.createPlayer(in2, "audio/aac");
System.out.println("REALIZE");
player.realize();
System.out.println("PREFETCH");
player.prefetch();
System.out.println("START");
player.start();
Edit:
When I use a URL from my .pls file I hear a little bit of my streaming but It stops immediately.
I suspect the problem is that you are trying to play playlist files instead of an actual stream. Generally, you need to parse those files yourself to get the real stream URLs.
If you open up that .m3u file, you will see that it is just a list of URLs. Take one of those URLs and then try it. Also, be sure you are setting the right content type. You can determine what that type is with cURL or VLC.
i have tried to record the audio in wav format and i succeeded but when i try to play that file, it gives an error that 'media file is of unsupported format'.
the main portion of code is as follows:
player = Manager.createPlayer("capture://audio?encoding=pcm&rate=44100&bits=16&channels=1");
player.realize();
controller = (RecordControl) player.getControl("RecordControl");
controller.setRecordLocation("file:///SDCard/BlackBerry/voicenotes/voice.wav");
//controller.setRecordSizeLimit(396900);
controller.startRecord();
player.start();
Thread.sleep(7000);
controller.commit();
player.close();
This code works well and it gave me a voice.wav file but i am not able to play this file.Is there is something which i forget?
Thanks in advance.
Sounds like the same problem as here:
http://supportforums.blackberry.com/t5/Java-Development/Bug-in-media-Manager-Player/td-p/1009027/page/2
Essentially, you need to add the RIFF/WAVE-header for the recorded data yourself, see the last post on the thread for example.