audio-streaming

问题 I am building an a/v html5 streaming web app. This question pretains to the audio portion of the project, but i'm sure i will run into a similar situation when i get started on the video portion. My target device is the iPad's safari browser (hence why i have to do this html5). Playback works fine, but i have a loading bar which needs to reflect how much of the track has been loaded. Following the w3 spec, i tried to implement this the following way using jQuery: var that = this; that.player

问题 I'm making something like streaming the audio from iPod library, send the data via network or bluetooth, and playback using audio queue. Thanks for this question and code. Help me a lot. I have two question about it. what should I send from one device to another? CMSampleBufferRef? AudioBuffer? mData? AudioQueueBuffer? packet? I have no idea. When the app finished the playing, it crashed, and I got error (-12733). I just want to know how to handle the errors instead of letting it crash.

问题 I am streaming audio by using AVPlayer. It is working well. But Now i need to make a slider to move audio forward and backward, like typical music player. I used the function seekToTime which works great with local audio file. But when i stream song from web url, the function stops audio when I forward slider with a large range. My guess is that the song is being downloaded and if i move slider forward with large range then may be system has not downloaded the song data packets for that time,

问题 I am trying to play a vehicle driven sound in a browser based game (continuously without break). My .wav file length is 1 second and has same frequency from beginning to end. But sound takes a little break before next iteration. Here is code: function playSound() { //alert(""); myAudio = new Audio('http://ithmbwp.com/feedback/SoundsTest/sounds/tank_driven.wav'); if (typeof myAudio.loop == 'boolean') { myAudio.loop = true; } else { myAudio.addEventListener('ended', function() { this

问题 I want to have a functionality of real time audio streaming on android device which is capturing audio through the MIC of the device and send it to the server. I know to send a send a audio file after recording but in case of real time I need help. May be it can be done by sending byte array continually to the server. If so how or if any other way, Please share your ideas. Thanks. EDIT- Android Client Code:- public class Main extends Activity { private MediaRecorder recorder; private final

问题 The goal of this question is trying to figure out how to play streaming audio using pyglet. The first is just making sure you're able to play mp3 files using pyglet, that's the purpose of this first snippet: import sys import inspect import requests import pyglet from pyglet.media import * pyglet.lib.load_library('avbin') pyglet.have_avbin = True def url_to_filename(url): return url.split('/')[-1] def download_file(url, filename=None): filename = filename or url_to_filename(url) with open

问题 The goal of this question is trying to figure out how to play streaming audio using pyglet. The first is just making sure you're able to play mp3 files using pyglet, that's the purpose of this first snippet: import sys import inspect import requests import pyglet from pyglet.media import * pyglet.lib.load_library('avbin') pyglet.have_avbin = True def url_to_filename(url): return url.split('/')[-1] def download_file(url, filename=None): filename = filename or url_to_filename(url) with open

问题 I have created a Server app that receives sound from client, i then broadcast this sound which is stored as bytes and send the bytes back to the clients that are connected to the server. now i am only using one client at the moment for testing and the client is receiving the voice back but the sound is stuttering all the time. Could some one please tell me what i am doing wrong? I think i understand some part of why the sound isn't playing smoothly but don't understand how to fix the problem.

问题 My simple audio streaming app works perfectly in 1.6 and 2.1 does not work in 2.2. I have no clue as to why. It works fine on the emulator, on actual 2.2 devices the audio stops unexpectedly after about 10 minutes or so. When I looked at logcat on my phone (HTC EVO) I see this: I/HTTPStream( 65): 1358 Bytes read, progress 34346/65536 I/HTTPStream( 65): 1358 Bytes read, progress 35704/65536 I/HTTPStream( 65): 1359 Bytes read, progress 37063/65536 I/HTTPStream( 65): 1358 Bytes read, progress

问题 I am trying to stream audio from the microphone to another iPhone via Apples Multipeer Connectivity framework. To do the audio capturing and playback I am using AVAudioEngine (much thanks to Rhythmic Fistman's answer here). I receive data from the microphone by installing a tap on the input, from this I am getting a AVAudioPCMBuffer which I then convert to an array of UInt8 which I then stream to the other phone. But when I am converting the array back to an AVAudioPCMBuffer I get an EXC_BAD