ios - glitches when playing audio in pieces with AVAudioPLayer -

i'm trying play audio ip camera streaming api http://...<ip-camera..>/audio.cgi

currently obtain play stream audio adapting mjpeg client, taking audio blocks , put them buffer (nsmutablearray) of consecutive sounds packages. here code receives data:

-(void) connection:(nsurlconnection *)connection didreceivedata:(nsdata *)data {     nslog(@"data recv.");       nslog(@"data lenght:%i",[data length]);      if ([data length]>audio_stream_header_size){         [recvdata appenddata:data]; //audio stream data - 1024 bytes         nslog(@"data append lenght:%i",[recvdata length]);      }     else {         [_encabezado appenddata:data]; //audio stream header - 44 bytes         nslog(@"crea encabezado:%@",_encabezado);         nslog(@"largo encabezado:%i",[_encabezado length]);      }       if ([recvdata length] >= packet_size_buffer_sonido) //packet_size_buffer_sonido = 81920     {          nsmutabledata *sounddata = [[nsmutabledata alloc] initwithcapacity:([recvdata length])];          nsmutabledata *soundcut = [[nsmutabledata alloc] initwithdata:recvdata];          [sounddata appenddata:_encabezado]; //audio stream header         [sounddata appenddata:soundcut];  //audio stream data          [_arrsonidos addobject:sounddata];          if ([_arrsonidos count]>(buffer_size_for_play-1)) {             if (playerbuffer.isplaying) {                 [playerbuffer agregadatasound:sounddata]; //adddatasound in buffer             } else {                 playerbuffer = [[sounddataplayer alloc]initwithfilenamequeue:_arrsonidos];             }         }          [recvdata setlength:0];          [sounddata release];     }  } 

each time when function called (function show up), header of data field getting (44 bytes) , bits of data bytes (1024 bytes) once per call. keep head receiving data in nsdata , receiving data until defined packet size, keeping package in buffer respective header data. , iterated adding packages buffer.

and other hand, create "playerbuffer" (inspired examples on web) called above code , responsible controlling buffer , playing every packet of buffer. object based on avaudioplayer:

sounddataplayer.h

#import <foundation/foundation.h> #import <avfoundation/avfoundation.h>  @interface sounddataplayer : nsobject <avaudioplayerdelegate> {     avaudioplayer* myplayer;     nsmutablearray* datasound;     int index; }  @property (nonatomic, retain) avaudioplayer* myplayer; @property (nonatomic, retain) nsmutablearray* datasound;  - (id)initwithfilenamequeue:(nsarray*)datas; - (void)audioplayerdidfinishplaying:(avaudioplayer *)player successfully:(bool)flag; - (void)playx:(int)i; - (void)stop; - (bool)isplaying; - (void)agregadatasound:(nsdata *)data;  @end 

sounddataplayer.m

#import "sounddataplayer.h" #import "extaudiofileconvertutil.h"  @implementation sounddataplayer  @synthesize myplayer; @synthesize datasound;  - (id)initwithfilenamequeue:(nsarray*)datas {     if ((self = [super init])) {         self.datasound = [[nsmutablearray alloc]initwitharray:datas];         index = 0;         [self playx:index];     }     return self; }  - (void)audioplayerdidfinishplaying:(avaudioplayer *)player successfully:(bool)flag {     if (index < datasound.count) {         [self playx:index];     } else {         //reached end of queue     } }  - (void)playx:(int)i {      self.myplayer = [[avaudioplayer alloc] initwithdata:[datasound objectatindex:i]  error:nil];     myplayer.delegate = self;     [myplayer preparetoplay];     [myplayer play];      index++; }  - (void)stop {     if (self.myplayer.playing) [myplayer stop]; }  - (bool)isplaying{     return self.myplayer.playing; }  - (void)agregadatasound:(nsdata *)data {     [datasound addobject:data]; //add sound data buffer }  @end 

that work , play looped audio, on each loop when avaudioplayer play new packet (nsdata) of buffer, hear annoying glitches between each packet.

i tried systemsoundand recording sound packs buffer data file see if data containing glitches, apparently fact glitches in end of data.

i found article in forum seems explain happens: here apparently not explain in case because x-wav audio , pcm codec, , think , understand uncompressed. although not know how uncompress audio nsdata packets.

additional information of streaming audio ip camera:

datos del audio en el header:

"cache-control" = "no-cache"; "content-type" = "audio/x-wav"; date = "tue jan 5 01:17:04 2010"; pragma = "no-cache"; server = "goahead-webs";

datos adicionales del audio codec: pcm samplerate: 16 khz

i appreciate if give me light on how solve problem... try solve many many hours... driving me crazy!!

thanks in advance , sorry bad english.

the "glitch" you're hearing combination of time takes receive "did finish playing" event, process it, initialize avaudioplayer next data packet, , send data sound card.

it's analogus waiting 1 cd player finish playing before starting next — it's impossible react fast enough glitch-free playback.

you want use audio queue services, , you'll want way compensate audio clock skew between camera , phone.