Search Results

Search found 5071 results on 203 pages for 'audio zoom'.

Page 24/203 | < Previous Page | 20 21 22 23 24 25 26 27 28 29 30 31  | Next Page >

  • Streaming Audio over UDP to Android

    - by Mr. Pig
    Is it possible to have Android (perhaps via MediaPlayer or a different existing class) accept media streams over UDP? I've successfully had MediaPlayer connect to an HTTP stream (as well as static files hosted on an HTTP server) but I'm wondering how one would go about accepting a stream from a UDP source. I've seen this and suppose a solution similar to that (where I download the stream via an independent UDP socket and then move the data to a MemoryBuffer that I then pass to MediaPlayer) is an option but I'm curious if a method already exists in the SDK, and if it does not, what other options do I have? Thanks

    Read the article

  • Audio recording and playback in Silverlight

    - by Ramesh
    I have a Silverlight 4 application that records user's voice through the mic. Now, as soon as the recording is completed, I need to play the recorded voice back to the user before posting it to the server. Is it at all possible to play it back to the user without getting into format conversions etc? Any ideas are welcome. Thanks!

    Read the article

  • Android - Getting audio to play through earpiece

    - by Donal Rafferty
    I currently have code that reads a recording in from the devices mic using the AudioRecord class and then playing it back out using the AudioTrack class. My problem is that when I play it out it plays vis the speaker phone. I want it to play out via the ear piece on the device. Here is my code: public class LoopProg extends Activity { boolean isRecording; //currently not used AudioManager am; int count = 0; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); am = (AudioManager) getSystemService(Context.AUDIO_SERVICE); am.setMicrophoneMute(true); while(count <= 1000000){ Record record = new Record(); record.run(); count ++; Log.d("COUNT", "Count is : " + count); } } public class Record extends Thread { static final int bufferSize = 200000; final short[] buffer = new short[bufferSize]; short[] readBuffer = new short[bufferSize]; public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); AudioRecord arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); AudioTrack atrack = new AudioTrack(AudioManager.STREAM_MUSIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize, AudioTrack.MODE_STREAM); am.setRouting(AudioManager.MODE_NORMAL,1, AudioManager.STREAM_MUSIC); int ok = am.getRouting(AudioManager.ROUTE_EARPIECE); Log.d("ROUTING", "getRouting = " + ok); setVolumeControlStream(AudioManager.STREAM_VOICE_CALL); //am.setSpeakerphoneOn(true); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); am.setSpeakerphoneOn(false); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); atrack.setPlaybackRate(11025); byte[] buffer = new byte[buffersize]; arec.startRecording(); atrack.play(); while(isRecording) { arec.read(buffer, 0, buffersize); atrack.write(buffer, 0, buffer.length); } arec.stop(); atrack.stop(); isRecording = false; } } } As you can see if the code I have tried using the AudioManager class and its methods including the deprecated setRouting method and nothing works, the setSpeatPoneOn method seems to have no effect at all, neither does the routing method. Has anyone got any ideas on how to get it to play via the earpiece instead of the spaker phone?

    Read the article

  • Core-audio - constructing an AudioBufferList struct (Q about c struct definition)

    - by mustISignUp
    The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

    Read the article

  • Finding out estimated duration of a stream using Core Audio

    - by Reflog
    I am streaming a MP3 over network using custom feeding code, not AVAudioPlayer (which only works with URLs) using APIs like AudioFileStreamOpen and etc. Is there any way to estimate a length of the stream? I know that I can get a 'elapsed' property using: if(AudioQueueGetCurrentTime(queue.audioQueue, NULL, &t, &b) < 0) return 0; return t.mSampleTime / dataFormat.mSampleRate; But what about total duration to create a progress bar? Is that possible?

    Read the article

  • Core-audio - constructing an AudioBufferList

    - by mustISignUp
    The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

    Read the article

  • processing an audio wav file with C

    - by sa125
    Hi - I'm working on processing the amplitude of a wav file and scaling it by some decimal factor. I'm trying to wrap my head around how to read and re-write the file in a memory-efficient way while also trying to tackle the nuances of the language (I'm new to C). The file can be in either an 8- or 16-bit format. The way I thought of doing this is by first reading the header data into some pre-defined struct, and then processing the actual data in a loop where I'll read a chunk of data into a buffer, do whatever is needed to it, and then write it to the output. #include <stdio.h> #include <stdlib.h> typedef struct header { char chunk_id[4]; int chunk_size; char format[4]; char subchunk1_id[4]; int subchunk1_size; short int audio_format; short int num_channels; int sample_rate; int byte_rate; short int block_align; short int bits_per_sample; short int extra_param_size; char subchunk2_id[4]; int subchunk2_size; } header; typedef struct header* header_p; void scale_wav_file(char * input, float factor, int is_8bit) { FILE * infile = fopen(input, "rb"); FILE * outfile = fopen("outfile.wav", "wb"); int BUFSIZE = 4000, i, MAX_8BIT_AMP = 255, MAX_16BIT_AMP = 32678; // used for processing 8-bit file unsigned char inbuff8[BUFSIZE], outbuff8[BUFSIZE]; // used for processing 16-bit file short int inbuff16[BUFSIZE], outbuff16[BUFSIZE]; // header_p points to a header struct that contains the file's metadata fields header_p meta = (header_p)malloc(sizeof(header)); if (infile) { // read and write header data fread(meta, 1, sizeof(header), infile); fwrite(meta, 1, sizeof(meta), outfile); while (!feof(infile)) { if (is_8bit) { fread(inbuff8, 1, BUFSIZE, infile); } else { fread(inbuff16, 1, BUFSIZE, infile); } // scale amplitude for 8/16 bits for (i=0; i < BUFSIZE; ++i) { if (is_8bit) { outbuff8[i] = factor * inbuff8[i]; if ((int)outbuff8[i] > MAX_8BIT_AMP) { outbuff8[i] = MAX_8BIT_AMP; } } else { outbuff16[i] = factor * inbuff16[i]; if ((int)outbuff16[i] > MAX_16BIT_AMP) { outbuff16[i] = MAX_16BIT_AMP; } else if ((int)outbuff16[i] < -MAX_16BIT_AMP) { outbuff16[i] = -MAX_16BIT_AMP; } } } // write to output file for 8/16 bit if (is_8bit) { fwrite(outbuff8, 1, BUFSIZE, outfile); } else { fwrite(outbuff16, 1, BUFSIZE, outfile); } } } // cleanup if (infile) { fclose(infile); } if (outfile) { fclose(outfile); } if (meta) { free(meta); } } int main (int argc, char const *argv[]) { char infile[] = "file.wav"; float factor = 0.5; scale_wav_file(infile, factor, 0); return 0; } I'm getting differing file sizes at the end (by 1k or so, for a 40Mb file), and I suspect this is due to the fact that I'm writing an entire buffer to the output, even though the file may have terminated before filling the entire buffer size. Also, the output file is messed up - won't play or open - so I'm probably doing the whole thing wrong. Any tips on where I'm messing up will be great. Thanks!

    Read the article

  • audio processing using java

    - by Sukhhhh
    We have a requirement where we need to convert from .wav file to .mp3 and we are currently using "Tritonus" library to do that . The concern with that library is that requires "installation" of some "dll" files to the class path. I am wondering are there any API's those allow better processing without local installation. And other question is ,having mp3 format files will make it easier to join the files into a single file than having .wav files ?

    Read the article

  • Play multiple audio files using AVAudioPlayer

    - by inScript09
    Hi all, I am planning on releasing 10 of my song recordings for free but bundled in an iphone app. They are not available on web or itunes or anywhere as of now. I am new to iphone sdk (latest) as you can imagine, so I have been going through the developer documentation, various forums and stackoverflow to learn. Apple's avTouch sample application was a great start. But I want my app to play all the 10 tracks one by one. All the songs are added to resources folder and are named as track1, track2...track10. In the avTouch app code I can see the following 2 parts which is where I think I need to make changes to achieve what I am looking for. But I am lost. // Load the array with the sample file NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: [[NSBundle mainBundle] pathForResource:@"sample" ofType:@"m4a"]]; - (void)audioPlayerDidFinishPlaying:(AVAudioPlayer *)player successfully:(BOOL)flag { if (flag == NO) NSLog(@"Playback finished unsuccessfully"); [player setCurrentTime:0.]; [self updateViewForPlayerState]; } can anyone please help me on 1. how to load the array with all the 10 tracks which are added to resources folder 2. and when I hit play, player should start the first track. when the 1st track ends 2nd track should start and so on for the remaining tracks. Thank You

    Read the article

  • blackberry implement audio player

    - by Prasad
    Hi, I am developing an application which let users to hear songs online. And I used Blackberry Player and Manager APIs. My application works fine and I can play songs. Now I wan't to add more controls to it. As an example I want pause, play songs. Mute the sound, Control the volume. Display the progress of the play back. Display the current time position of the song like that. I started research on that. And I tried to do that with PlayerListener. But unfortunately all the time I am getting IllegalStateException. So I can't go ahead with that research. As a help can someone please tell me how can I implement above kind of controls for a player. Appreciate if someone can post a sample code to do that. Further I will put my playback source code here. public void run() { try { p = Manager.createPlayer(requestedSong + SystemSettings.strNetwork); p.setLoopCount(1); p.start(); } catch (IOException ioe) { } catch (MediaException me) { } } public void run() { try { p = Manager.createPlayer(strSongURL); p.setLoopCount(1); p.start(); } catch (IOException ioe) { } catch (MediaException me) { } } Thank you very much. Prasad

    Read the article

  • Recognising tone of the audio

    - by terabytest
    Hi, I have a guitar and I need my pc to be able to tell what note is being played, recognizing the tone. Is it possible to do it in python, also is it possible with pygame? Being able of doing it in pygame would be very helpful.

    Read the article

  • Streaming audio (YouTube)

    - by wvd
    Hello all, I'm writing a CLI for a music-media-platform. One of the features is going to be that you can directly play YouTube videos from the CLI. I don't really have an idea to do it but this one sounded the most reasonable: I'm going to use of those sites where you can download music from YouTube, e.g. http://keepvid.com/ - then I directly stream & play this -- but I have one problem. Is there any Python library capable of doing this and if so, do you have any concrete examples? I've been looking but found nothing, even not with gstreamer. Thanks, William van Doorn

    Read the article

  • Qt Audio Recording Question

    - by Cenoc
    This is sort of a follow-up/branch off a previous question, which still stands unresolved. Are there other codecs besides pcm for qt QAudio class? I cant seem to find any... I want to have a way of playing stuff recorded by qt on vlc. Thanks in advance.

    Read the article

  • Resampling audio output for A2DP (from PCM WAV)

    - by user1669982
    The question is how to bring stereo PCM WAV 32,000 Hz with a stream of 1024 kbps (125 KB) to the headset with Bluetooth 2.1 on a CM7 smartphone with DSPManager. Is it possible? SBC is really bad idea. To TJD: Because it compresses the compressed stream. My Epic 4G don`t have Apt-X support. My headset Gemix BH-04A yellow. May be its possible with the Headset Profile (HSP)? I dont know about supported codecs in this profile.

    Read the article

  • Audio Playback Rate in Android

    - by Marquis
    So, I know that this has been done with a few Android apps before, but I cannot for the life of me figure out how, since it's not currently possible through the API. How does one adjust the playback rate of a sound played through MediaPlayer; either with or without adjusting the pitch is fine for now, though the latter is definitely preferred. If someone can point me in the direction of an open source app that I can use as guidance, that would also be fine. Thanks in advance.

    Read the article

  • Playback audio data with GWT

    - by Henrik
    I am creating a GWT client application which interacts with a server and I am getting all my response data from the server in JSON format. Amongst others there are wave data on the server's database which I would like to retrieve and then playback on the client. I am able to get the wave data as an array of bytes in the JSON format. My problem is, how do I playback the wave array data in a browser? Is it even possible or do I have to find another solution? I've searched the web and found some GWT packages which are able to playback sound, but they are all playing back directly from an url.

    Read the article

  • Real-time equalizer for all audio on computer

    - by greye
    Is it possible to capture all the sound from a computer and have it pass through a equalizer before reaching the speakers? How can you program a band pass filter on it? EDIT: I'm trying to get this on Windows (with Python? heh) but if there is a generic, cross-platform approach that would be great.

    Read the article

  • Inserting a WAV at a certain point in an audio file using python

    - by Onion
    My problem is the following: I have a 2-minute long WAV file, and my aim is to insert another WAV file (7 seconds long), at a certain point in the first WAV file (say, 0:48), essentially combining the two WAVs, using python. Unfortunately I haven't been able to figure out how to do that, and was wondering if there was some obvious solution that I was missing, or if it is even feasible to do with python. Is there perhaps a library available that might provide a solution? Thanks to all in advance.

    Read the article

  • Filter design for audio signal.

    - by beanyblue
    What I am trying to do is simple. I have a few .wav files. I want to remove noise and filter out specific frequencies. I don't have matlab and I intend to write my own code for all the filters. Right now, I have a way to read the .wav file and dump out the structure into a text file. My questions are the following: Can I directly apply the digital filters on this sampled data?{ ie, can I directly do a convolution between my input samples and h(n) for the filter function that i choose?). How do I choose the number of coefficients for the Window function? I have octave, so if someone can point me to anything that gives me some idea on how to process the .wav file using octave, that would be great too. I want to be able to filter out the frequency and then listen to the sound again. Is this possible with octave? I'm just a beginner with these kinds of things, so please bear with me if my questions are too naive. Any help will be great.

    Read the article

  • Multiple Audio Issues

    - by Lerp
    I am having issues with my audio on Ubuntu 12.04, I will try and give as much detail as possible so sorry if there's too much detail. The Problem Audio plays from both speakers and headphone regardless of what connector I choose and regardless of the profile I use. The microphone is constantly being played through headphones & speakers. The headphone audio is extremely quiet but plays from both ears when I select "Headphones" for the connector in Sound Settings. The headphone audio only plays from one ear and is quiet (but not as quiet as above) when I select "Analogue Output" for the connector in Sound Settings. I can only select "Headphones" as the connector in Sound Settings if I set the profile to either "Analogue Stereo Output/Duplex", all others only allow me to choose "Analogue Output" for the connector. Despite the headphone sound issues, the speaker sound is fine apart from the fact that I am not able to select which output is used, they just both play. My headphone and microphone are plugged into the front and my speakers are plugged into the back. What I have tried I have put everything in alsamixer to 100 apart from "Front Mic Boost" which I have set to 0. Command Output aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 2/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf7ff8000 irq 70 cat /proc/asound/modules 0 snd_hda_intel cat /proc/asound/card*/codec* | grep "Codec" Codec: Analog Devices AD1989B cat /etc/modprobe.d/alsa-base.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 # Keep snd-usb-audio from beeing loaded as first soundcard options snd-usb-audio index=-2 Hopefully I have provided enough information, I will happily provide anymore information needed. Thank you. Update Reinstalling alsa-base and pulseaudio fixed the headphone issues I was having.

    Read the article

  • How to compare mp3, flac audio data in a file, ignoring header data (ID3 tag) etc.?

    - by Rob
    I've backed up some audio files up in 2 places and added ID3 tags into one backup but not the other, since time has passed my own memory has faded on whether the backups are actually the same, but now one has ID3 data and the other doesn't, basic binary compare will fail and inspection will be cumbersome. Is there a tool to compare just the audio data (not the header, ID3) in mp3s, flac files, and other files using header data such as ID3. started a thread on beyond compare here: http://www.scootersoftware.com/vbulletin/showthread.php?t=7413 would consider other comparison software that does this task

    Read the article

< Previous Page | 20 21 22 23 24 25 26 27 28 29 30 31  | Next Page >