I cannot find simple code that ouput sound just from array through speaker on J2ME platform (I know that CreatePlayer class need WAV header with parameters).
int[] wav_hdr = {0x52,0x49,0x46,0x46,
0xC6,0x0F,0x00,0x00,
0x57,0x41,0x56,0x45,
0x66,0x6D,0x74,0x20,
0x12,0x00,0x00,0x00,
0x01,0x00,0x01,0x00,
0x40,0x1F,0x00,0x00,
0x40,0x1F,0x00,0x00,
0x01,0x00,0x08,0x00,
0x00,0x00,0x64,0x61,
0x74,0x61,0xA0,0x0F,
0x00,0x00};\\ wav header
byte[] adata = new byte[wav_hdr.length+4000];
for(int i=0; i<wav_hdr.length; i++) {
adata[i] = (byte)wav_hdr[i];
}
Player player = Manager.createPlayer(new ByteArrayInputStream(adata), "audio/x-wav");
player.start();
Related
I'm using pjsua to create a video call from a monitor to a phone. I'm able to establish an audio call without problem, but if I try to establish a video call (vid_cnt=1), I'm getting an error.
My purpose is to get and save the audio and video of the phone.
This is my configuration:
void hard_account_config(pjsua_acc_config& acc_cfg, pjsua_transport_id transport_tcp) {
pjsua_acc_config_default(&acc_cfg);
acc_cfg.ka_interval = 15;
// VIDEO
acc_cfg.vid_in_auto_show = PJ_TRUE;
acc_cfg.vid_out_auto_transmit = PJ_TRUE;
acc_cfg.vid_cap_dev = VideoCaptureDeviceId();
acc_cfg.vid_wnd_flags = PJMEDIA_VID_DEV_WND_BORDER | PJMEDIA_VID_DEV_WND_RESIZABLE;
acc_cfg.reg_timeout = 300;
acc_cfg.use_srtp = PJMEDIA_SRTP_DISABLED;
pjsua_srtp_opt_default(&acc_cfg.srtp_opt);
acc_cfg.ice_cfg_use = PJSUA_ICE_CONFIG_USE_CUSTOM;
acc_cfg.ice_cfg.enable_ice = PJ_FALSE;
acc_cfg.allow_via_rewrite = PJ_FALSE;
acc_cfg.allow_sdp_nat_rewrite = acc_cfg.allow_via_rewrite;
acc_cfg.allow_contact_rewrite = acc_cfg.allow_via_rewrite ? 2 : PJ_FALSE;
acc_cfg.publish_enabled = PJ_TRUE;
acc_cfg.transport_id = transport_tcp;
acc_cfg.cred_count = 1;
acc_cfg.cred_info[0].username = pj_string(USER);
acc_cfg.cred_info[0].realm = pj_string("*");
acc_cfg.cred_info[0].scheme = pj_string("Digest");
acc_cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
acc_cfg.cred_info[0].data = pj_string(PASS);
}
Once registration is completed, I run the following code:
prn("=== Test Call ===");
pj_str_t uri = pj_string("sip:" + call_target + "#" + SERVER);
pjsua_call_id call_id;
pjsua_call_setting call_setting;
pjsua_call_setting_default(&call_setting);
call_setting.flag = 0;
call_setting.vid_cnt = PJMEDIA_HAS_VIDEO ? 1 : 0;
pjsua_msg_data msg_data;
pjsua_msg_data_init(&msg_data);
pj_status_t status = pjsua_call_make_call(acc_id, &uri, &call_setting, NULL, &msg_data, &call_id);
if (status != PJ_SUCCESS) {
prn("Error trying: pjsua_call_make_call");
return;
}
I know that PJMEDIA_HAS_VIDEO is equal to 1 on the conf_site.h and pjsua_call_make_call return PJ_SUCCESS.
I've seen that if I have headphones connected, there is no problem. But if I disconnect them, the following error is shown:
#pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]
If I connect the headphones, I enable the video and run my code, the following error is shown:
#pjsua_media.c ......pjsua_vid_channel_update() failed for call_id 0 media 1: Unable to find default video device (PJMEDIA_EVID_NODEFDEV)
So, using PJSUA it is necessary to have audio and video devices on the monitor and phone? Should I create virtual ports if I don't have the devices?
You can use the following code to get a list of audio/video devices in PJSUA, which will most likely provide you with a loopback device (among others).
pjmedia_aud_dev_info audio_device[64];
unsigned int audio_device_cnt = 64;
status = pjsua_enum_aud_devs(audio_device, &audio_device_cnt);
printf("There are %d audio devices\n", audio_device_cnt);
for (int i = 0; i < audio_device_cnt; i++) {
printf("%d: %s\n", i, audio_device[i].name);
}
pjmedia_vid_dev_info video_device[64];
unsigned int video_device_cnt = 64;
status = pjsua_vid_enum_devs(video_device, &video_device_cnt);
printf("There are %d video devices\n", video_device_cnt);
for (int i = 0; i < video_device_cnt; i++) {
printf("%d: %s\n", i, video_device[i].name);
}
I have not personally tried capturing a loopback audio device but for video, PJSUA provides an internal colorbar generator (Colorbar generator in this list), which you can use.
Once you find the indices of loopback or dummy audio/video devices you want to use, you can set them by using
pjsua_set_snd_dev(<YOUR DUMMY CAPTURE DEVICE>, <YOUR DUMMY PLAYBACK DEVICE>);
acc_cfg.vid_cap_dev = <YOUR VIDEO CAPTURE DEVICE>;
I am learning JUCE and I am writing a program that just reads the input from the audio card and plays it back. Obviously this is just for learning purposes. I am using the audio application template. This is the code inside the getNextAudioBlock() function:
void getNextAudioBlock (const AudioSourceChannelInfo& bufferToFill) override
{
if(true) // this is going to be replaced by checking the value of a button
{
const int channel = 0;
if(true) // this is going to be replaced too
{
const float* inBuffer = bufferToFill.buffer->getReadPointer(channel, bufferToFill.startSample);
float* outBuffer = bufferToFill.buffer->getWritePointer(channel, bufferToFill.startSample);
for(int sample = 0; sample < bufferToFill.numSamples; ++sample)
outBuffer[sample] = inBuffer[sample];
}
else
{
bufferToFill.buffer->clear(0, bufferToFill.startSample, bufferToFill.numSamples);
}
}
else
{
bufferToFill.buffer->clear(0, bufferToFill.startSample, bufferToFill.numSamples);
}
}
The code is really simple: the content from the input buffer is copied directly to the output buffer. However, I am not hearing anything. What am I doing wrong?
Here, I am having a problem while converting ogg file to mp3 format. Reading ogg file is done successfully but while encoding it is throwing exception like,"Exception from HRESULT: 0xC00D3E85". Presently I am working on windows server 2012(64 bit).
public byte[] DecodeOGG(byte[] data,string trgtfilename,int bitrate)
{
byte[] dt = null;
NVorbis.NAudioSupport.VorbisWaveReader vr = null;
using(MemoryStream ms = new MemoryStream(data))
{
ms.Position = 0;
vr = new NVorbis.NAudioSupport.VorbisWaveReader(ms);
}
var samp = new SampleChannel(vr);
var ws = new SampleToWaveProvider16(samp);
MediaFoundationEncoder.EncodeToMp3(ws, trgtfilename, bitrate);
}
You need to call MediaFoundationInterop.Startup() somewhere in your application. NAudio may be updated in the future to call this automatically.
Here in this code,
I am using NAudio and Lame to convert wav file to mp3 stream and I am getting mp3 stream output without any issues. But apart from this I also want to set volume to the final mp3 stream. Any help is greatly appreciated.
public byte[] ConvertWavToMP3(byte[] bt, uint bitrate)
{
MemoryStream ms = new MemoryStream(bt);
ms.Seek(0, SeekOrigin.Begin);
var ws = new WaveFileReader(ms);
byte[] wavdata = null;
using (MemoryStream wavstrm = new MemoryStream())
using (WaveFileWriter wavwri = new WaveFileWriter(wavstrm, ws.WaveFormat))
{
ws.CopyTo(wavwri);
wavdata = wavstrm.ToArray();
}
WaveLib.WaveFormat fmt = new WaveLib.WaveFormat(ws.WaveFormat.SampleRate, ws.WaveFormat.BitsPerSample, ws.WaveFormat.Channels);
Yeti.Lame.BE_CONFIG beconf = new Yeti.Lame.BE_CONFIG(fmt, bitrate);
using (MemoryStream mp3strm = new MemoryStream())
using (Mp3Writer mp3wri = new Mp3Writer(mp3strm, fmt, beconf))
{
mp3wri.Write(wavdata, 0, wavdata.Length);
byte[] mp3data = mp3strm.ToArray();
return mp3data;
}
}
You can use AudioFileReader which will give you a Volume property you can adjust (1.0 is full scale). It will also turn the audio into an ISampleProvider with IEEE float samples, but I think LAME accepts IEEE float, otherwise use SampleToWaveProvider16 to get back down to 16 bit integer samples.
I'm porting an app from wp8 that requires playback of various sounds that can overlap. The only way I've found so far it to use MediaElement, but this doesn't allow overlapping sounds.
QUESTION - what is the easiest and best audio engine to use to play overlapping audio? Ideally I need a small example of how I can do this.
I've looked into WASAPI (http://code.msdn.microsoft.com/windowsapps/Windows-Audio-Session-22dcab6b), but it doesn't look like it supports simple playback ?
Maybe I can wrap the MediaFoundation and call it from winrt? (MediaEngine audio playback on WinRT)
Here is my code now, but when I play a new sound it cuts off the previously playing one rather than blending them.
ThreadUtility.runOnUiThread(
async delegate()
{
// TODO doesn't allow sounds to overlap!
Uri uri = new Uri(R.base_uri, R.raw.URI_PREFIX + resourceId);
StorageFile storageFile =
await Windows.Storage.StorageFile.GetFileFromApplicationUriAsync(
uri);
MediaElement element = new MediaElement();
var randomAccessStream = await storageFile.OpenReadAsync();
element.SetSource(randomAccessStream, storageFile.ContentType);
element.Volume = volume;
element.PlaybackRate = pitch;
//TODO element.Pan = pan;
element.Play();
}
);
SOLUTION (as per Filip's answer):
in the page class:
var mediaElements = new LinkedList<MediaElement>();
{
for (int channel = 0; channel < TeacherSoundGroover.NUM_CHANNELS; channel++)
{
var mediaElement = new MediaElement();
mediaElements.add(mediaElement);
// Must be in the tree otherwise it won't overlap!
m_titlePanel.Children.Add(mediaElement);
}
}
m_soundPlayer = new MySoundPlayer(mediaElements);
}
in the MySoundPlayer class:
ThreadUtility.runOnUiThread(
async delegate()
{
Uri uri = new Uri(R.base_uri, R.raw.URI_PREFIX + resourceId);
StorageFile storageFile =
await Windows.Storage.StorageFile.GetFileFromApplicationUriAsync(
uri);
if(m_mediaElements != null)
{
int count = m_mediaElements.size();
if (count > 0)
{
int channel = m_nextMediaElementToUse % count;
m_nextMediaElementToUse++;
MediaElement element = m_mediaElements.get(channel);
var randomAccessStream = await storageFile.OpenReadAsync();
element.Stop();
element.DefaultPlaybackRate = rate;
element.SetSource(randomAccessStream, storageFile.ContentType);
element.Volume = volume;
element.Balance = pan;
element.Play();
}
}
}
);
The easiest thing to do is use multiple MediaElement controls, though that might not give you desired results. The best way is to use XAudio2 either directly or through SharpDX if you want to avoid creating a C++/CX WinRT component.