I am trying to write a listener using the CoreAudio API for when the default audio output is changed (e.g.: a headphone jack is plugged in). I found sample code, although a bit old and using deprecated functions (http://developer.apple.com/mac/library/samplecode/AudioDeviceNotify/Introduction/Intro.html, but it didn't work. Re-wrote the code in the 'correct' way using AudioHardwareAddPropertyListener method, but still it doesn't seem to work. When I plug in a headphone the function that I registered is not triggered. I'm a bit of a loss here... I suspect the problem may lay some where else, but I can't figure out where...
The Listener Registration Code:
OSStatus err = noErr;
AudioObjectPropertyAddress audioDevicesAddress = { kAudioHardwarePropertyDefaultOutputDevice, KAudioObjectPropertyScopeGlobal, KAudioObjectPropertyElementMaster };
err = AudioObjectAddPropertyListener ( KAudioObjectAudioSystemObject, &AudioDevicesAddress, coreaudio_property_listener, NULL);
if (err) trace ("error on AudioObjectAddPropertyListener");
After a search in sourceforge for projects that used the CoreAudio API, I found the rtaudio project, and more importantly these lines:
// This is a largely undocumented but absolutely necessary
// requirement starting with OS-X 10.6. If not called, queries and
// updates to various audio device properties are not handled
// correctly.
CFRunLoopRef theRunLoop = NULL;
AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
if ( result != noErr ) {
errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
error( RtError::WARNING );
}
After adding this code I didn't even need to register a listener myself.
Try CFRunLoopRun() - it has the same effect. i.e. making sure the event loop that is calling your listener is running.
Related
I'm making a program that will output text to speech, the user will need to be able to change the output device (for example to virtual audio cable). At the moment I'm using https://www.npmjs.com/package/say to produce the speech
e.g.
const say = require("say");
say.speak("Hello World");
but I've no clue on how I could go about choosing the audio output.
What I've found so far has been pretty useless, I'm guessing largely because I don't know the proper terminology to search for answers, regardless this is it:
I first found out about navigator.MediaDevices, then I found how I could make an audio element and change the audio device of that element via setSinkId, then I realized these things are probably(?) irrelevant since the say module seems to play sounds using powershell commands. I've even gone as far as trying to change powershell's output device in app volume device preferences(img), but that seems to do nothing.
I'm pretty much stumped right now, so I'd appreciate any help please. I'm not set on using Say as a module, it just seemed easy to use at first.
Edit:
A workaround I've settled with is making my own TTS class and using SpVoice.
I have something similar to this:
const childProcess = require('child_process');
class TTS {
constructor(channel, speed) {
this.speed = speed;
this.channel = channel;
this.baseCommand = "$speak = New-Object -ComObject SAPI.SPVoice;"
}
speak(text){
var command = this.baseCommand +
`$speak.AudioOutput = foreach ($o in $speak.GetAudioOutputs()) {if ($o.getDescription() -eq '${this.channel}') {$o; break;}}; `
+ "$speak.Speak([Console]::In.ReadToEnd());"
this.child = childProcess.spawn('powershell', [command], {shell: true})
this.child.stdin.setEncoding('ascii')
this.child.stdin.end(text);
this.child.addListener('exit', (code, signal) => {
if (code === null || signal !== null) {
console.log(new Error(`error [code: ${code}] [signal: ${signal}]`))
}
this.child = null
})
}
}
Then I can pass in an audio channel like
tts = new TTS("CABLE Input (VB-Audio Virtual Cable)", 0);
tts.speak("Hello there");
and it will output TTS in my desired channel
Some of the browsers support the built in speechSynthesis API.
Save the following code in 'test.html' file and open the file in chrome web browser for testing the speech api.
<script>
//---------------- SpeechAPI Start ------------------------
function speak(message) {
try {
var utterance= new SpeechSynthesisUtterance("");
utterance.lang='en-GB';
utterance.text=message;
window.speechSynthesis.speak(utterance);
}catch(e){
console.log('Exception in speak : ' + e)
}
}
//---------------- SpeechAPI End ------------------------
</script>
<button onclick="speak('Hello, how are you doing?');">Press to speak</button>
Is this what you are looking for ?
I'm using the videojs-playlist plugin along with Google's videojs-ima plugin. Everything works swimmingly except I am only getting a preload ad before the first video. I want one before each video in the playlist.
Basic setup is boilerplate, but for reference:
this.player = videojs('currentvideo', { autoplay : true, fluid : true });
this.player.playlist(this.playlist);
this.player.playlist.autoadvance(5);
const skippable_linear = {google's test ad};
const options = {
id: 'currentvideo',
adTagUrl: skippable_linear,
debug : true
};
this.player.ima(
options
);
this.player.ima.requestAds();
I have tried various ways of manually calling ads from inside an 'ended' event handler, such as calling requestAds again:
const _this = this;
this.player.on( 'ended', function(){
/* some other stuff */
_this.player.ima.requestAds();
});
This does play an ad where I want it, but
this breaks playlist's 'autoadvance' setting (next video doesn't start playing when the ad is finished), and
this puts the player into "ad display" mode (scrubber is unavailable, etc).
Is there a simple way to just say, "play an ad now" programmatically? I've tried, without joy, to use all of the seemingly applicable methods exposed by both the ima plugin and the contrib-ads plugin it relies on. I'll admit here that this is the first time I've ever had to deal with videos that run ads, so I'm kind of a noob.
I am trying to do the same thing. Just like you I failed when calling player.ima.requestAds() on events. I dug deeper and the best I could come up with is what I share bellow.
According to the videojs-ima API you have to use the setContentWithAdTag method instead of whatever you are using to switch the player content. In our case it is the player.playlist.next method.
I combined the code found in the videojs-ima examples with the original playlist.next to write my own next.
Then quite brutally I overrode the original plugin method.
Here's the code:
player.playlist(myPlayilst);
player.playlist.autoadvance(2);
player.playlistUi(); //videojs-playlist-ui
player.ima({
id: 'video5',
adTagUrl: 'thy adserver request'
});
//override playlist.next
player.playlist.next = function(){
var nextIndex = 0,
playlist = this.player_.playlist,
list = this.player_.playlist();
//everything below is copied directly from the original `next` (except for the "//load with ad")
// Repeat
if (playlist.repeat_) {
nextIndex = playlist.currentIndex_ + 1;
if (nextIndex > list.length - 1) {
nextIndex = 0;
}
} else {
// Don't go past the end of the playlist.
nextIndex = Math.min(playlist.currentIndex_ + 1, list.length - 1);
}
// Make the change
if (nextIndex !== playlist.currentIndex_) {
//load with ad
this.player_.playlist.currentItem(nextIndex);
this.player_.ima.setContentWithAdTag(
this.player_.playlist.currentItem(),
null,
true);
this.player_.ima.requestAds();
/////
return list[playlist.currentItem()];
}
}
You will probably need to override other methods that change the current playback, like playlist.previous.
I use videojs-playlist-ui so in my case it was neccessary to change the onclick handler called switchPlaylistItem_. I used some good old brute force to do that like this:
videojs.getComponent('PlaylistMenuItem').prototype.switchPlaylistItem_ = function(e){
this.player_.playlist.currentItem(this.player_.playlist().indexOf(this.item));
this.player_.ima.setContentWithAdTag(
this.player_.playlist.currentItem(),
null,
true);
this.player_.ima.requestAds();
};
PlaylistMenuItem's prototype should be changed before initializing the player.
This solution works, but it feels hacky, so if anyone can come up with something cleaner, please share!
I ended up forking videojs-playlist, and adding the option to override the player.src method. Feel free to use it:
fw-videojs-playlist
Details on how to use it are all in the github readme (including an example with ima.setContentWithAdTag)
I'm writing to ask you some advices for a particular problem regarding SAPI engine. I have an application that can speak both to the speakers and to a WAV file. I also need some events to be aware, i.e. word boundary and end input.
m_cpVoice->SetNotifyWindowMessage(m_hWnd, TTS_MSG, 0, 0);
hr = m_cpVoice->SetInterest(SPFEI_ALL_EVENTS, SPFEI_ALL_EVENTS);
Just for test I added all events! When the engine speaks to speakers all events are triggered and sent to the m_hWnd window, but when I set output to the WAV file, none of them are sent
CSpStreamFormat fmt;
CComPtr<ISpStreamFormat> pOld;
m_cpVoice->GetOutputStream(&pOld);
fmt.AssignFormat(pOld);
SPBindToFile(file, SPFM_CREATE_ALWAYS, &m_wavStream, &fmt.FormatId(), fmt.WaveFormatExPtr());
m_cpVoice->SetOutput(m_wavStream, false);
m_cpVoice->Speak(L"Test", SPF_ASYNC, 0);
Where file is a path passed as argument.
Really this code is taken from the TTS samples found on the SAPI SDK. It seems a little bit obscure the part setting the format...
Can you help me in finding the problem? Or does anyone of you know a better way to write TTS to WAV? I can not use manager code, it should be better to use the C++ version...
Thank you very much for help
EDIT 1
This seems to be a thread problem and searching in the spuihelp.h file, that contains the SPBindToFile helper I found that it uses the CoCreateInstance() function to create the stream. Maybe this is where the ISpVoice object looses its ability to send event in its creation thread.
What do you think about that?
I adopted an on-the-fly solution that I think should be acceptable in most of the cases, In fact when you write speech on files, the major event you would be aware is the "stop" event.
So... take a look a the class definition:
#define TTS_WAV_SAVED_MSG 5000
#define TTS_WAV_ERROR_MSG 5001
class CSpeech {
public:
CSpeech(HWND); // needed for the notifications
...
private:
HWND m_hWnd;
CComPtr<ISpVoice> m_cpVoice;
...
std::thread* m_thread;
void WriteToWave();
void SpeakToWave(LPCWSTR, LPCWSTR);
};
I implemented the method SpeakToWav as follows
// Global variables (***)
LPCWSTR tMsg;
LPCWSTR tFile;
long tRate;
HWND tHwnd;
ISpObjectToken* pToken;
void CSpeech::SpeakToWave(LPCWSTR file, LPCWSTR msg) {
// Using, for example wcscpy_s:
// tMsg <- msg;
// tFile <- file;
tHwnd = m_hWnd;
m_cpVoice->GetRate(&tRate);
m_cpVoice->GetVoice(&pToken);
if(m_thread == NULL)
m_thread = new std::thread(&CSpeech::WriteToWave, this);
}
And now... take a look at the WriteToWave() method:
void CSpeech::WriteToWav() {
// create a new ISpVoice that exists only in this
// new thread, so we need to
//
// CoInitialize(...) and...
// CoCreateInstance(...)
// Now set the voice, i.e.
// rate with global tRate,
// voice token with global pToken
// output format and...
// bind the stream using tFile as I did in the
// code listed in my question
cpVoice->Speak(tMsg, SPF_PURGEBEFORESPEAK, 0);
...
Now, because we did not used the SPF_ASYNC flag the call is blocking, but because we are on a separate thread the main thread can continue. After the Speak() method finished the new thread can continue as follow:
...
if(/* Speak is went ok */)
::PostMessage(tHwn, TTS_WAV_SAVED_MSG, 0, 0);
else
::PostMessage(tHwnd, TTS_WAV_ERROR_MSG, 0, 0);
}
(***) OK! using global variables is not quite cool :) but I was going fast. Maybe using a thread with the std::reference_wrapper to pass parameters would be more elegant!
Obviously, when receiving the TTS messages you need to clean the thread for a next time call! This can be done using a CSpeech::CleanThread() method like this:
void CSpeech::CleanThread() {
m_thread->join(); // I prefer to be sure the thread has finished!
delete m_thread;
m_thread = NULL;
}
What do you think about this solution? Too complex?
I'm trying to make a program in Spotify that collects the audio data. I saw in the API that there is a callback get_audio_buffer_stats, which has stutter and samples. I tried adding that to the program (I am just modifying the jukebox example), but it only ever prints 0 for stutter and samples, even when I turn off the wifi and wait for the song to stop playing. And by adding the code, I mean that I made a callback function for it, and I added it to the session callbacks. Am I missing something? Can anyone help me to get this callback to work? Thanks! My code is below:
static void get_audio_buffer_stats(sp_session *sess, sp_audio_buffer_stats *stats)
{
pthread_mutex_lock(&g_notify_mutex);
//log session data
stuttervariable = stats->stutter;
samplesvariable = stats->samples;
printf("stutter, %d\n", stuttervariable);
printf("samples, %d\n", samplesvariable);
pthread_cond_signal(&g_notify_cond);
pthread_mutex_unlock(&g_notify_mutex);
}
/**
* The session callbacks
*/
static sp_session_callbacks session_callbacks = {
.logged_in = &logged_in,
.notify_main_thread = ¬ify_main_thread,
.music_delivery = &music_delivery,
.metadata_updated = &metadata_updated,
.play_token_lost = &play_token_lost,
.log_message = NULL,
.end_of_track = &end_of_track,
.get_audio_buffer_stats = &get_audio_buffer_stats,
};
I think the idea with get_audio_buffer_stats is that you are supposed to tell libspotify if you've suffered stuttering and how many samples are left in your buffer. When it calls get_audio_buffer_stats, it passes a pointer to a struct that you are supposed to fill in. Presumably if you tell libspotify that you're suffering stutter it will try to send you a bit more data to keep your buffer more full. By telling libspotify how full your buffer is, it can accommodate for drift in your clock causing you to consume audio slightly faster or slower than it expects.
I have a helper method which encrypts some data on the iPhone. If the operatin is interrupted because the device gets locked, I want to delete the file I have just been processing. Therefore I add a notifiaction listsner if the method is called.
Two issues:
1. I get a warning that the method I use to add the listener is obsolete. How else would I do it?
2. If processing is done I would like to get rid of the listener - but how?
private static foo(string sDestPathAndFile)
{
NSNotificationCenter.DefaultCenter.AddObserver ( "UIApplicationProtectedDataWillBecomeUnavailable",
delegate( NSNotification oNotification )
{
Util.DeleteFile ( sDestPathAndFile );
throw new InvalidOperationException ( "Protected data became unavailable - device locked?" );
} );
// Do some processing here.
// ...
// Now get rid of the notification listener - but how?
}
To get rid of the obsolete warning, you should use the following:
NSNotificationCenter.DefaultCenter.AddObserver(UIApplication.ProtectedDataWillBecomeUnavailable, Handler);
This applies to all observers, e.g.:
UIKeyboard.WillHideNotification
UIKeyboard.WillShowNotification
UIDevice.OrientationDidChangeNotification
and so on. These are the appropriate NSString's that NSNotificationCenter is expecting.
As for getting rid of it, I can't verify this first hand as I'm currently not in a position to do so but one possible way is:
Declare the addobserver as an NSObject, then use the NSNotificationCenter.DefaultCenter.RemoveObserver to remove it:
NSObject obj = NSNotificationCenter.DefaultCenter.AddObserver(UIApplication.ProtectedDataWillBecomeUnavailable, handler);
// do whatever you need to do
// time to remove:
NSNotificationCenter.DefaultCenter.RemoveObserver(obj);