When you create a WaveOut object, and initialize it with WaveOffsetStream, the PlaybackStopped event is not raised at the end of the playback. Code:
WaveOut myWaveOut = new WaveOut();
myWaveOut.PlaybackStopped += OnPlaybackStopped;
WaveOffsetStream OffsetStream = new WaveOffsetStream(MyOtherStream);
myWaveOut.Init(OffsetStream);
myWaveOut.Play();
WaveOutOffsetStream always returns the requested number of bytes from Read so it is a never-ending stream. You'd have to use something different, or detect when the Position went past the point you wanted to play to
Related
I have created a global tweepy stream and I have one function where I want to disconnect it if it's active and after to start to track the word received as argument (with async=True).
When I call the function for the first time it's working properly but after I receive errors.
Here is the sequence where I m creating the stream:
myStreamListener = MyStreamListener()
myStream = tweepy.Stream(auth = api.auth, listener = myStreamListener)
And here I use it:
def cauta_dupa_cuvant(word):
myStream.disconnect()
myStream.filter(languages=["en"], track = [word],is_async=True)
Any idea?
I need to convert audio data from AV_CODEC_ID_PCM_S16LE to AV_CODEC_ID_PCM_ALAW and I am using this code as an example. The example code does essentially this (error checking omitted for brevity):
const AVCodec* codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
AVCodecContext* c = avcodec_alloc_context3(codec);
c->bit_rate = 64000;
c->sample_fmt = AV_SAMPLE_FMT_S16;
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
avcodec_open2(c, codec, NULL);
AVFrame* frame = av_frame_alloc();
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
The example code subsequently uses c->frame_size in a for loop.
My code is similar to the above with the following differences:
const AVCodec* codec = avcodec_find_encoder(AV_CODEC_ID_PCM_ALAW);
c->sample_rate = 8000;
c->channel_layout = AV_CH_LAYOUT_MONO;
c->channels = 1;
After calling avcodec_open2, c->frame_size is zero. The example code never sets the frame size so I assume that it expects either avcodec_alloc_context3 or avcodec_open2 to set it. Is this a correct assumption? Is the setting of the frame size based on the codec being used? If I have to set the frame size explicitly, is there a recommended size?
EDIT:
Based on #the-kamilz answer it appears that the example code is not robust. The example assumes that c->frame_size will be set but that appears to be dependent on the codec. In my case, codec->capabilities was in fact set to AV_CODEC_CAP_VARIABLE_FRAME_SIZE. So I modified my code to check c->frame_size and use it only if it is not zero. If it is zero, I just picked an arbitrary one second worth of data for frame->nb_samples.
In the FFmpeg documentation it is mentioned as:
int AVCodecContext::frame_size
Number of samples per channel in an audio frame.
encoding: set by libavcodec in avcodec_open2(). Each submitted frame except the last must contain exactly frame_size samples per channel.
May be 0 when the codec has AV_CODEC_CAP_VARIABLE_FRAME_SIZE set, then
the frame size is not restricted.
decoding: may be set by some decoders to indicate constant frame size
Hope that helps.
you don't control the frame size explicitly, it is set by the encoder depending on the codecs provided at initialization (opening) time
once avcodec_open2() is successful, you can retrieve the frame's buffer size with av_samples_get_buffer_size()
I'm writing a MOD player, trying to playback a sample using Allegro5 raw stream capabilities, I can't figure out the exact init parameters for the stream to play the loaded sample data from the mod file.
This is what I have:
xf::ModLoader ml;
ml.loadFromFile("C:\\Users\\bubu\\Downloads\\agress.mod");
// getSampleLength() returns # of data words
int sample_length = ml.getSampleLength(1) * 2;
const int8_t* sample_data = ml.getSampleData(1);
ALLEGRO_MIXER* mixer = al_get_default_mixer();
ALLEGRO_AUDIO_STREAM* stream = al_create_audio_stream(1, sample_length, 8363, ALLEGRO_AUDIO_DEPTH_INT8, ALLEGRO_CHANNEL_CONF_1);
al_attach_audio_stream_to_mixer(stream, mixer);
al_set_audio_stream_gain(stream, 0.7f);
al_set_audio_stream_playmode(stream, ALLEGRO_PLAYMODE_ONCE);
al_set_audio_stream_playing(stream, true);
al_set_audio_stream_fragment(stream, (void*)sample_data);
al_drain_audio_stream(stream);
First of all, freq param is hardcoded for the test (8363Hz), but, playing back at the specified frequency I don't get what I expect, and al_drain_audio_stream() gets stuck forever playing garbage in a loop...
Any help would be appreciated.
At the very least, you need to be calling al_get_audio_stream_fragment before you call al_set_audio_stream_fragment. Typically you'd feed these streams in a while loop, while responding to the ALLEGRO_EVENT_AUDIO_STREAM_FRAGMENT event. See the ex_saw example in the Allegro's source for some sample code: https://github.com/liballeg/allegro5/blob/master/examples/ex_saw.c
I use the vlc-qt lib and try to access the frames per second information of an openend video file.
The player:
_instance = new VlcInstance(VlcCommon::args(), this);
_player = new VlcMediaPlayer(_instance);
_media = new VlcMedia(file, true, _instance);
_player->openOnly(_media);
the player has a public function playbackRate() but it gives only the current playback ration, so it is 1 if no slowmotion is applied.
float playbackRate = _player->playbackRate();
I also tried to get it over the codec, but the codec itself is not a class but only an enum with possible codecnames.
How can I access the fps, so get back something like 30 frames per second?
Using python vlc.py:
Where self.player is:
self.Instance = vlc.Instance()
self.player = self.Instance.media_player_new()
It has a function get_fps()
def mspf(self):# Milliseconds per frame.
return int(1000 // (self.player.get_fps() or 25))
EDIT:
Having dug deeper, there appears to be no reference to fps in the vlc-qt sources, except where it offers fps as an experimental input to the Media::duplicate and Media::record functions within VlcMedia.cpp although fps is available within vlc
Hello i have following code to play raw file.my raw file duration is 25 second. this code is working fine but after sometime my program play raw file very slow almost 50% speed and my raw file duration is increase 36 second when i restart my pc and run my program its working normaly again. i need to restart my pc every one hour to work correctly please check what is wrong with my code here is my code
Try
Dim rawStream = File.OpenRead("C:\myFile.raw")
Dim waveFormat = New NAudio.Wave.WaveFormat(8000, 16, 1)
Dim rawSource = New RawSourceWaveStream(rawStream, waveFormat)
Dim audioBufferSize = 320
Dim offset As Integer = 0
Dim buffer As Byte() = New Byte(audioBufferSize - 1) {}
Dim buffer2 As Byte() = New Byte(320) {}
While (rawSource.Read(buffer2, offset, audioBufferSize) > 0)
msport.Write(buffer2, offset, audioBufferSize)
Thread.Sleep(20)
End While
Catch ex As Exception
MsgBox(ex.ToString)
End Try
NAudio is not having any effect at all in your code sample. All you are doing is reading data from a file, and sending it to the serial port. RawSourceWaveStream simply attaches a WaveFormat to the file stream, but nothing is reading that stream. Whatever device you have listening to the audio data you're sending over the serial port will have an audio format it is expecting. You need to find out what that is. Then you can use NAudio to convert the audio to the correct format if it is not already at the right sample rate / channel count (which would be the two most likely causes of audio playing at 50% speed)