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
Related
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'm completely new to QMultimedia. At the moment, I try to get the audio stream from the microphone in my webcam for further processing. Right now I just try to continuously show the volume level of the sound "heard" by the mic with a slider. So I googled some code together (found nearly 10 tons of examples how I can play an audio, but only a few blocks of C++ code about audio input) and got stuck.
This is my actual code:
import sys, time
from PyQt4 import Qt, QtGui, QtCore, QtMultimedia
class VolumeSlider(QtGui.QSlider):
def __init__(self, parent=None):
super(VolumeSlider, self).__init__(parent)
self.audio = None
self.volumeSlider = QtGui.QSlider(QtCore.Qt.Horizontal)
self.volumeSlider.setTickInterval(1)
self.volumeSlider.setMaximum(100)
self.volumeSlider.setValue(49)
self.volumeSlider.show()
self.openMicStream()
# THIS IS WHAT I WANT - DOESN'T WORK
while True:
self.volumeSlider.setValue(self.audio.volume())
time.sleep(0.02)
def openMicStream( self ):
#audioInputDevices = QtMultimedia.QAudioDeviceInfo.availableDevices(QtMultimedia.QAudio.AudioInput)
#for d in audioInputDevices: d.deviceName()
info = QtMultimedia.QAudioDeviceInfo(QtMultimedia.QAudioDeviceInfo.defaultInputDevice())
print "Default audio input device:", info.deviceName()
audioFormat = QtMultimedia.QAudioFormat()
audioFormat.setFrequency(8000);
audioFormat.setChannels(1);
audioFormat.setSampleSize(8);
audioFormat.setCodec("audio/pcm");
audioFormat.setByteOrder(QtMultimedia.QAudioFormat.LittleEndian);
audioFormat.setSampleType(QtMultimedia.QAudioFormat.UnSignedInt);
audioDeviceInfo = QtMultimedia.QAudioDeviceInfo.defaultInputDevice();
if not audioDeviceInfo.isFormatSupported(audioFormat):
sys.stderr("default audioFormat not supported try to use nearest")
audioFormat = audioDeviceInfo.nearestFormat(audioFormat);
self.audioInput = QtMultimedia.QAudioInput(audioFormat);
fmtSupported = info.isFormatSupported(audioFormat)
print "Is the selected format supported?", fmtSupported
if not fmtSupported:
audioFormat = info.nearestFormat(audioFormat)
print "Is the nearest format supported?", info.isFormatSupported(audioFormat)
self.audio = QtMultimedia.QAudioInput(audioFormat, None)
self.audio.start()
if __name__ == "__main__":
app = Qt.QApplication(sys.argv)
x = VolumeSlider()
sys.exit(app.exec_())
Could anybody poke me in the head what I have to do at the "#THIS IS WHAT I WANT" place to calculate and show the current level of volume?
There is no inbuilt function for computing the current volume level of the input sound signal when recorded with QAudioInput neither in Qt 4 (QAudioInput documentation) nor in Qt 5.
But you could calculate it for yourself. The root-mean-square in a moving window of the signal is often used as a measure for current loudness. See How can I determine how loud a WAV file will sound? for more suggestions.
Solved it after a while of working on another parts. Now I can at least hear the sound out of the boxes, after I changed the openMicStream(self) to this:
def openMicStream( self ):
info = QAudioDeviceInfo(QAudioDeviceInfo.defaultInputDevice())
print "Default audioInput input device: ", info.deviceName()
audioFormat = QAudioFormat()
audioFormat.setFrequency(44100);
audioFormat.setChannels(1);
audioFormat.setSampleSize(16);
audioFormat.setCodec("audioInput/pcm");
audioFormat.setByteOrder(QAudioFormat.LittleEndian);
audioFormat.setSampleType(QAudioFormat.UnSignedInt);
audioDeviceInfo = QAudioDeviceInfo.defaultInputDevice();
if not audioDeviceInfo.isFormatSupported(audioFormat):
messages.error(__name__, "default audioFormat not supported try to use nearest")
audioFormat = audioDeviceInfo.nearestFormat(audioFormat);
print audioFormat.frequency()
print audioFormat.channels()
print audioFormat.sampleSize()
print audioFormat.codec()
print audioFormat.byteOrder()
print audioFormat.sampleType()
self.audioInput = QAudioInput(audioFormat);
audioFmtSupported = info.isFormatSupported(audioFormat)
messages.info(__name__, "Is the selected format supported?"+str(audioFmtSupported))
if not audioFmtSupported:
audioFormat = info.nearestFormat(audioFormat)
messages.info(__name__, "Is the nearest format supported?"+str(info.isFormatSupported(audioFormat)))
self.audioInput = QAudioInput(audioFormat, None)
self.audioOutput = QAudioOutput(audioFormat, None)
device = self.audioOutput.start()
self.audioInput.start(device)
First of all, i am new to audio-programming, so bear with me.
I am trying to play spotify music with NAudio or BASS.Net or any other .net audio-library.
As far as i known, libspotify delivers music as raw PCM data. what is the sample rate of spotify stream (libspotify)?
From the spotify docs:
Samples are delivered as integers, see sp_audioformat. One frame consists of the same number of samples as there are channels. I.e. interleaving is on the sample level.
When i try to play a song, spotify makes a callback with a 8192 byte buffer
channels = 2
sample_rate = 44100
num_frames = 2048
I need a little help translating this information to NAudio terms.
I have also tried with a spotify to Bass.Net sample (BASSPlayer.cs). But i haven't heard a single note from my speakers yet.
I have tried to play an mp3-song with NAudio and Bass.NET and this works fine, so the speaker volume is ok.
https://github.com/Alxandr/SpotiFire/blob/master/SpotiFire.Server/BASSPlayer.cs
There is breakthrough with NAudio. This is what i came up with, by using the trial and error method. I'm not sure if this is the right way to calculate the parameters from sampleRate/channels...
But the song is playing :-)
IWavePlayer waveOutDevice = new WaveOut();
using (var pcmStream = new FileStream(PcmFile, FileMode.Open))
{
const int songDuration = 264000;
const int sampleRate = 44100;
const int channels = 2;
var waveFormat = WaveFormat.CreateCustomFormat(WaveFormatEncoding.Pcm, sampleRate * channels, 1, sampleRate * 2 * channels, channels, 16);
var waveStream = new RawSourceWaveStream(pcmStream, waveFormat);
waveOutDevice.Init(waveStream);
waveOutDevice.Play();
Thread.Sleep(songDuration);
waveOutDevice.Stop();
waveStream.Close();
waveOutDevice.Dispose();
}
I am using Media Foundation on client side to display live mpeg4 stream from AXIS Camera through RTSP server.
Client side video works very fine if i set GOV # camera = 1, i.e. camera will only send I-Frames. But if GOV is increased and camera starts sending P-Frames also, my video suddenly gets distorted at regular intervals. I cannot set GOV = 1 for always because it consumes a lot of bandwidth.
Following is the code for RequestSample method where i supply Samples to Media Foundation :
RTPFrame frame = null;
byte[] frameBytes = null;
frame = _VideoJitter.GetNextFrame();
frameBytes = frame.GetFrameAsBytes();
frame.FrameType= RTPFrame.PredictFrameType(frameBytes);
_videoEncapsulatedSample.ReadSampleFrom(frameBytes);
videoSample = _videoEncapsulatedSample.MfSample;
long timestamp = nextSampleTimestamp ?? 0;
videoSample.SetSampleTime(timestamp);
duration_video = (long)GetPresentationTime(frame);
videoSample.SetSampleDuration(duration_video);
nextSampleTimestamp = timestamp + duration_video;
if (frame.FrameType == FrameType.IFrame)
{
videoSample.SetUINT32(MFAttributesClsid.MFSampleExtension_CleanPoint, 1);
}
return videoSample;
Do i need to set any attribute for processing P-Frames??
Any help would be highly appreciated....
Update (2012/02/22) :
I ran some statistics and found that some times I-frames never reach client, i.e. suppose GOV = 15, so every 15th frame should be an I-Frame, but sometimes (at irregular intervals) client receives an I-Frame after 28 or 30 or 59 P-frames.
Any pointers ??
Thanks,
Prateek