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
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 have been trying to send a value from a Python program via serial to an Arduino, but I have been unable to get the Arduino to store and echo back the value to Python. My code seems to match that I've found in examples online, but for whatever reason, it's not working.
I am using Python 3.5 on Windows 10 with an Arduino Uno. Any help would be appreciated.
Arduino code:
void readFromPython() {
if (Serial.available() > 0) {
incomingIntegrationTime = Serial.parseInt();
// Collect the incoming integer from PySerial
integration_time = incomingIntegrationTime;
Serial.print("X");
Serial.print("The integration time is now ");
// read the incoming integer from Python:
// Set the integration time to what you just collected IF it is not a zero
Serial.println(integration_time);
Serial.print("\n");
integration_time=min(incomingIntegrationTime,2147483648);
// Ensure the integration time isn't outside the range of integers
integration_time=max(incomingIntegrationTime, 1);
// Ensure the integration time isn't outside the range of integers
}
}
void loop() {
readFromPython();
// Check for incoming data from PySerial
delay(1);
// Pause the program for 1 millisecond
}
Python code:
(Note this is used with a PyQt button, but any value could be typed in instead of self.IntegrationTimeInputTextbox.text() and the value is still not receieved and echoed back by Arduino).
def SetIntegrationTime(self):
def main():
# global startMarker, endMarker
#This sets the com port in PySerial to the port with the Genuino as the variable arduino_ports
arduino_ports = [
p.device
for p in serial.tools.list_ports.comports()
if 'Genuino' in p.description
]
#Set the proper baud rate for your spectrometer
baud = 115200
#This prints out the port that was found with the Genuino on it
ports = list(serial.tools.list_ports.comports())
for p in ports:
print ('Device is connected to: ', p)
# --------------------------- Error Handling ---------------------------
#Tell the user if no Genuino was found
if not arduino_ports:
raise IOError("No Arduino found")
#Tell the user if multiple Genuinos were found
if len(arduino_ports) > 1:
warnings.warn('Multiple Arduinos found - using the first')
# ---------------------------- Error Handling ---------------------------
#=====================================
spectrometer = serial.Serial(arduino_ports[0], baud)
integrationTimeSend = self.IntegrationTimeInputTextbox.text()
print("test value is", integrationTimeSend.encode())
spectrometer.write(integrationTimeSend.encode())
for i in range(10): #Repeat the following 10 times
startMarker = "X"
xDecoded = "qq"
xEncoded = "qq"
while xDecoded != startMarker: #Wait for the start marker (X))
xEncoded = spectrometer.read() #Read the spectrometer until 'startMarker' is found so the right amound of data is read every time
xDecoded = xEncoded.decode("UTF-8");
print(xDecoded);
line = spectrometer.readline()
lineDecoded = line.decode("UTF-8")
print(lineDecoded)
#=====================================
spectrometer.close()
#===========================================
#WaitForArduinoData()
main()
First, this is a problem:
incomingValue = Serial.read();
Because read() returns the first byte of incoming serial data reference. On the Arduino the int is a signed 16-bit integer, so reading only one byte of it with a Serial.read() is going to give you unintended results.
Also, don't put writes in between checking if data is available and actual reading:
if (Serial.available() > 0) {
Serial.print("X"); // Print your startmarker
Serial.print("The value is now ");
incomingValue = Serial.read(); // Collect the incoming value from
That is bad. Instead do your read immediately as this example shows:
if (Serial.available() > 0) {
// read the incoming byte:
incomingByte = Serial.read();
That's two big issues there. Take care of those and let's take a look at it after those fundamental issues are corrected.
PART 2
Once those are corrected, the next thing to do is determine which side of the serial communication is faulty. Generally what I like to do is determine one side is sending properly by having its output show up in a terminal emulator. I like TeraTerm for this.
Set your python code to send only and see if your sent values show up properly in a terminal emulator. Once that is working and you have confidence in it, you can attend to the Arduino side.
I have downloaded the audio-echo app from the android NDK portal for opensl. Due to the lack of documentation I'm not able to identify how to change the sampling rate and buffer size of the audio in and out.
If anybody has any idea on how to:
Change the buffer size and sampling rate on OpenSL
Read the buffers to be fed to a C code to be processed
Fed to the output module of OpenSL to be fed to the speakers
Another alternative I feel is read it at the preferred sampling rate and buffer size but downsample and upsample in the code itself and use a circular buffer to get desired data. But how are we reading and feeding the data in openSL?
In the OpenSL ES API, there are calls to create either a Player or a Recorder:
SLresult (*CreateAudioPlayer) (
SLEngineItf self,
SLObjectItf * pPlayer,
SLDataSource *pAudioSrc,
SLDataSink *pAudioSnk,
SLuint32 numInterfaces,
const SLInterfaceID * pInterfaceIds,
const SLboolean * pInterfaceRequired
);
SLresult (*CreateAudioRecorder) (
SLEngineItf self,
SLObjectItf * pRecorder,
SLDataSource *pAudioSrc,
SLDataSink *pAudioSnk,
SLuint32 numInterfaces,
const SLInterfaceID * pInterfaceIds,
const SLboolean * pInterfaceRequired
);
Note that both of these take a SLDataSource *pAudioSrc parameter.
To use a custom playback rate or recording rate, you have to set up this data source properly.
I use an 11Khz playback rate using this code:
// Configure data format.
SLDataFormat_PCM pcm;
pcm.formatType = SL_DATAFORMAT_PCM;
pcm.numChannels = 1;
pcm.samplesPerSec = SL_SAMPLINGRATE_11_025;
pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
pcm.containerSize = 16;
pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
// Configure Audio Source.
SLDataSource source;
source.pFormat = &pcm;
source.pLocator = &bufferQueue;
To feed data to the speakers, a buffer queue is used that is filled by a callback. To set this callback, use SLAndroidSimpleBufferQueueItf, documented in section 8.12 SLBufferQueueItf of the OpenGL ES specification.
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
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)