I have a WAV file that has been recorded using a custom codec. The codec has been installed on my machine, and the WAV file plays fine on my machine using Windows Media player. I am using the NAudio routines to try and play the WAV file via some C# code. The values for the custom format look weird, but I have painstakingly checked them by analysing the WAV file header. Here is the best C# code I have come up with for playing the file:
WaveFormat wfOKI = WaveFormat.CreateCustomFormat(WaveFormatEncoding.DialogicOkiAdpcm, 8000, 1, 3000, 48, 3);
WaveStream wsRaw = new WaveFileReader(txtFileName.Text);
wsRaw = WaveFormatConversionStream.CreatePcmStream(wsRaw); // Line A
wsRaw = new BlockAlignReductionStream(wsRaw); // Line B
WaveStream wsOKI = new RawSourceWaveStream(wsRaw, wfOKI);
WaveOut woCall = new WaveOut();
woCall.Init(wsOKI); // <-- This line gives an error.
woCall.Play();
while (woCall.PlaybackState == PlaybackState.Playing)
{
System.Threading.Thread.Sleep(300);
}
The Init() causes the following error: An unhandled exception of type 'NAudio.MmException' occurred in NAudio.dll. Additional information: WaveBadFormat calling waveOutOpen.
Is the code the correct strategy for playing a WAV with a custom codec? I have tried all four combinations of commenting out/in Lines A and B (with no difference to the error message).
I'm using Windows 7 64-bit, Visual Studio 2010 professional (project is set to x86), and version 1.6 of NAudio. I'm very new to NAudio, but did get a few lines going that played a "standard" WAV (i.e. a file that did not use a custom codec).
If you have got a WAV file then there is no need for the RawSourceWaveStream. You can play the converted stream directly.
var wsRaw = new WaveFileReader(txtFileName.Text);
wsRaw = WaveFormatConversionStream.CreatePcmStream(wsRaw);
WaveOut woCall = new WaveOut();
woCall.Init(wsRaw);
woCall.Play();
Also, you should not be calling thread.sleep to wait for it to finish if you are using WaveOut. Try WaveOutEvent instead if you are not using this from an application with a GUI.
Related
In my app I am recording voice using AudioRecorder as given in the following site, Audio Recorder it is working but it produce large size WAV file.
For example : If I record audio for 1 minute it takes 4MB to 5MB. So that I want to convert the wave file into MP3 file to reduce the size of the file. Please help me to compress the wav file ,give some example. Thanks in advance.
I never tried converting files before so i looked up on
some threads that might be helpful to you.
One is converting wav to mp3 which require file conversion into a byte[]
public byte[] ConvertToMp3(Uri uri)
{
using (var client = new WebClient())
{
var file = client.DownloadData(uri);
var target = new WaveFormat(8000, 16, 1);
using (var outPutStream = new MemoryStream())
using (var waveStream = new WaveFileReader(new MemoryStream(file)))
using (var conversionStream = new WaveFormatConversionStream(target, waveStream))
using (var writer = new LameMP3FileWriter(outPutStream, conversionStream.WaveFormat, 32, null))
{
conversionStream.CopyTo(writer);
return outPutStream.ToArray();
}
}
}
however on this method he is using a third party service which downloads the
wav file and then to be called on that method but this does not guaranty if the file size will be reduced.
however i have check that you can compress wav files using a library called zlib.
just decompress it whenever u need it.
Please check the link below:
How to convert wav file to mp3 in memory?
Reducing WAV sound file size, without losing quality
I'm using SharpAvi's AviWriter to create AVI files. I can get any standard video player to play these AVI files. 20 frames per second with no compression. using example from this link
so that seems to be working fine.
failing to find SharpAVi's AviReader anywhere I've resorted to using AForge.Video.VFW's AVIReader but it can only show me a black screen for each frame before index 128 and then if fails to get next frame.
the example I'm using is straighforward
// instantiate AVI reader
AVIReader reader = new AVIReader( );
// open video file
reader.Open( "test.avi" );
// read the video file
while ( reader.Position - reader.Start < reader.Length )
{
// get next frame
Bitmap image = reader.GetNextFrame( );
// .. process the frame somehow or display it
}
I have my app's Build set to x86 to accommodate both these AVI apps 32 bit settings.
and using AForge.Video.VFW AVIWriter fails to write files with more than some 500 +/- frames.(video player needs to rebuild index and C#IDE "fails opening AVI file".
does SharpAVI have an AVIReader? because I haven't found one.
I'm writing a thing that uses the SpeechSynthesizer to generate wave files on request, but I'm having problems with crackling noises. The weird thing is that output directly to the sound card is just fine.
This short powershell script demonstrates the issue, though I'm writing my program in C#.
Add-Type -AssemblyName System.Speech
$speech = New-Object System.Speech.Synthesis.SpeechSynthesizer
$speech.Speak('Guybrush Threepwood, mighty pirate!')
$speech.SetOutputToWaveFile("${PSScriptRoot}\foo.wav")
$speech.Speak('Guybrush Threepwood, mighty pirate!')
What this should do, is output to the speakers, and then save that same sound as "foo.wav" next to the script.
What it does is output to the speakers, and then save a crackling, old record player sounding version as a wave file. I've tested this on three different machines, and though they select different voices by default (all Microsoft provided default ones), they all sound like garbage falling down stairs in the wave file.
Why?
EDIT: I am testing this on Windows 10 Pro, with the latest updates that add that annoying "People" button on the taskbar.
EDIT 2: Here's a link to an example sound generated with the above script. Notice the crackling voice, that's not there when the script outputs directly to the speakers.
EDIT 3: It's even more noticeable with a female voice
EDIT 4: The same voice as above, saved to file with TextAloud 3 - no cracking, no vertical spikes.
I find it hard to believe this is a PoSH issue.
It's not PoSH doing the encoding on the serialization to disk. Its the API/Class that is being used.
'msdn.microsoft.com/en-us/library/system.speech.synthesis.speechsynthesizer(v=vs.110).aspx'
As per the MSDN, there is no option to control the encoding, bit rate, etc.
.wav has never been HQ stuff. So, I'd wonder if you take that .wav through a converter to make it an .mp3 or mp4, if that would correct your quality concerns. But that also means getting the converter on users systems.
Secondly, since Win8, the default player does not even play .wav correctly or at all. Sure, you can still set the default play of .wav to Windows Media Player or call the file via VLC, but it's still a .wav file. Yet, that also means, you having to set the Media Player assignment on every target system.
This is an issue with the SpeechSynthesizer API, which simply provides bad quality, crackling audio as seen in the samples above. The solution is to do what TextAloud does, which is to use the SpeechLib COM objects directly.
This is done by adding a COM reference to "Microsoft Speech Object Library (5.4)". Here is a snippet of the code I ended up with, which produces audio clips of the same quality as TextAloud:
public new static byte[] GetSound(Order o)
{
const SpeechVoiceSpeakFlags speechFlags = SpeechVoiceSpeakFlags.SVSFlagsAsync;
var synth = new SpVoice();
var wave = new SpMemoryStream();
var voices = synth.GetVoices();
try
{
// synth setup
synth.Volume = Math.Max(1, Math.Min(100, o.Volume ?? 100));
synth.Rate = Math.Max(-10, Math.Min(10, o.Rate ?? 0));
foreach (SpObjectToken voice in voices)
{
if (voice.GetAttribute("Name") == o.Voice.Name)
{
synth.Voice = voice;
}
}
wave.Format.Type = SpeechAudioFormatType.SAFT22kHz16BitMono;
synth.AudioOutputStream = wave;
synth.Speak(o.Text, speechFlags);
synth.WaitUntilDone(Timeout.Infinite);
var waveFormat = new WaveFormat(22050, 16, 1);
using (var ms = new MemoryStream((byte[])wave.GetData()))
using (var reader = new RawSourceWaveStream(ms, waveFormat))
using (var outStream = new MemoryStream())
using (var writer = new WaveFileWriter(outStream, waveFormat))
{
reader.CopyTo(writer);
return o.Mp3 ? ConvertToMp3(outStream) : outStream.GetBuffer();
}
}
finally
{
Marshal.ReleaseComObject(voices);
Marshal.ReleaseComObject(wave);
Marshal.ReleaseComObject(synth);
}
}
This is the code to convert a wave file to mp3. It uses NAudio.Lame from nuget.
internal static byte[] ConvertToMp3(Stream wave)
{
wave.Position = 0;
using (var mp3 = new MemoryStream())
using (var reader = new WaveFileReader(wave))
using (var writer = new LameMP3FileWriter(mp3, reader.WaveFormat, 128))
{
reader.CopyTo(writer);
mp3.Position = 0;
return mp3.ToArray();
}
}
I've written a video generator that rights a video in h264 format (mp4). When I stream the video from my azure service, i'm seeing the following network traffic:
The AVCodecContext layout I'm using is as follows:
AVCodec* videoCodec = ffmpeg.avcodec_find_encoder(AVCodecID.AV_CODEC_ID_H264)
AVCodecContext* videoCodecContext = ffmpeg.avcodec_alloc_context3(videoCodec);
videoCodecContext->bit_rate = 400000;
videoCodecContext->width = 1280;
videoCodecContext->height = 720;
videoCodecContext->gop_size = 12;
videoCodecContext->max_b_frames = 1;
videoCodecContext->pix_fmt = videoCodec->pix_fmts[0];
videoCodecContext->codec_id = videoCodec->id;
videoCodecContext->codec_type = videoCodec->type;
videoCodecContext->time_base = new AVRational
{
num = 1,
den = 30
};
ffmpeg.av_opt_set(videoCodecContext->priv_data, "preset", "ultrafast");
I'm also tried setting the "movflags" option for avformat_write_header() via an AVDictionary, but then av_write_trailer() returns -2, cause the file to not finish writing.
I cannot figure out how to solve this problem. Videos generating using Windows Movie Maker stream perfectly.
I know this has something to do with mdat and mov positions.
Also, this appears to only happening in Google Chrome.
OK, figured this out. I've been writing the video frames first and the audio frames afterwards. Instead, you have to write them side by side in order for faststart to actually work and allow the video to stream.
So, write a specific amount of audio and then determine if a video frame should be written by checking the timebases against the current writing indexes.
This example will show you how its done.
Also, to get the video and audio streams to have accurate PTS/DTS values, look at this question.
In a text to speech application by c# I use SpeechSynthesizer class, it has an event named SpeakProgress which is fired for every spoken word. But for some voices including "Microsoft Anna" the parameter e.AudioPosition is not synchronized with the output audio stream, and the audio stream is played faster than what AuidoPosition indicates.
void reader_SpeakProgress(object sender, SpeakProgressEventArgs e)
{
Console.Write(e.AudioPosition + "");
}
I thought maybe the problem is the bitrate and the WaveStream as the output which I use as the following.
FileStream AudioStream
= new FileStream(fname, FileMode.Create, FileAccess.Write);
reader.SetOutputToWaveStream(AudioStream);
I tried
var formats = CurVoice.VoiceInfo.SupportedAudioFormats;
reader.SetOutputToAudioStream(AudioStream, formats[0]);
But the problem now is that the output file is not played.
Do you have any suggestions for why the voice is not synch and why my solution has no
playable output audio?
As I have guessed the problem was related to bitrate, to enforce the audio format I used an alternative method named SetOutputToWaveFile
var formats = CurVoice.VoiceInfo.SupportedAudioFormats;
reader.SetOutputToWaveFile(fname, formats[0]);
with the code above, the audio output is playable and also the synching problem resolved!