When I upsampled wav file(in PCM format) from 6khz to 16khz, and got the spectogram(shown in my google drive link spectrogram_img), I found that the high frequency part of the spectrogram was lost.
I have tried using sox and spicy.signal.resample to upsample, but both of them stucked in the same problem mentioned above.
wavsignal = signal.resample(w, (int)(len(w) / 6 * 16))
So why does it happen and how to keep the high frequency information when upsampling?
Related
I have a console program which I have used for years, for (among other things) displaying info about certain audio-file formats, including mp3. I used data from the mpeghdr site to calculate the frame sizes, in order to further calculate playing time for the tracks. The equation that I got from mpeghdr was:
// Read the BitRate, SampleRate and Padding of the frame header.
// For Layer I files use this formula:
//
// FrameLengthInBytes = (12 * BitRate / SampleRate + Padding) * 4
//
// For Layer II & III files use this formula:
//
// FrameLengthInBytes = 144 * BitRate / SampleRate + Padding
This works well for most mp3 files, but there have always been a small subset for whom this equation failed. Recently, I've been looking at a set of very small mp3 files, and have found that for these files this formula fails much more often, so I'm trying to finally nail down what is going on. All of these mp3 files were generated using Lame V3.100, with default settings, on Windows 7 64-bit.
In all cases, I can successfully find the first frame header, but when I used the above formula to calculate the offset to the next frame header, it is sometimes not correct.
As an example, I have a file 'wolf howl.mp3'; analytical files such as MPEGAudioInfo show frame size as 288 bytes. When I run my program, though, it shows length of first frame as 576 bytes (2 * 288). When I look at the mp3 file in a hex editor, with first frame at 0x154, I can see that the next frame is at 0x154 + 208 bytes, but this calculation does in fact result in 576 bytes...
File info:
mpegV2.5, layer III
frame: bitrate=32, sample_rate=8000, pad=0, bytes=576
mtemp->frame_length_in_bytes =
(144 * (mtemp->bitrate * 1000) / mtemp->sample_rate) + mtemp->padding_bit;
which equals 576
I've looked at numerous other references, and they all show this equation...
At first I thought is was an issue with MPEG 2.5, which is an unofficial standard, but I have also seen this with MPEG2 files as well. Only happens with small files, though.
Does anyone have any insights on what I am missing here??
//**************************************
Later notes:
I thought maybe audio format would be relevant to this issue, so I dumped channel_mode and mode_extension for each of my test files (3 calculate properly, 2 don't). Sadly, all of them are cmode=3, mode_ext=0
(i.e., last byte of the header is 0xC4)... so that doesn't help...
Okay, I found the answer to this queston... it was in the MPEGAudioInfo program on CodeProject site. Here is the vital key:
//*************************************************************************************
// This reference data is from MPEGAudioInfo app
// Samples per Frame / 8
static const u32 m_dwCoefficients[2][3] =
{
{ // MPEG 1
12, // Layer1 (must be multiplied with 4, because of slot size)
144, // Layer2
144 // Layer3
},
{ // MPEG 2, 2.5
12, // Layer1 (must be multiplied with 4, because of slot size)
144, // Layer2
72 // Layer3
}
};
It is unfortunately that none of the reference pages mention this detail !!
My program now successfully calculates frame sizes for all of my mp3 files, including the small ones.
I had the same problem. Some documents, I've read, don't define dividing by 2 in Frame-Size formula for MPEG2.5L3. But some src-code, I encountered - does.
It's hard to find out any proof.
I have nothing better than this link:
https://link.springer.com/chapter/10.1007/978-1-4615-0327-9_12
(it's better to share that link in "add a comment"-form, but I have insufficient rank)
I'm attempting to write a simple windows media foundation command line tool to use IMFSourceReader and IMFSyncWriter to load in a video, read the video and audio as uncompressed streams and re-encode them to H.246/AAC with some specific hard-coded settings.
The simple program Gist is here
sample video 1
sample video 2
sample video 3
(Note: the video's i've been testing with are all stereo, 48000k sample rate)
The program works, however in some cases when comparing the newly outputted video to the original in an editing program, I see that the copied video streams match, but the audio stream of the copy is pre-fixed with some amount of silence and the audio is offset, which is unacceptable in my situation.
audio samples:
original - |[audio1] [audio2] [audio3] [audio4] [audio5] ... etc
copy - |[silence] [silence] [silence] [audio1] [audio2] [audio3] ... etc
In cases like this the first video frames coming in have a non zero timestamp but the first audio frames do have a 0 timestamp.
I would like to be able to produce a copied video who's first frame from the video and audio streams is 0, so I first attempted to subtract that initial timestamp (videoOffset) from all subsequent video frames which produced the video i wanted, but resulted in this situation with the audio:
original - |[audio1] [audio2] [audio3] [audio4] [audio5] ... etc
copy - |[audio4] [audio5] [audio6] [audio7] [audio8] ... etc
The audio track is shifted now in the other direction by a small amount and still doesn't align. This can also happen sometimes when a video stream does have a starting timestamp of 0 yet WMF still cuts off some audio samples at the beginning anyway (see sample video 3)!
I've been able to fix this sync alignment and offset the video stream to start at 0 with the following code inserted at the point of passing the audio sample data to the IMFSinkWriter:
//inside read sample while loop
...
// LONGLONG llDuration has the currently read sample duration
// DWORD audioOffset has the global audio offset, starts as 0
// LONGLONG audioFrameTimestamp has the currently read sample timestamp
//add some random amount of silence in intervals of 1024 samples
static bool runOnce{ false };
if (!runOnce)
{
size_t numberOfSilenceBlocks = 1; //how to derive how many I need!? It's aribrary
size_t samples = 1024 * numberOfSilenceBlocks;
audioOffset = samples * 10000000 / audioSamplesPerSecond;
std::vector<uint8_t> silence(samples * audioChannels * bytesPerSample, 0);
WriteAudioBuffer(silence.data(), silence.size(), audioFrameTimeStamp, audioOffset);
runOnce= true;
}
LONGLONG audioTime = audioFrameTimeStamp + audioOffset;
WriteAudioBuffer(dataPtr, dataSize, audioTime, llDuration);
Oddly, this creates an output video file that matches the original.
original - |[audio1] [audio2] [audio3] [audio4] [audio5] ... etc
copy - |[audio1] [audio2] [audio3] [audio4] [audio5] ... etc
The solution was to insert extra silence in block sizes of 1024 at the beginning of the audio stream. It doesn't matter what the audio chunk sizes provided by IMFSourceReader are, the padding is in multiples of 1024.
My problem is that there seems to be no detectable reason for the the silence offset. Why do i need it? How do i know how much i need? I stumbled across the 1024 sample silence block solution after days of fighting this problem.
Some videos seem to only need 1 padding block, some need 2 or more, and some need no extra padding at all!
My question here are:
Does anyone know why this is happening?
Am I using Media Foundation incorrectly in this situation to cause this?
If I am correct, How can I use the video metadata to determine if i need to pad an audio stream and how many 1024 blocks of silence need to be in the pad?
EDIT:
For the sample videos above:
sample video 1 : the video stream starts at 0 and needs no extra blocks, passthrough of original data works fine.
sample video 2 : video stream starts at 834166 (hns) and needs 1 1024 block of silence to sync
sample video 3 : video stream starts at 0 and needs 2 1024 blocks of silence to sync.
UPDATE:
Other things I have tried:
Increasing the duration of the first video frame to account for the offset: Produces no effect.
I wrote another version of your program to handle NV12 format correctly (yours was not working) :
EncodeWithSourceReaderSinkWriter
I use Blender as video editing tools. Here is my results with Tuning_against_a_window.mov :
from the bottom to the top :
Original file
Encoded file
I changed the original file by settings "elst" atoms with the value of 0 for number entries (I used Visual Studio hexa editor)
Like Roman R. said, MediaFoundation mp4 source doesn't use the "edts/elst" atoms. But Blender and your video editing tools do. Also the "tmcd" track is ignored by mp4 source.
"edts/elst" :
Edits Atom ( 'edts' )
Edit lists can be used for hint tracks...
MPEG-4 File Source
The MPEG-4 file source silently ignores hint tracks.
So in fact, the encoding is good. I think there is no audio stream sync offset, comparing to the real audio/video data. For example, you can add "edts/elst" to the encoded file, to get the same result.
PS: on the encoded file, i added "edts/elst" for both audio/video tracks. I also increased size for trak atoms and moov atom. I confirm, Blender shows same wave form for both original and encoded file.
EDIT
I tried to understand relation between mvhd/tkhd/mdhd/elst atoms, in the 3 video samples. (Yes I know, i should read the spec. But i'm lazy...)
You can use a mp4 explorer tool to get atom's values, or use the mp4 parser from my H264Dxva2Decoder project :
H264Dxva2Decoder
Tuning_against_a_window.mov
elst (media time) from tkhd video : 20689
elst (media time) from tkhd audio : 1483
GREEN_SCREEN_ANIMALS__ALPACA.mp4
elst (media time) from tkhd video : 2002
elst (media time) from tkhd audio : 1024
GOPR6239_1.mov
elst (media time) from tkhd video : 0
elst (media time) from tkhd audio : 0
As you can see, with GOPR6239_1.mov, media time from elst is 0. That's why there is no video/audio sync problem with this file.
For Tuning_against_a_window.mov and GREEN_SCREEN_ANIMALS__ALPACA.mp4, i tried to calculate the video/audio offset.
I modified my project to take this into account :
EncodeWithSourceReaderSinkWriter
For now, i didn't find a generic calculation for all files.
I just find the video/audio offset needed to encode correctly both files.
For Tuning_against_a_window.mov, i begin encoding after (movie time - video/audio mdhd time).
For GREEN_SCREEN_ANIMALS__ALPACA.mp4, i begin encoding after video/audio elst media time.
It's OK, but I need to find the right unique calculation for all files.
So you have 2 options :
encode the file and add elst atom
encode the file using right offset calculation
it depends on your needs :
The first option permits you to keep the original file.But you have to add the elst atom
With the second option you have to read atom from the file before encoding, and the encoded file will loose few original frames
If you choose the first option, i will explain how I add the elst atom.
PS : i'm intersting by this question, because in my H264Dxva2Decoder project, the edts/elst atom is in my todo list.
I parse it, but i don't use it...
PS2 : this link sounds interesting :
Audio Priming - Handling Encoder Delay in AAC
I’m using NAudio and Lame Audio to Convert Wav to Mp3, I’m newbie too for this Audio Conversion code. Thanks to Mark I’m using his Audio File Inspector to get the details
Here is the details
Input - Wave Format details
Opening D:\Data\Test\NAudio\Wav\8777828760-e5749e4c563bf5411c954442085d1ce1#10.58.13.40.wav
DviAdpcm 8000Hz 2 channels 4 bits per sample
Extra Size: 2 Block Align: 512 Average Bytes Per Second: 8110
WaveFormat: DviAdpcm
Length: 788808 bytes: 00:01:37.2640000
Chunk: fact, length 420 D9 0B 00
Output Mp3
Opening D:\Data\Test\NAudio\Mp3\8777828760-e5749e4c563bf5411c954442085d1ce1#10.58.13.40.mp3
MP3 File WaveFormat: MpegLayer3 8000Hz 2 channels 0 bits per sample
Extra Size: 12 Block Align: 1 Average Bytes Per Second: 3000
ID: Mpeg Flags: PaddingIso Block Size: 216 Frames per Block: 1
Length: 3119616 bytes: 00:01:37.4880000
ID3v1 Tag: None
ID3v2 Tag: None
Version25,Layer3,8000Hz,JointStereo,24000bps, length 216
Version25,Layer3,8000Hz,JointStereo,24000bps, length 216
….
….
I’m Converting Wav to Mp3 ( voice recording files).
Question : I’m seeing some compromise in Mp3 Quality, My converted Mp3 is lower file size when compared to Wav, but my audio quality is little poor than Wav, Wonder if i can increase the quality of the Mp3 file ?
Something like increasing the Bitrate etc.
Code for Wav to Mp3 conversation using NAudio / Lame Audio
string filePath = #"D:\Data\Test\NAudio\Wav\11mb.wav";
string outputPath = #"D:\Data\Test\NAudio\Mp3\11mb.mp3";
using (WaveFileReader wavReader = new WaveFileReader(filePath))
using (WaveStream pcm = WaveFormatConversionStream.CreatePcmStream(wavReader))
using (LameMP3FileWriter fileWriter = new LameMP3FileWriter(outputPath, pcm.WaveFormat, LAMEPreset.VBR_90))
{
pcm.CopyTo(fileWriter);
}
This link has more details on my above question
http://mark-dot-net.blogspot.com/search/label/NAudio
MP3 is a heavily compressed codec, it will never get close to the original .Wav quality.
However, if you look at the original .Wav quality, you are starting from a very poor recording. When the Hz and bit depth are that low, there is all sorts of artifacts getting created as the wavform is very poorly represented digitally to start with.
anything under CD quality is going to have a LOT of problems being compressed because so much is missing to begin with.
Perfect for making "Boards of Canada" music though. :)
I have AM I/Q wave file that can be played on SDRSHARP. I want to make it play through portaudio at first without performing any kind of demodulation. before that i want to clear my concepts through these questions.
The sample file is seen as 22050 sample rate sample file with 32bit floating points sample in audacity software. Now these files are saved from USB dongle data sampled at 2.048 Msps by RTL2832u ADC using sdrsharp. So how should i read it when file sample rate is 22050? Should i consider it as live source with I/Q samples coming at rate of 22050 samples/sec, this sampling rate is important as it would be required for applying DSP technique like filtering etc?
I have made a small program using portaudio to redirect whatever it has input(microphone) to output (speakers) using callback, How can i read my any wave file using same call back whereas in my call back i have taken framesperbuffer to be 256 what value it should be for wave file saved at 22050 sample rate, there must be some relation between framesperbuffer and file sample rate?. i am using C++ (Visual Studio 2010).
I'm using WinAPI - Wave functions to create a recording program that records the microphone for X seconds. I've searched a bit over the net, and found out PCM data is too large, and it'll be a problem to send it through sockets...
How can I compress it to something smaller? Any simple / "cheap" way ?
I've also noticed, when I'm declaring the format using the Wave API functions, I'm using this code :
WAVEFORMATEX pFormat;
pFormat.wFormatTag= WAVE_FORMAT_PCM; // simple, uncompressed format
pFormat.nChannels=1; // 1=mono, 2=stereo
pFormat.nSamplesPerSec=sampleRate; // 44100
pFormat.nAvgBytesPerSec=sampleRate*2; // = nSamplesPerSec * n.Channels * wBitsPerSample/8
pFormat.nBlockAlign=2; // = n.Channels * wBitsPerSample/8
pFormat.wBitsPerSample=16; // 16 for high quality, 8 for telephone-grade
pFormat.cbSize=0;
As you can see, pFormat.wFormatTag= WAVE_FORMAT_PCM;
maybe I can insert instead of WAVE_FORMAT_PCM something else, so it'll be compressed right away?
I've checked MSDN for other values, though none of them works for me in my Visual Studio...
So what can I do?
Thanks!
The simplest way is to simply reduce your sample rate from 44100 to something more manageable like 22050, 16000, 11025, or even 8000. Most voice codecs don't go higher than 16000 hz anyway. And the older ones are optimized for 8khz.
The next step is to find a codec. There's a handful of codecs to use with the Windows Audio Compression Manager, but almost all of them date back to Windows 95 and sound terrible by modern standards after being decompressed.
You can always convert to WMA in real time using the Format SDK or with Media Foundation APIs. Or just go get an open source MP3 library like LAME.
For telephone quality speech you can change to 8 bits per sample and a sample rate of 8000. This will greatly reduce the amount of data.
GSM has good compression. You can convert a block of PCM data to GSM (or any other codec you have installed) using acmStreamConvert(). Refer to MSDN for more details:
Converting Data from One Format to Another