pcm Questions

4

Solved

I can convert wav file to pcm ffmpeg -i file.wav -f s16le -acodec pcm_s16le file.pcm How can I revert this operation?
Casimiracasimire asked 16/8, 2012 at 11:37

1

Solved

I want to implement SIP calls in my application, and first problem, that I need to solve, is converting audio from compressed AAC format with ADTS header to linear PCM. My input data is an NSArray...
Wagoner asked 22/3, 2017 at 17:55

1

I want to record input from the microphone, attach a reverb effect, and persist the result to a file. My use-case is an app that lets you sing a song and select different preset reverb options afte...
Meyers asked 26/1, 2017 at 21:50

4

I know that there are a lot of resources online explaining how to deinterleave PCM data. In the course of my current project I have looked at most of them...but I have no background in audio proces...
Provender asked 20/8, 2015 at 20:53

2

Can someone explain what is the difference between I2S interface and PCM interface. Does I2S interface only supports i2s format audio data and not PCM format audio data ?
Donnie asked 3/9, 2015 at 5:29

1

In JavaScript, I need to convert two bytes into a 16 bit integer, so that I can convert a stream of audio data into an array of signed PCM values. Most answers online for converting bytes to 16 bi...
Dewie asked 11/7, 2016 at 2:41

1

I wrote a program in c++ to generate a .wav file for an 800Hz sine wave (1 channel, 8-bit, 16000Hz sampling, 32000 samples so 2 seconds long), but when I play it or examine its spectrogram in Audac...
Kansas asked 23/4, 2016 at 18:8

3

You can use MediaRecorder to record a stream directly to AAC but there doesn't seem to be a way to encode an existing PCM/WAV file to AAC. The ability to encode to AAC exists natively in Android an...
Mythopoeic asked 15/12, 2011 at 21:4

4

Solved

I've been playing with this now for sometime, I cant work out what I am meant to be doing here. I am reading in PCM audio data into an audioData array: recorder.read(audioData,0,bufferSize); //r...
Shawnee asked 4/10, 2011 at 13:53

4

Solved

I have pcm audio stored in a byte array. It is 16 bits per sample. I want to make it 8 bit per sample audio. Can anyone suggest a good algorithm to do that? I haven't mentioned the bitrate becau...
Lanugo asked 19/4, 2011 at 13:52

1

Solved

The remote device is sending live raw PCM audio(no header included) in UDP packets and I need to implement a program in java to receive these packets and play them on the PC live. As I know that ra...
Deejay asked 30/9, 2015 at 18:53

2

I am trying to convert a raw file (header-less) to wav It is in ADPCM 16 rate 8000 I have tried three console commands sox -e ima-adpcm -r 8000 input.raw output.wav sox -e ima-adpcm -r 8000 -b 1...
Koo asked 25/7, 2014 at 8:45

3

Solved

When I play a file with the following code: private void PlayAudioFileViaAudioTrack(int ResId) throws IOException { int intSize = android.media.AudioTrack.getMinBufferSize(11025, AudioFormat.CH...
Millar asked 10/9, 2011 at 15:45

2

My android application needs to convert PCM(22khz) to AMR , but the API AmrInputStream only supports with pcm of 8khz. How can i downsample the pcm from 22 khz to 8 khz?
Cozmo asked 18/2, 2013 at 4:25

3

Solved

I've been working on a audio-recognize demo for some time, and the api needs me to pass an .wav file with sample rate of 8000 or 16000, so I have to downsample it. I have tried 2 algorithms as foll...
Sadi asked 4/8, 2015 at 19:54

6

Solved

I'm trying to write a program to display PCM data. I've been very frustrated trying to find a library with the right level of abstraction, but I've found the python wave library and have been using...
Erase asked 9/2, 2010 at 5:1

2

I have a PCM sound file with sample rate 16000, sample bit 16 and channel 1. I want to play it , but there's no software can do. I also tried ffplay like : ffplay -ar 16000 -ac 1 snake.raw But s...
Denominationalism asked 1/12, 2013 at 17:0

1

A clear explanation seems to be missing in the PulseAudio documentation, and I cannot find any simple examples.
Checklist asked 30/4, 2015 at 20:58

3

Solved

I have a wave file in 16bit PCM form. I've got the raw data in a byte[] and a method for extracting samples, and I need them in float format, i.e. a float[] to do a Fourier Transform. Here's my cod...
Bolus asked 7/1, 2011 at 20:58

1

Solved

I'm debugging an audio codec and need to test if the data is coming out of a codec properly. I have my code dumping the raw pcm stream into a file and now I want to try to play it in VLC or another...
Tintype asked 27/1, 2015 at 15:57

3

I was confused with resampling result in new ffmpeg. I decode an AAC audio into PCM, the ffmpeg show audio information as: Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 122 kb/s In new ffmpeg...
Crowson asked 2/4, 2014 at 20:37

3

Solved

I would like to extract a channel audio from the an LPCM raw file ie extract left and right channel of a stereo LPCM file. The LPCM is 16 bit depth,interleaved, 2 channels,litle endian. From what I...
Dispense asked 7/1, 2011 at 22:46

0

I'm developing an android application. What I need to do is reading a per recorded audio file. Audio file has been recorded and saved as .pcm file. I use this code to read the .pcm file. public do...
Bridgettebridgewater asked 2/8, 2014 at 17:29

1

Solved

I understand that PCM data is stored as [left][right][left][right].... Am trying to convert a stereo PCM to mono Vorbis (*.ogg) which I understand is achievable by halving the left and the right ch...
Coma asked 17/6, 2014 at 13:45

2

Solved

There isn't too much info about apply this codec when we need to streaming audio. Without apply the codec, my code work like a charm establishing a communication between 2 devices but I need encode...
Padang asked 24/4, 2014 at 15:53

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