sox

问题 I'm new to libsox programming, and I want to reduce a chennel from a stereo audio which named 'a.wav' then generate a mono audio 'b.wav' with the following code: sox_format_t * in, * out; sox_effects_chain_t * chain; sox_effect_t * e; char * args[10]; sox_init(); in = sox_open_read("E:\\a.wav", NULL, NULL, NULL); out = sox_open_write("E:\\b.wav", &in->signal, NULL, NULL, NULL, NULL); out->signal.channels = 1; chain = sox_create_effects_chain(&in->encoding, &out->encoding); e = sox_create

问题 I am trying to use libsox to increase the channel of a wav file. I have read the related question: Reducing a channel from wav file using libsox I also read the libsox's exampl3.c. I take a deep copy of in->signal as interm_signal. But the resulting file is still half the duration of the original file. Here is my code: int main(int argc, char * argv[]) { sox_format_t *input = NULL; sox_format_t *output = NULL; sox_effects_chain_t *chain = NULL; sox_effect_t *e = NULL; char * args[10]; assert

问题 I'd like to use aubiopitch to continuously output the frequency of a signal coming from an input source. Since aubiopitch likes to have its input be a file, not a stream, I tried using process substitution: $ aubiopitch -i <(sox -q -d -t wav -) I'd expect this to output the frequency of the signal being read off of my default audio input device. Instead, I got this: ./sox WARN wav: Length in output .wav header will be wrong since can't seek to fix it AUBIO ERROR: source_apple_audio: Failed

问题 I am reading an audio stream via ffpmeg like this: ffmpeg -i http://icecast.radiovox.org:8000/live.ogg -f mp3 filename and want to pipe it to a sox command: rec filename rate 32k silence 1 0.1 3% 1 3.0 3%. Ultimately, what I am trying to achieve, is to record the audio from a live Icecast stream of a talk show. I only want recordings though of the individual's speaking. Everytime there is silence, I want to stop the recording and start a new one once they start speaking again. 回答1: What

问题 Is there any npn packages to convert a wav file to mp3 file in node js? I want to run this script in aws lambda function. I tried sox and sox-audio packages, but it is not supported in lambda. I googled ffmpeg, didn't find any convertion between wav to mp3. Does any one give a good convertion package in node js? console.log('New Lambda Call'); var async = require('async'); var aws = require('aws-sdk'); var fs = require('fs'); var uuid = require('node-uuid'); var SoxCommand = require('sox

问题 I have a text file looks like this text file ,and I want to use sox to trim the audio file based on the time in the text file, so I can have different audio clips from 0.0 to 6.16, 6.16 to 13.44, 13.44 to 17.54 etc.. I understand the basic script for sox trim is *$ sox audio.wav newaudio.wav trim starttime duration* But how can I get the duration from the text file and use sox to trim the audio? 回答1: Updated Answer I think the following code may be closer: #!/bin/bash index=0 while read this;

问题 The task is to remove silence by threshold from the start and end of audio recording. I use this sox port to iOS. https://github.com/shieldlock/SoX-iPhone-Lib/ I've found that command line sox tool makes my task by following command: sox in.wav out.wav silence 1 0.1 1% reverse silence 1 0.1 1% reverse (taken from here: http://digitalcardboard.com/blog/2009/08/25/the-sox-of-silence/) but I cannot to translate it in iOS lib format like this: sox_create_effect(sox_find_effect("silence")); args[0

问题 The task is to remove silence by threshold from the start and end of audio recording. I use this sox port to iOS. https://github.com/shieldlock/SoX-iPhone-Lib/ I've found that command line sox tool makes my task by following command: sox in.wav out.wav silence 1 0.1 1% reverse silence 1 0.1 1% reverse (taken from here: http://digitalcardboard.com/blog/2009/08/25/the-sox-of-silence/) but I cannot to translate it in iOS lib format like this: sox_create_effect(sox_find_effect("silence")); args[0

问题 C:\Program Files\sox-14-4-0>sox Sample.wav -n stat The above code gives below result Samples read: 26640 Length (seconds): 3.330000 Scaled by: 2147483647.0 Maximum amplitude: 0.515625 Minimum amplitude: -0.734375 Midline amplitude: -0.109375 Mean norm: 0.058691 Mean amplitude: 0.000122 RMS amplitude: 0.101146 Maximum delta: 0.550781 Minimum delta: 0.000000 Mean delta: 0.021387 RMS delta: 0.041831 Rough frequency: 526 Volume adjustment: 1.362 Now i need only Mean amplitude. How to do that? 回答1

问题 I want to exchange 16kHz pcm --> 48kHz wav using sox. however, pcm file isn't applied in sox. so, I just changed pcm to raw, and then sox -r 16000 -e signed -b 16 -c 1 test.raw -r 48000 out.wav Can I apply for pcm file not convert raw? 回答1: For the PCM file, since PCM's are headerless, you need to add '-t raw' as the first argument. sox -t raw -r 16000 -e signed -b 16 -c 1 test.raw -r 48000 out.wav Try that out. Also try the different Endian settings; -L; -B; -x though only use one at a time,