EncodedAudioStream, WAVEncoder and setting a sampling rate

[hash6iron]

Hi @pschatzmann.

I was using EncodedAudioStream to write samples directly to WAV file and this worked fine, generating WAV files at 22.050KHz.

A few days ago I refactor my project and now I´m using SD_MMC mode to SD access in order to use FLAC playing too. I noticed that now EncodedAudioStream not transfer sampling rate information to WAVEncoder and always WAV files (that I need to record at 22.05KHz) are recording to 44.1KHz (the default sampling of EncodedAudioStream constructor)

Part of the skecth that worked fine was so below

    EncodedAudioStream encoderOutWAV("/WAV/test.wav", new WAVEncoder());
    AudioInfo wavencodercfg(22050, 2, 16);
    encoderOutWAV.begin(wavencodercfg);
    ...
    site_t bytes_written

    ...
    // This while read samples and write to streams.
    while(track_size<total_size)
    {
         ...
         kitstream.write(buffer,bytes_tobe_written);
         encoderOutWAV.write(buffer,bytes_tobe_written);
   }

Thanks in advance.

[pschatzmann]

I committed a correction!

Setting the AudioInfo with setAudioInfo() was working as expected, only when calling begin(AudioInfo), the encoder was not set up correctly. This correction should correct the begin(AudioInfo).

[hash6iron]

Ok. I'm going to try again.

[hash6iron]

Hi @pschatzmann.

I ensured that EncodedAudio.h was updated. Now, begin is as below.

  /// Starts the processing - sets the status to active
  virtual bool begin(AudioInfo newInfo) override {
    setAudioInfo(newInfo);
    return begin();
  }

I try again but not was success. This apply in my code at time to set the encoder. The file recorded at 44.1KHz. On the other hand a I read the setting after applied and, encoderOutWAV return me, 44100Hz (see below)

my custom log.

        // Verificamos la configuración final
        logln("Sampling rate changed for TAP file: " + String(SAMPLING_RATE)  + "Hz");
        logln("WAV encoder - Out to WAV: " + String(encoderOutWAV.audioInfo().sample_rate) + 
              "Hz, Bits: " + String(encoderOutWAV.audioInfo().bits_per_sample) + 
              ", Channels: " + String(encoderOutWAV.audioInfo().channels));  

the result

Sampling rate for TAP playing: 22050Hz
WAV encoder - Out to WAV: 44100Hz, Bits: 16, Channels: 2

in the sketch I coded this.

        SAMPLING_RATE = 22050

        EncodedAudioStream encoderOutWAV("/WAV/test.wav", new WAVEncoder());
        AudioInfo wavencodercfg(SAMPLING_RATE, 2, 16);
        // Initializing the encoder.
        encoderOutWAV.begin(wavencodercfg);

Any idea?

[pschatzmann]

I tried to call audioInfo on the WAVEncoder but I was getting an compile error, which I corrected as well.

However, I can't reproduce your issue:

#include "AudioTools.h"
#include "SD.h"

WAVEncoder wav;
File file;
EncodedAudioStream enc(&file, &wav);
AudioInfo info(22000, 1, 16);

void setup() {
  Serial.begin(115200);
  delay(1000);
  Serial.println("starting...");
  AudioToolsLogger.begin(Serial, AudioToolsLogLevel::Info);

  enc.begin(info);
  wav.audioInfo().logInfo("WAVEncoder");
  enc.audioInfo().logInfo("EncodedAudioStream");

  assert(wav.audioInfo() == info);
  assert(enc.audioInfo() == info);
}

void loop() {}

gives

10:17:56.382 -> starting...
10:17:56.382 -> [I] AudioTypes.h : 126 - in: sample_rate: 22000 / channels: 1 / bits_per_sample: 16
10:17:56.382 -> [I] AudioTypes.h : 126 - out: sample_rate: 44100 / channels: 2 / bits_per_sample: 16
10:17:56.416 -> [I] CodecWAV.h : 577 - sample_rate: 22000
10:17:56.416 -> [I] CodecWAV.h : 578 - channels: 1
10:17:56.416 -> [I] CodecWAV.h : 588 - is_streamed! because length is 2147418112
10:17:56.416 -> [I] AudioEncoded.h : 75 - virtual void audio_tools::EncodedAudioOutput::addNotifyAudioChange(audio_tools::AudioInfoSupport&)
10:17:56.416 -> [I] CodecWAV.h : 577 - sample_rate: 22000
10:17:56.416 -> [I] CodecWAV.h : 578 - channels: 1
10:17:56.446 -> [I] CodecWAV.h : 588 - is_streamed! because length is 4294967295
10:17:56.446 -> [I] AudioTypes.h : 126 - WAVEncoder sample_rate: 22000 / channels: 1 / bits_per_sample: 16
10:17:56.446 -> [I] AudioTypes.h : 126 - EncodedAudioStream sample_rate: 22000 / channels: 1 / bits_per_sample: 16

[hash6iron]

I don´t know what´s happening but I follow your steps in the configuration phase and I don't achieve that WAV gets custom sampling rate. The only different is that I'm using SD_MMC to open the file.

[pschatzmann]

That does not make any difference.
Did you try my test sketch ?
Did you double check the Info Logs ?

[hash6iron]

Could you pass me the commit of this correction please?
Thanks.

[pschatzmann]

770f590
I renamed the conflicting audioInfo to wav_info

[hash6iron]

No, I trying to use in my project, later I playing the WAV file with ESP32 and Windows audio player with the same result, accelerated audio and WAv file properties says me that WAV audio is a 44100Hz, stereo at 16 bits.

I'm going to test with your sketch only in my AudioKit.

[hash6iron]

Hi.

it seems that VSCode not updated correctly AudioTools. After full clean, recloning and compiled again all works fine.
Thanks so much, Phil.

[hash6iron]

@pschatzmann

About AudioSourceIdxSDMMC. Is possible to implement a function to forze regeneration of the index file without to have to remove index files? I write a funcion using your code is as below, included in SDIndex.h

 void reindex(const char *startDir, const char *extension) {
    TRACED();
    if (startDir == nullptr || strlen(startDir) == 0) {
      LOGE("Invalid directory path");
      return;
    }
    
    // Update internal state
    this->start_dir = startDir;
    this->ext = extension;
    
    // Update paths for the new directory
    idx_path = filePathString(startDir, "idx.txt");
    idx_defpath = filePathString(startDir, "idx-def.txt");
    
    // Delete existing index files if they exist
    if (p_sd->exists(idx_path.c_str())) {
      LOGI("Removing existing index file");
      p_sd->remove(idx_path.c_str());
    }
    if (p_sd->exists(idx_defpath.c_str())) {
      LOGI("Removing existing index definition file");
      p_sd->remove(idx_defpath.c_str());
    }
    
    // Begin indexing process
    begin(startDir, extension, "*", true);
    
    // Create new key with provided parameters
    String keyNew = String(startDir) + "|" + extension + "|*";
    
    // Create new index file
    FileT idxfile = p_sd->open(idx_path.c_str(), FILE_WRITE);
    if (!idxfile) {
      LOGE("Failed to open index file for writing");
      return;
    }
    
    LOGW("Creating new index file");
    listDir(idxfile, startDir);
    LOGI("Indexing completed");
    idxfile.close();
    
    // Create new index definition file
    saveIndexDef(keyNew);
    
    // Reset max_idx since we've recreated the index
    max_idx = -1;
  }

[pschatzmann]

The index file is regenerated by calling begin() when the setup_index variable is set to true.
The can be done

• in the constructor
• (new): by calling setCreateIndex(true)

[hash6iron]

@pschatzmann

After update the FLAC playing is bad again. I achieved to play very fluid without breaks and I see that again the FLAC playing is not fine.
Any idea?

[hash6iron]

@pschatzmann

After update the FLAC playing is bad again. I achieved to play very fluid without breaks and I see that again the FLAC playing is not fine. Any idea?

Hi again.

I discovered the problem, I was writing inside the loop a value in kitStream.setVolume to react at volumen change. I corrected the error. Sorry. All is ok.