Not able to save .wav

[Nowhk]

I'm trying to save a buffer of samples on a wav file using your fancy library. Unfortunately, it outputs weird samples on the waveform, not sure why.

Here's the code:

bool Wave::saveWave(std::string &pathFile) {
    drwav wav;
    drwav_data_format format;

    format.container = drwav_container_riff;
    format.format = DR_WAVE_FORMAT_PCM;

    format.channels = mNumChannels;
    format.sampleRate = mSampleRateWave;
    format.bitsPerSample = 16;

    if (drwav_init_file_write(&wav, pathFile.c_str(), &format, NULL)) {
        std::vector<float> samples;
        for (int i = 0; i < mNumSamples; i++) {
            samples.push_back(mBuffer[0][i]);

            if (mNumChannels == 2) {
                samples.push_back(mBuffer[1][i]);
            }
        }

        drwav_uint64 framesWritten = drwav_write_pcm_frames(&wav, mNumSamples, samples.data());

        if (framesWritten == mNumSamples) {
            return true;
        }
    }

    return false;
}

It seems it write the correct number of samples, but when I open the wav, it sounds crap. Where am I wrong? Or is it a bug?

Note: It seems it also ignore the num channels and the bit depth I set. It always save a 32bit/2 channels wave file, also if I set 1 channel and 16 bit.

[mackron]

It's because you're writing 32-bit floats:

std::vector<float> samples;

but are specifying an output format of 16-bit integers:

format.format = DR_WAVE_FORMAT_PCM;
format.bitsPerSample = 16;

You need to either change your output format to 32-bit floats:

format.format = DR_WAVE_FORMAT_IEEE_FLOAT;
format.bitsPerSample = 32;

or convert your samples to 16 bit integers:

std::vector<drwav_int16> samples;
samples.push_back(f32_to_s16(mBuffer[0][i]));
samples.push_back(f32_to_s16(mBuffer[1][i]));

...

drwav_int16 f32_to_s16(float x)
{
    x = ((x < -1) ? -1 : ((x > 1) ? 1 : x));    /* clip */
    return (drwav_int16)(x * 32767);
}

[Nowhk]

You are right, now it sounds ok! Thanks. But there are still one problem: drwav_write_pcm_frames return 173436 as written samples, but when I open an audio editor, the length is 173050. It seems it cut last samples, any reason?

[Nowhk]

Also: if I store with:

format.format = DR_WAVE_FORMAT_IEEE_FLOAT;
format.bitsPerSample = 32;

I'm not able later to open the sample with drwav_open_file_and_read_pcm_frames_f32 :O

[mackron]

It's probably because you aren't uninitializing the drwav object. As a result, the internal file isn't being closed with fclose() which means the last bit is probably never flushed to disk. Do this:

if (framesWritten == mNumSamples) {
    drwav_uninit(&wav); /* <-- Add this line. */
    return true;
}

I expect that should fix your drwav_open_file_and_read_pcm_frames_f32().

[Nowhk]

Perfect, it was that!!!! :) Thanks dude, you make my day!

Anyway, experimenting this, I've maybe found a "real" but ahah.

Id I do this:

unsigned int channels;
unsigned int sampleRateWave;
drwav_uint64 numTotalSamples;
float *pSampleData = drwav_open_file_and_read_pcm_frames_f32(mPathFile.c_str(), &channels, &sampleRateWave, &numTotalSamples, NULL);

if (pSampleData) {
    // ok, process samples
}

drwav_free(pSampleData, NULL);

When I load correctly a sample, it works perfect. If, for example, I create a txt file, and I rename it .wav, and I try to open, obviously it doesn't process and skip if (pSampleData), going directly to drwav_free(pSampleData, NULL);

But at that time, if I go to file system and I try to rename/cancel the fake sample already loaded, it can't. It seems it "blocked" by the drwav process. i.e. its doesn't free the file.

Is it normal? Bug? Or am I freeing wrongly?

[mackron]

That's definitely not intended. Good catch! I've pushed a fix to the dev branch. Are you able to try that? https://github.com/mackron/dr_libs/blob/dev/dr_wav.h

Commit: https://github.com/mackron/dr_libs/commit/aab128563678fd622ab063c13a596cad57718dec

[Nowhk]

Perfect, it seems fixed. Thanks. I wait the official release to be downloaded so ;)

Bonus tip: on documentation, you know forgot to specify to set NULL in most of the function. For example, this doesn't work as "copy/paste":

If you just want to quickly open and read the audio data in a single operation you can do something like this:

    ```c
    unsigned int channels;
    unsigned int sampleRate;
    drwav_uint64 totalPCMFrameCount;
    float* pSampleData = drwav_open_file_and_read_pcm_frames_f32("my_song.wav", &channels, &sampleRate, &totalPCMFrameCount);
    if (pSampleData == NULL) {
        // Error opening and reading WAV file.
    }

because lacks , NULL in the end of drwav_open_file_and_read_pcm_frames_f32 ;)

[mackron]

Thanks for testing that. I've fixed those documentation errors and have released version 0.12.2 to the master branch.

I'll go ahead and close this one now. Feel free to open a new issue if you find anything new. Thanks for the report!