[Flac-dev] Getting subframe type=verbatim on 16 bit files

[James Smith]

Here's how I set up the data for processing:

    // For moving data into 32 bit shape
    uint8_t            *buffer8    = NULL;
    uint16_t        *buffer16    = NULL;
    uint32_t        *buffer32    = NULL;
    unsigned        sample32;
    unsigned        sample, channel;
    uint32_t        bitsPerSample = this->get_bits_per_sample();
    
    numFrames = inData.GetSize();
    numChannels = this->get_channels();
        
    // How big is our sample that we want to give to FLACC?
    // bitsPerSample is 8,16,24,32
    // So 8 = no change for numFrames
    //    16 = half it
    //    24,32 = 1/4 the needs..
    if (bitsPerSample == 16)
        numFrames = numFrames / 2;
    else if (bitsPerSample > 16)
        numFrames = numFrames / 4;
    // if stereo then spread data over both channels and not just one!
    if (numChannels == 2)  numFrames = numFrames / 2;

    // buffer for FLAC use
    flacBuffer = (int32_t **)calloc(numChannels, sizeof(int32_t *));
    
    if (flacBuffer == NULL)
        return false;
        
    // channel buffer to zero 1 or 2 (stereo)
    for(channel = 0; channel < numChannels; ++channel)
    {
        flacBuffer[channel] = NULL;
    }
        
    // set up buffers for channels
    for(channel = 0; channel < numChannels; ++channel)
    {
        flacBuffer[channel] = (int32_t*)calloc(numFrames, sizeof(int32_t));
        if (flacBuffer[channel] == NULL)
                    return false;
    }

   // uint32_t tempjunk;
    
    // Need to get our pcm Data into 32 bit shape and with with channels...
    // See how sample32 is the index into the 32 bit array
    // while the sample index can handle things when stereo (i.e. 2
channels) occurs.
    
    switch(bitsPerSample)
    {
            
        case SIXTEEN_BITS:
            buffer16 = (uint16_t*)inData.GetData(); // so we index thru out
data in 16 byte chunks
            for(sample = sample32 = 0; sample32 < numFrames; sample32++)
            {
                for(channel = 0; channel < numChannels; channel++, sample++)
                {
                    flacBuffer[channel][sample32] =
(uint32_t)(uint16_t)buffer16[sample];
                }
            }
            break;
        ..........
        
            }
     
     set_total_samples_estimate (numFrames);  // note we are doing this
before the init function is called!
     return true;
}

On 9/6/06 8:28 PM, "Josh Coalson" <xflac at yahoo.com> wrote:

> looks fine, I would suspect how the PCM sample are formatted
> and sent to process(), could you show that part of the code?
> 
> Josh
> 
> 
> --- James Smith <jsmith at landmarkdigital.com> wrote:
> 
>> 
>> I'm using libFLACC++ and libFLAC and I think that I'm using the calls
>> in the
>> typical order (see code below).  But every monoe or stereo file that
>> I send
>> thru I get files that are the same sze as the orginal wave files.
>> 
>> Doing a flac -a on the flac files I see that I get:
>> 
>> frame=9 blocksize=4608  sample_rate=8000        channels=1
>> channel_assignment=INDEPENDENT
>>         subframe=0      wasted_bits=0   type=VERBATIM
>>  .......
>> 
>> Any idea why/ where I have goofed?
>> 
>> Thanks,
>> 
>> James
>> 
>> 
>> Code snippet:
>> ===================================================
>>     FlacEncoder     flacCompressor;
>>     bool            setValue = false;
>>     
>>     // set up regular parameters
>>     setValue = flacCompressor.set_channels (numChannels);
>>     setValue = flacCompressor.set_bits_per_sample (bitsPerSample);
>>     setValue = flacCompressor.set_sample_rate (sampleRate);
>>     setValue = flacCompressor.set_blocksize(4608);
>>     setValue = flacCompressor.set_qlp_coeff_precision (0);
>>     setValue = flacCompressor.set_min_residual_partition_order (3);
>>     setValue = flacCompressor.set_max_residual_partition_order (3);
>>     setValue = flacCompressor.set_max_lpc_order (8);
>>     setValue = flacCompressor.set_rice_parameter_search_dist(0);
>>     setValue = flacCompressor.set_do_exhaustive_model_search(false);
>>     setValue = flacCompressor.set_do_escape_coding(false);
>>     
>>     
>>     if (numChannels > 1)
>>     {
>>         setValue = flacCompressor.set_do_mid_side_stereo(true);
>>         setValue = flacCompressor.set_loose_mid_side_stereo(true);
>>     }    
>>     
>>     if (!flacCompressor.SetupInboundBuffer(pcmData))   return false;
>> 
>>     
>>     FLAC__StreamMetadata            padding;
>>     FLAC__StreamMetadata            *metadata [1];
>> 
>>     // stuff some pading for metadata
>>     padding.type        = FLAC__METADATA_TYPE_PADDING;
>>     padding.is_last        = 0;
>>     padding.length        = 40;
>>     metadata[0]            = &padding;
>>     setValue = flacCompressor.set_metadata(metadata, 1);
>>     
>>     FlacEncoder::Stream::State flacState = flacCompressor.init();
>>     
>>     /*
>>     // verify that the values are correct...they are fine, why still
>> verbatim?
>>     int i;
>>     i = flacCompressor.get_channels ();
>>     i = flacCompressor.get_bits_per_sample ();
>>     i = flacCompressor.get_sample_rate ();
>>     i = flacCompressor.get_blocksize();
>>     i = flacCompressor.get_qlp_coeff_precision ();  // default=0 so
>> encoding
>> software picks the best
>>     i = flacCompressor.get_min_residual_partition_order ();
>>     i = flacCompressor.get_max_residual_partition_order ();     //
>> Had tried
>> 0,6  now try 3,3
>>     i = flacCompressor.get_max_lpc_order ();
>>     setValue = flacCompressor.get_do_mid_side_stereo();
>>     setValue = flacCompressor.get_loose_mid_side_stereo();
>>     */
>>     
>>     // is this state okay?
>>     if (flacState  != FLAC__STREAM_ENCODER_OK)
>>         return false; // bad settings for doing this compression so
>> quit...
>>     
>>     // shoot the final data buffer to the encoder....
>>     // we would have called process_interleaved if we had not created
>> an
>> array per channel above...
>>     if (!flacCompressor.DoProcess())
>>     {
>>         flacState = flacCompressor.get_state();
>>         return false;
>>     }
>>     
>>     flacState = flacCompressor.get_state();
>>     flacCompressor.finish();
>>     
>>      flacCompressor.Release();
>>     
>>     return true;
>> }
> 
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