IAudioSamples.cpp

/*******************************************************************************
 * Copyright (c) 2024, 2026, Olivier Ayache.  All rights reserved.
 *
 * This file is part of AVPKit.
 *
 * AVPKit is free software: you can redistribute it and/or modify
 * it under the terms of the GNU Lesser General Public License as published by
 * the Free Software Foundation, either version 3 of the License, or
 * (at your option) any later version.
 *
 * AVPKit is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public License
 * along with AVPKit.  If not, see <http://www.gnu.org/licenses/>.
 *******************************************************************************/
 
#include "IAudioSamples.h"
#include "Global.h"
#include "AudioSamples.h"
 
#include "FfmpegIncludes.h"
 
namespace com { namespace avpkit { namespace core
{
 
  IAudioSamples :: IAudioSamples()
  {
  }
 
  IAudioSamples :: ~IAudioSamples()
  {
  }
  
  int32_t
  IAudioSamples :: findSampleBitDepth(Format format)
  {
    int bits = av_get_bytes_per_sample((enum AVSampleFormat) format)<<3;
    return bits;
  }
 
  IAudioSamples*
  IAudioSamples :: make(int32_t numSamples, int32_t numChannels)
  {
    Global::init();
    return AudioSamples::make(numSamples, numChannels);
  }
 
  IAudioSamples*
  IAudioSamples :: make(int32_t numSamples, int32_t numChannels,
      IAudioSamples::Format format)
  {
    Global::init();
    return AudioSamples::make(numSamples, numChannels, format);
  }
  
 
  IAudioSamples*
  IAudioSamples :: make(com::avpkit::ferry::IBuffer* buffer, int channels,
      IAudioSamples::Format format)
  {
    Global::init();
    return AudioSamples::make(buffer, channels, format);
  }
  
  int64_t
  IAudioSamples :: samplesToDefaultPts(int64_t samples, int sampleRate)
  {
    // Note: These need to always round up!  a "partial sample" actually must
    // be whole (and similar with time stamps).
    int64_t retval = Global::NO_PTS;
    Global::init();
    if (sampleRate > 0)
    {
      int64_t num = samples * Global::DEFAULT_PTS_PER_SECOND;
      int64_t den = sampleRate;
      long double calc = ((long double)num)/((long double)den);
      retval = (int64_t)(calc);
    }
    return retval;
  }
 
  int64_t
  IAudioSamples :: defaultPtsToSamples(int64_t duration, int sampleRate)
  {
    int64_t retval = 0;
    Global::init();
    if (duration != Global::NO_PTS)
    {
      int64_t num = duration * sampleRate;
      int64_t den = Global::DEFAULT_PTS_PER_SECOND;
      long double calc = ((long double)num)/((long double)den);
      retval = (int64_t)(calc);
    }
    return retval;
  }
 
 
}}}