Inheritance diagram for com::avpkit::core::AudioResampler:

Collaboration diagram for com::avpkit::core::AudioResampler:

[legend]

Public Member Functions
virtual int	getOutputChannels ()
	number of channels in output audio. More...

virtual int	getOutputRate ()
	sample rate of output audio. More...

virtual int	getInputChannels ()
	number of channels expected in input audio. More...

virtual int	getInputRate ()
	sample rate expected in input audio. More...

virtual int	resample (IAudioSamples pOutputSamples, IAudioSamples pInputSamples, int32_t numSamples)
	Re-sample up to numSamples from inputSamples to outputSamples. More...

virtual IAudioSamples::Format	getOutputFormat ()
	Get the sample format we expect to resample to. More...

virtual IAudioSamples::Format	getInputFormat ()
	Get the sample format we expect to resample from. More...

virtual int32_t	getFilterLen ()
	Get the length of each filter in the resampler filter bank. More...

virtual int32_t	getLog2PhaseCount ()
	Get log2(number of entries in filter bank). More...

virtual bool	isLinear ()
	Are we linearly interpolating between filters? More...

virtual double	getCutoffFrequency ()
	What is the cuttoff frequency used? More...

virtual int32_t	getMinimumNumSamplesRequiredInOutputSamples (IAudioSamples *inSamples)
	Get the minimum number of samples that must be placeable in an output set of samples in order for a resample with this resampler to succeed. More...

virtual int32_t	getMinimumNumSamplesRequiredInOutputSamples (int32_t numSamples)
	Get the minimum number of samples that must be placeable in an output set of samples in order for a resample with this resampler to succeed. More...

Public Member Functions inherited from com::avpkit::ferry::RefCounted
virtual int32_t	acquire ()
	Internal Only. More...

virtual int32_t	release ()
	Internal Only. More...

virtual RefCounted *	copyReference ()
	Create a new Java object that refers to the same native object. More...

virtual int32_t	getCurrentRefCount ()
	Return the current reference count on this object. More...

void	setJavaAllocator (void *allocator)
	This method is public but not part of the standard API. More...

void *	getJavaAllocator ()
	This method is public but not part of the standard API. More...

Static Public Member Functions
static AudioResampler *	make (int32_t outputChannels, int32_t inputChannels, int32_t outputRate, int32_t inputRate)

static AudioResampler *	make (int32_t outputChannels, int32_t inputChannels, int32_t outputRate, int32_t inputRate, IAudioSamples::Format outputFmt, IAudioSamples::Format inputFmt)

static AudioResampler *	make (int32_t outputChannels, int32_t inputChannels, int32_t outputRate, int32_t inputRate, IAudioSamples::Format outputFmt, IAudioSamples::Format inputFmt, int32_t filterLen, int32_t log2PhaseCount, bool isLinear, double cutoff)

Static Public Member Functions inherited from com::avpkit::core::IAudioResampler
static IAudioResampler *	make (int32_t outputChannels, int32_t inputChannels, int32_t outputRate, int32_t inputRate)
	Create a new IAudioResampler object. More...

static IAudioResampler *	make (int32_t outputChannels, int32_t inputChannels, int32_t outputRate, int32_t inputRate, IAudioSamples::Format outputFmt, IAudioSamples::Format inputFmt)
	Create a new IAudioResampler object. More...

static IAudioResampler *	make (int32_t outputChannels, int32_t inputChannels, int32_t outputRate, int32_t inputRate, IAudioSamples::Format outputFmt, IAudioSamples::Format inputFmt, int32_t filterLen, int32_t log2PhaseCount, bool isLinear, double cutoffFrequency)
	Create a new IAudioResampler object. More...

Additional Inherited Members
Protected Member Functions inherited from com::avpkit::ferry::RefCounted
virtual void	destroy ()
	This method is called by RefCounted objects when their Ref Count reaches zero and they are about to be destroyed.

Protected Attributes inherited from com::avpkit::ferry::RefCounted
AtomicInteger *	mRefCount
	This is the internal reference count, represented as an AtomicInteger to make sure it is thread safe.

void *	mAllocator
	Not part of public API.

Detailed Description

Definition at line 29 of file AudioResampler.h.

Member Function Documentation

◆ getCutoffFrequency()

double com::avpkit::core::AudioResampler::getCutoffFrequency ( )

virtual

What is the cuttoff frequency used?

Returns: the cuttoff frequency

Implements com::avpkit::core::IAudioResampler.

Definition at line 472 of file AudioResampler.cpp.

   {
     return mCutoff;
   }  

◆ getFilterLen()

int32_t com::avpkit::core::AudioResampler::getFilterLen ( )

virtual

Get the length of each filter in the resampler filter bank.

Returns: the filter length

Implements com::avpkit::core::IAudioResampler.

Definition at line 454 of file AudioResampler.cpp.

   {
     return mFilterLen;
   }

◆ getInputChannels()

int com::avpkit::core::AudioResampler::getInputChannels ( )

virtual

number of channels expected in input audio.

Returns: Number of channels we'll expect in the input samples

Implements com::avpkit::core::IAudioResampler.

Definition at line 179 of file AudioResampler.cpp.

   {
     VS_ASSERT(swrContext, "no context");
     return mIChannels;
   }

◆ getInputFormat()

IAudioSamples::Format com::avpkit::core::AudioResampler::getInputFormat ( )

virtual

Get the sample format we expect to resample from.

Returns: the sample format for input.

Implements com::avpkit::core::IAudioResampler.

Definition at line 448 of file AudioResampler.cpp.

   {
     return mIFmt;
   }

◆ getInputRate()

int com::avpkit::core::AudioResampler::getInputRate ( )

virtual

sample rate expected in input audio.

Returns: Sample rate we'll expect in the input samples

Implements com::avpkit::core::IAudioResampler.

Definition at line 186 of file AudioResampler.cpp.

   {
     VS_ASSERT(swrContext, "no context");
     return mISampleRate;
   }

◆ getLog2PhaseCount()

int32_t com::avpkit::core::AudioResampler::getLog2PhaseCount ( )

virtual

Get log2(number of entries in filter bank).

Returns: log2(number of entries in filter bank).

Implements com::avpkit::core::IAudioResampler.

Definition at line 460 of file AudioResampler.cpp.

   {
     return mLog2PhaseCount;
   }

◆ getMinimumNumSamplesRequiredInOutputSamples() [1/2]

int32_t com::avpkit::core::AudioResampler::getMinimumNumSamplesRequiredInOutputSamples ( IAudioSamples * inSamples )

virtual

Get the minimum number of samples that must be placeable in an output set of samples in order for a resample with this resampler to succeed.

Parameters

inSamples The input samples that will be passed to resample.

Returns: The minimum number of samples, or < 0 on error.

Since: 3.2

Implements com::avpkit::core::IAudioResampler.

Definition at line 193 of file AudioResampler.cpp.

   {
     int32_t retval = -1;
     try {
       int32_t numSamples = 0;
       if (inSamples)
       {
         if (!inSamples->isComplete())
           throw std::invalid_argument("input samples are not complete");
         
         if (inSamples->getSampleRate() != mISampleRate)
           throw std::invalid_argument("unexpected input sample rate");
  
         if (inSamples->getChannels() != mIChannels)
           throw std::invalid_argument("unexpected # of input channels");
         
         if (inSamples->getFormat() != mIFmt)
           throw std::invalid_argument("unexpected sample format");
  
         numSamples = inSamples->getNumSamples();
       } else {
         numSamples = 0;
       }
       retval = getMinimumNumSamplesRequiredInOutputSamples(numSamples);
     }
     catch (std::invalid_argument & e)
     {
       VS_LOG_DEBUG("invalid argument: %s", e.what());
       retval = -1;
     }
     catch (std::exception & e)
     {
       VS_LOG_DEBUG("Unknown exception: %s", e.what());
     }
     return retval;
   }

References com::avpkit::core::IAudioSamples::getChannels(), com::avpkit::core::IAudioSamples::getFormat(), com::avpkit::core::IAudioSamples::getNumSamples(), com::avpkit::core::IAudioSamples::getSampleRate(), and com::avpkit::core::IAudioSamples::isComplete().

◆ getMinimumNumSamplesRequiredInOutputSamples() [2/2]

int32_t com::avpkit::core::AudioResampler::getMinimumNumSamplesRequiredInOutputSamples ( int32_t numSamples )

virtual

Get the minimum number of samples that must be placeable in an output set of samples in order for a resample with this resampler to succeed.

Parameters

numSamples The number of input samples.

Returns: The minimum number of samples, or < 0 on error.

Since: 3.2

Implements com::avpkit::core::IAudioResampler.

Definition at line 232 of file AudioResampler.cpp.

   {
     int32_t retval = -1;
  
     try
     {
       if (numSamples < 0)
         throw std::invalid_argument("numSamples < 0 not allowed");
       
       double conversionRatio = 1;
       {
         double top = mOSampleRate;
         VS_ASSERT(top, "should never be zero");
         double bot = mISampleRate;
         VS_ASSERT(bot, "should never be zero");
         conversionRatio = top/bot;
         VS_ASSERT(conversionRatio > 0, "the variables used should have been checked on construction");
       }
       if (conversionRatio <= 0)
         throw std::invalid_argument("programmer error");
  
       // FFMPEG's re-sample function doesn't let you specify the size of your
       // output buffer, but does use up all the space you might expect
       // plus 16-bytes as a lead-in/lead-out for it to seed the resampler.
       // Hence, the hard-coded 16 here.
       // NOTE: 16 might change IF the value of filters in the audio_resample
       // method in libavcodec/resample.c changes.
 #define VS_FFMPEG_AUDIO_RESAMPLER_LEADIN 16
       retval = 
           (int32_t)((numSamples * conversionRatio)+VS_FFMPEG_AUDIO_RESAMPLER_LEADIN+0.5);
     }
     catch (std::invalid_argument & e)
     {
       VS_LOG_DEBUG("invalid argument: %s", e.what());
       retval = -1;
     }
     catch (std::exception & e)
     {
       VS_LOG_DEBUG("Unknown exception: %s", e.what());
     }
  
     return retval;
   }

◆ getOutputChannels()

int com::avpkit::core::AudioResampler::getOutputChannels ( )

virtual

number of channels in output audio.

Returns: Number of channels we'll resample the output to.

Implements com::avpkit::core::IAudioResampler.

Definition at line 165 of file AudioResampler.cpp.

   {
     VS_ASSERT(swrContext, "no context");
     return mOChannels;
   }

◆ getOutputFormat()

IAudioSamples::Format com::avpkit::core::AudioResampler::getOutputFormat ( )

virtual

Get the sample format we expect to resample to.

Returns: the sample format for output.

Implements com::avpkit::core::IAudioResampler.

Definition at line 442 of file AudioResampler.cpp.

   {
     return mOFmt;
   }

◆ getOutputRate()

int com::avpkit::core::AudioResampler::getOutputRate ( )

virtual

sample rate of output audio.

Returns: Sample Rate we'll resample the output to.

Implements com::avpkit::core::IAudioResampler.

Definition at line 172 of file AudioResampler.cpp.

   {
     VS_ASSERT(swrContext, "no context");
     return mOSampleRate;
   }

◆ isLinear()

bool com::avpkit::core::AudioResampler::isLinear ( )

virtual

Are we linearly interpolating between filters?

Returns: true if interpolating, false if just choosing closest.

Implements com::avpkit::core::IAudioResampler.

Definition at line 466 of file AudioResampler.cpp.

   {
     return mIsLinear;
   }

◆ resample()

int com::avpkit::core::AudioResampler::resample	(	IAudioSamples *	outputSamples,
		IAudioSamples *	inputSamples,
		int32_t	numSamples
	)

virtual

Re-sample up to numSamples from inputSamples to outputSamples.

This function re-samples the audio in inputSamples to have the same number of channels, and the same sample rate, as this IAudioResampler was initialized with.

Caller is responsible for making sure that the IAudioSamples object passed in has enough space. This object requires space for an additional 16-samples * output-channels per IAudioSamples passed in.

If the

outputSamples

is not large enough to fit the required resamples, then the IAudioResampler will attempt to allocate a new output buffer and override the buffer set on

outputSamples

. To ensure you have enough space in your output buffer, use the getMinimumNumSamplesRequiredInOutputSamples(int) method.

Parameters

outputSamples	[out] The sample buffer we output to.
inputSamples	[in] The samples we're going to re-sample.
numSamples	[in] The number of samples from inputSamples to use. if 0, this defaults to inputSamples.getNumSamples().

Returns: Number of samples written to outputSamples, or <0 on error.

Implements com::avpkit::core::IAudioResampler.

Definition at line 278 of file AudioResampler.cpp.

   {
     int retval = -1;
     AudioSamples* outSamples = static_cast<AudioSamples*>(pOutSamples);
     AudioSamples* inSamples = static_cast<AudioSamples*>(pInSamples);
     unsigned int sampleSize=0;
  
     try {
       if (!outSamples)
         throw std::invalid_argument("no output samples");
       
       if (outSamples == inSamples)
         throw std::invalid_argument("resampling into the same IAudioSamples is not allowed");
  
       // null out the output samples.
       outSamples->setComplete(false, 0, mOSampleRate, mOChannels,
           mOFmt, Global::NO_PTS);
  
       if (inSamples)
       {
         if (!inSamples->isComplete())
           throw std::invalid_argument("input samples are not complete");
         
         if (inSamples->getSampleRate() != mISampleRate)
           throw std::invalid_argument("unexpected input sample rate");
  
         if (inSamples->getChannels() != mIChannels)
           throw std::invalid_argument("unexpected # of input channels");
         
         if (inSamples->getFormat() != mIFmt)
           throw std::invalid_argument("unexpected sample format");
  
         if (numSamples == 0)
           numSamples = inSamples->getNumSamples();
         else
           numSamples = FFMIN(numSamples, inSamples->getNumSamples());
         sampleSize = inSamples->getSampleBitDepth()/8;
       } else {
         numSamples = 0;
         sampleSize = IAudioSamples::findSampleBitDepth(mIFmt)/8;
       }
  
       int32_t neededSamples = getMinimumNumSamplesRequiredInOutputSamples(numSamples);
       int32_t bytesPerOutputSample = mOChannels*IAudioSamples::findSampleBitDepth(mOFmt)/8;
       int32_t neededBytes = neededSamples * bytesPerOutputSample; //av_samples_get_buffer_size(NULL, mOChannels, numSamples, (enum AVSampleFormat)mOFmt, 0);// 
       // This causes a buffer resize to occur if needed
       if (outSamples->ensureCapacity(neededBytes) < 0)
         throw std::runtime_error("attempted to resize output buffer but failed");
       
       int32_t outBufSize = outSamples->getMaxBufferSize();
       int32_t gap = (neededSamples*bytesPerOutputSample)-outBufSize;
       
       if (gap > 0) {
 //        VS_LOG_ERROR("maxBufferSize: %d; neededSampleRoom: %d; sampleSize: %d; numSamples: %d; conversionRatio: %f;",
 //            (int32_t)outSamples->getMaxBufferSize(),
 //            neededSampleRoom,
 //            sampleSize,
 //            numSamples,
 //            conversionRatio);
         (void) sampleSize; // to avoid a -Werror error
         throw std::invalid_argument("not enough room in output buffer");
       }
       short * inBuf = inSamples ? inSamples->getRawSamples(0) : 0;
  
       short *outBuf = outSamples->getRawSamples(0);
       if (!outBuf)
         throw std::invalid_argument("could not get output bytes");
  
       VS_ASSERT(swrContext, "Should have been set at initialization");
       if (!swrContext)
         throw std::invalid_argument("programmer error");
  
       // Now we should be far enough along that we can safely try a resample.
       retval = swr_convert(swrContext, (uint8_t**)&outBuf, swr_get_out_samples(swrContext, numSamples), (const uint8_t**)&inBuf, numSamples);
  
 #if 0
       if (retval >0){
         char string[2048*16+1];
         unsigned int i=0;
         for (i = 0; i < sizeof(string)-1; i++)
           string[i] = 'X';
  
         bool allZero = true;
  
         for (i=0; i< FFMIN(numSamples, 2000);i++)
         {
           snprintf(string+(5*i), sizeof(string)-5*i, "%04hX.", inBuf[i]);
           if (inBuf[i] != 0)
             allZero = false;
         }
         VS_LOG_DEBUG("Input Buffer (%d): %s", numSamples, string);
  
         for (i=0; i< FFMIN((unsigned int)retval, 2000);i++)
         {
           snprintf(string+(9*i), sizeof(string)-9*i, "%04hX%04hX.",
               outBuf[2*i], outBuf[2*i+1]);
           if (outBuf[2*i] != 0 || outBuf[2*i+1] != 0)
             allZero = false;
         }
         VS_LOG_DEBUG("Output Buffer (%d): %s", retval, string);
         if (!allZero)
           VS_LOG_DEBUG("Got an audio buffer with content");
       }
 #endif // 0
  
       if (retval >= 0)
       {
         // copy the Pts
         int64_t pts = Global::NO_PTS;
         if (inSamples)
         {
           pts = inSamples->getPts();
           mNextPts = pts + IAudioSamples::samplesToDefaultPts(retval, mOSampleRate);
         }
         else
         {
           pts = mNextPts;
         }
         if (pts != Global::NO_PTS)
           pts += mPtsOffset;
  
         outSamples->setComplete(true, retval,
             mOSampleRate, mOChannels,
             mOFmt,
             pts);
         int expectedSamples = 0;
         if (inSamples)
         {
           double top = mOSampleRate;
           double bottom = mISampleRate;
           double sampleOnlyConverstionRatio = top / bottom;
           expectedSamples = (int)(numSamples * sampleOnlyConverstionRatio);
         }
         else
         {
           VS_LOG_TRACE("Got null samples; outputted all cached and set pts offset from %lld to 0",
               mPtsOffset);
           expectedSamples = retval;
           // and reset the offset
           mPtsOffset = 0;
         }
  
         if (retval != expectedSamples)
         {
           // we got a different number of samples than expected; we need to update
           // our pts offset
           int sampleDelta = retval - expectedSamples;
           int64_t ptsDelta = IAudioSamples::samplesToDefaultPts(sampleDelta, mOSampleRate);
           mPtsOffset += ptsDelta;
         }
       }
     }
     catch (std::invalid_argument & e)
     {
       VS_LOG_DEBUG("invalid argument: %s", e.what());
       retval = -1;
     }
  
     return retval;
   }

The documentation for this class was generated from the following files:

csrc/com/avpkit/core/AudioResampler.h
csrc/com/avpkit/core/AudioResampler.cpp

Public Member Functions

Static Public Member Functions

Additional Inherited Members

Detailed Description

Member Function Documentation

◆ getCutoffFrequency()

◆ getFilterLen()

◆ getInputChannels()

◆ getInputFormat()

◆ getInputRate()

◆ getLog2PhaseCount()

◆ getMinimumNumSamplesRequiredInOutputSamples() [1/2]

◆ getMinimumNumSamplesRequiredInOutputSamples() [2/2]

◆ getOutputChannels()

◆ getOutputFormat()

◆ getOutputRate()

◆ isLinear()

◆ resample()