Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp - WebKit - Git at Google

 /*
  * Copyright (C) 2017 Apple Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  * 1. Redistributions of source code must retain the above copyright
  *    notice, this list of conditions and the following disclaimer.
  * 2. Redistributions in binary form must reproduce the above copyright
  *    notice, this list of conditions and the following disclaimer in the
  *    documentation and/or other materials provided with the distribution.
  *
  * THIS SOFTWARE IS PROVIDED BY APPLE INC. ``AS IS'' AND ANY
  * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
  * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
  * PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL APPLE INC. OR
  * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
  * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
  * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
  * OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
  * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
  * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */

 #include "config.h"
 #include "AudioSampleBufferList.h"

 #if ENABLE(MEDIA_STREAM)

 #include "Logging.h"
 #include "VectorMath.h"
 #include <Accelerate/Accelerate.h>
 #include <AudioToolbox/AudioConverter.h>
 #include <wtf/SetForScope.h>

 namespace WebCore {

 using namespace VectorMath;

 Ref<AudioSampleBufferList> AudioSampleBufferList::create(const CAAudioStreamDescription& format, size_t maximumSampleCount)
 {
     return adoptRef(*new AudioSampleBufferList(format, maximumSampleCount));
 }

 AudioSampleBufferList::AudioSampleBufferList(const CAAudioStreamDescription& format, size_t maximumSampleCount)
     : m_internalFormat(makeUniqueRef<CAAudioStreamDescription>(format))
     , m_sampleCapacity(maximumSampleCount)
     , m_maxBufferSizePerChannel(maximumSampleCount * format.bytesPerFrame() / format.numberOfChannelStreams())
     , m_bufferList(makeUniqueRef<WebAudioBufferList>(m_internalFormat, m_maxBufferSizePerChannel))
 {
     ASSERT(format.sampleRate() >= 0);
 }

 AudioSampleBufferList::~AudioSampleBufferList() = default;

 void AudioSampleBufferList::setSampleCount(size_t count)
 {
     ASSERT(count <= m_sampleCapacity);
     if (count <= m_sampleCapacity)
         m_sampleCount = count;
 }

 void AudioSampleBufferList::applyGain(AudioBufferList& bufferList, float gain, AudioStreamDescription::PCMFormat format)
 {
     for (uint32_t i = 0; i < bufferList.mNumberBuffers; ++i) {
         switch (format) {
         case AudioStreamDescription::Int16: {
             int16_t* buffer = static_cast<int16_t*>(bufferList.mBuffers[i].mData);
             int frameCount = bufferList.mBuffers[i].mDataByteSize / sizeof(int16_t);
             for (int i = 0; i < frameCount; i++)
                 buffer[i] *= gain;
             break;
         }
         case AudioStreamDescription::Int32: {
             int32_t* buffer = static_cast<int32_t*>(bufferList.mBuffers[i].mData);
             int frameCount = bufferList.mBuffers[i].mDataByteSize / sizeof(int32_t);
             for (int i = 0; i < frameCount; i++)
                 buffer[i] *= gain;
             break;
             break;
         }
         case AudioStreamDescription::Float32: {
             float* buffer = static_cast<float*>(bufferList.mBuffers[i].mData);
             vDSP_vsmul(buffer, 1, &gain, buffer, 1, bufferList.mBuffers[i].mDataByteSize / sizeof(float));
             break;
         }
         case AudioStreamDescription::Float64: {
             double* buffer = static_cast<double*>(bufferList.mBuffers[i].mData);
             double gainAsDouble = gain;
             vDSP_vsmulD(buffer, 1, &gainAsDouble, buffer, 1, bufferList.mBuffers[i].mDataByteSize / sizeof(double));
             break;
         }
         case AudioStreamDescription::None:
             ASSERT_NOT_REACHED();
             break;
         }
     }
 }

 void AudioSampleBufferList::applyGain(float gain)
 {
     applyGain(m_bufferList.get(), gain, m_internalFormat->format());
 }

 OSStatus AudioSampleBufferList::mixFrom(const AudioSampleBufferList& source, size_t frameCount)
 {
     ASSERT(source.streamDescription() == streamDescription());

     if (source.streamDescription() != streamDescription())
         return kAudio_ParamError;

     if (frameCount > source.sampleCount())
         frameCount = source.sampleCount();

     if (frameCount > m_sampleCapacity)
         return kAudio_ParamError;

     m_sampleCount = frameCount;

     auto& sourceBuffer = source.bufferList();
     for (uint32_t i = 0; i < m_bufferList->bufferCount(); i++) {
         switch (m_internalFormat->format()) {
         case AudioStreamDescription::Int16: {
             int16_t* destination = static_cast<int16_t*>(m_bufferList->buffer(i)->mData);
             int16_t* source = static_cast<int16_t*>(sourceBuffer.buffer(i)->mData);
             for (size_t i = 0; i < frameCount; i++)
                 destination[i] += source[i];
             break;
         }
         case AudioStreamDescription::Int32: {
             int32_t* destination = static_cast<int32_t*>(m_bufferList->buffer(i)->mData);
             vDSP_vaddi(destination, 1, reinterpret_cast<int32_t*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
             break;
         }
         case AudioStreamDescription::Float32: {
             float* destination = static_cast<float*>(m_bufferList->buffer(i)->mData);
             vDSP_vadd(destination, 1, reinterpret_cast<float*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
             break;
         }
         case AudioStreamDescription::Float64: {
             double* destination = static_cast<double*>(m_bufferList->buffer(i)->mData);
             vDSP_vaddD(destination, 1, reinterpret_cast<double*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
             break;
         }
         case AudioStreamDescription::None:
             ASSERT_NOT_REACHED();
             break;
         }
     }

     return 0;
 }

 OSStatus AudioSampleBufferList::copyFrom(const AudioSampleBufferList& source, size_t frameCount)
 {
     ASSERT(source.streamDescription() == streamDescription());

     if (source.streamDescription() != streamDescription())
         return kAudio_ParamError;

     if (frameCount > source.sampleCount())
         frameCount = source.sampleCount();

     if (frameCount > m_sampleCapacity)
         return kAudio_ParamError;

     m_sampleCount = frameCount;

     for (uint32_t i = 0; i < m_bufferList->bufferCount(); i++) {
         uint8_t* sourceData = static_cast<uint8_t*>(source.bufferList().buffer(i)->mData);
         uint8_t* destination = static_cast<uint8_t*>(m_bufferList->buffer(i)->mData);
         memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
     }

     return 0;
 }

 OSStatus AudioSampleBufferList::copyTo(AudioBufferList& buffer, size_t frameCount)
 {
     if (frameCount > m_sampleCount)
         return kAudio_ParamError;
     if (buffer.mNumberBuffers > m_bufferList->bufferCount())
         return kAudio_ParamError;

     for (uint32_t i = 0; i < buffer.mNumberBuffers; i++) {
         uint8_t* sourceData = static_cast<uint8_t*>(m_bufferList->buffer(i)->mData);
         uint8_t* destination = static_cast<uint8_t*>(buffer.mBuffers[i].mData);
         memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
     }

     return 0;
 }

 void AudioSampleBufferList::reset()
 {
     m_sampleCount = 0;
     m_timestamp = 0;
     m_hostTime = -1;

     m_bufferList->reset();
 }

 void AudioSampleBufferList::zero()
 {
     zeroABL(m_bufferList.get(), m_internalFormat->bytesPerPacket() * m_sampleCapacity);
 }

 void AudioSampleBufferList::zeroABL(AudioBufferList& buffer, size_t byteCount)
 {
     for (uint32_t i = 0; i < buffer.mNumberBuffers; ++i)
         memset(buffer.mBuffers[i].mData, 0, byteCount);
 }

 struct AudioConverterFromABLContext {
     const AudioBufferList& buffer;
     size_t packetsAvailableToConvert;
     size_t bytesPerPacket;
 };

 static const OSStatus kRanOutOfInputDataStatus = '!mor';

 static OSStatus audioConverterFromABLCallback(AudioConverterRef, UInt32* ioNumberDataPackets, AudioBufferList* ioData, AudioStreamPacketDescription**, void* inRefCon)
 {
     if (!ioNumberDataPackets || !ioData || !inRefCon) {
         LOG_ERROR("AudioSampleBufferList::audioConverterCallback() invalid input to AudioConverterInput");
         return kAudioConverterErr_UnspecifiedError;
     }

     auto& context = *static_cast<AudioConverterFromABLContext*>(inRefCon);
     if (!context.packetsAvailableToConvert) {
         *ioNumberDataPackets = 0;
         return kRanOutOfInputDataStatus;
     }

     *ioNumberDataPackets = static_cast<UInt32>(context.packetsAvailableToConvert);

     for (uint32_t i = 0; i < ioData->mNumberBuffers; ++i) {
         ioData->mBuffers[i].mData = context.buffer.mBuffers[i].mData;
         ioData->mBuffers[i].mDataByteSize = context.packetsAvailableToConvert * context.bytesPerPacket;
     }
     context.packetsAvailableToConvert = 0;

     return 0;
 }

 OSStatus AudioSampleBufferList::copyFrom(const AudioBufferList& source, size_t frameCount, AudioConverterRef converter)
 {
     reset();

     AudioStreamBasicDescription inputFormat;
     UInt32 propertyDataSize = sizeof(inputFormat);
     AudioConverterGetProperty(converter, kAudioConverterCurrentInputStreamDescription, &propertyDataSize, &inputFormat);
     ASSERT(frameCount <= source.mBuffers[0].mDataByteSize / inputFormat.mBytesPerPacket);

     AudioConverterFromABLContext context { source, frameCount, inputFormat.mBytesPerPacket };

 #if !LOG_DISABLED
     AudioStreamBasicDescription outputFormat;
     propertyDataSize = sizeof(outputFormat);
     AudioConverterGetProperty(converter, kAudioConverterCurrentOutputStreamDescription, &propertyDataSize, &outputFormat);

     ASSERT(CAAudioStreamDescription(outputFormat).numberOfChannelStreams() == m_bufferList->bufferCount());
     for (uint32_t i = 0; i < m_bufferList->bufferCount(); ++i) {
         ASSERT(m_bufferList->buffer(i)->mData);
         ASSERT(m_bufferList->buffer(i)->mDataByteSize);
     }
 #endif

     UInt32 samplesConverted = m_sampleCapacity;
     OSStatus err = AudioConverterFillComplexBuffer(converter, audioConverterFromABLCallback, &context, &samplesConverted, m_bufferList->list(), nullptr);
     if (!err || err == kRanOutOfInputDataStatus) {
         m_sampleCount = samplesConverted;
         return 0;
     }

     LOG_ERROR("AudioSampleBufferList::copyFrom(%p) AudioConverterFillComplexBuffer returned error %d (%.4s)", this, (int)err, (char*)&err);
     m_sampleCount = std::min(m_sampleCapacity, static_cast<size_t>(samplesConverted));
     zero();
     return err;
 }

 OSStatus AudioSampleBufferList::copyFrom(AudioSampleBufferList& source, size_t frameCount, AudioConverterRef converter)
 {
     return copyFrom(source.bufferList(), frameCount, converter);
 }

 OSStatus AudioSampleBufferList::copyFrom(CARingBuffer& ringBuffer, size_t sampleCount, uint64_t startFrame, CARingBuffer::FetchMode mode)
 {
     reset();
     if (ringBuffer.fetch(bufferList().list(), sampleCount, startFrame, mode) != CARingBuffer::Ok)
         return kAudio_ParamError;

     m_sampleCount = sampleCount;
     return 0;
 }

 } // namespace WebCore

 #endif // ENABLE(MEDIA_STREAM)
	/*
	* Copyright (C) 2017 Apple Inc. All rights reserved.
	*
	* Redistribution and use in source and binary forms, with or without
	* modification, are permitted provided that the following conditions
	* are met:
	* 1. Redistributions of source code must retain the above copyright
	* notice, this list of conditions and the following disclaimer.
	* 2. Redistributions in binary form must reproduce the above copyright
	* notice, this list of conditions and the following disclaimer in the
	* documentation and/or other materials provided with the distribution.
	*
	* THIS SOFTWARE IS PROVIDED BY APPLE INC. ``AS IS'' AND ANY
	* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
	* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
	* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR
	* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
	* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
	* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
	* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
	* OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
	* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
	* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
	*/

	#include "config.h"
	#include "AudioSampleBufferList.h"

	#if ENABLE(MEDIA_STREAM)

	#include "Logging.h"
	#include "VectorMath.h"
	#include <Accelerate/Accelerate.h>
	#include <AudioToolbox/AudioConverter.h>
	#include <wtf/SetForScope.h>

	namespace WebCore {

	using namespace VectorMath;

	Ref<AudioSampleBufferList> AudioSampleBufferList::create(const CAAudioStreamDescription& format, size_t maximumSampleCount)
	{
	return adoptRef(*new AudioSampleBufferList(format, maximumSampleCount));
	}

	AudioSampleBufferList::AudioSampleBufferList(const CAAudioStreamDescription& format, size_t maximumSampleCount)
	: m_internalFormat(makeUniqueRef<CAAudioStreamDescription>(format))
	, m_sampleCapacity(maximumSampleCount)
	, m_maxBufferSizePerChannel(maximumSampleCount * format.bytesPerFrame() / format.numberOfChannelStreams())
	, m_bufferList(makeUniqueRef<WebAudioBufferList>(m_internalFormat, m_maxBufferSizePerChannel))
	{
	ASSERT(format.sampleRate() >= 0);
	}

	AudioSampleBufferList::~AudioSampleBufferList() = default;

	void AudioSampleBufferList::setSampleCount(size_t count)
	{
	ASSERT(count <= m_sampleCapacity);
	if (count <= m_sampleCapacity)
	m_sampleCount = count;
	}

	void AudioSampleBufferList::applyGain(AudioBufferList& bufferList, float gain, AudioStreamDescription::PCMFormat format)
	{
	for (uint32_t i = 0; i < bufferList.mNumberBuffers; ++i) {
	switch (format) {
	case AudioStreamDescription::Int16: {
	int16_t* buffer = static_cast<int16_t*>(bufferList.mBuffers[i].mData);
	int frameCount = bufferList.mBuffers[i].mDataByteSize / sizeof(int16_t);
	for (int i = 0; i < frameCount; i++)
	buffer[i] *= gain;
	break;
	}
	case AudioStreamDescription::Int32: {
	int32_t* buffer = static_cast<int32_t*>(bufferList.mBuffers[i].mData);
	int frameCount = bufferList.mBuffers[i].mDataByteSize / sizeof(int32_t);
	for (int i = 0; i < frameCount; i++)
	buffer[i] *= gain;
	break;
	break;
	}
	case AudioStreamDescription::Float32: {
	float* buffer = static_cast<float*>(bufferList.mBuffers[i].mData);
	vDSP_vsmul(buffer, 1, &gain, buffer, 1, bufferList.mBuffers[i].mDataByteSize / sizeof(float));
	break;
	}
	case AudioStreamDescription::Float64: {
	double* buffer = static_cast<double*>(bufferList.mBuffers[i].mData);
	double gainAsDouble = gain;
	vDSP_vsmulD(buffer, 1, &gainAsDouble, buffer, 1, bufferList.mBuffers[i].mDataByteSize / sizeof(double));
	break;
	}
	case AudioStreamDescription::None:
	ASSERT_NOT_REACHED();
	break;
	}
	}
	}

	void AudioSampleBufferList::applyGain(float gain)
	{
	applyGain(m_bufferList.get(), gain, m_internalFormat->format());
	}

	OSStatus AudioSampleBufferList::mixFrom(const AudioSampleBufferList& source, size_t frameCount)
	{
	ASSERT(source.streamDescription() == streamDescription());

	if (source.streamDescription() != streamDescription())
	return kAudio_ParamError;

	if (frameCount > source.sampleCount())
	frameCount = source.sampleCount();

	if (frameCount > m_sampleCapacity)
	return kAudio_ParamError;

	m_sampleCount = frameCount;

	auto& sourceBuffer = source.bufferList();
	for (uint32_t i = 0; i < m_bufferList->bufferCount(); i++) {
	switch (m_internalFormat->format()) {
	case AudioStreamDescription::Int16: {
	int16_t* destination = static_cast<int16_t*>(m_bufferList->buffer(i)->mData);
	int16_t* source = static_cast<int16_t*>(sourceBuffer.buffer(i)->mData);
	for (size_t i = 0; i < frameCount; i++)
	destination[i] += source[i];
	break;
	}
	case AudioStreamDescription::Int32: {
	int32_t* destination = static_cast<int32_t*>(m_bufferList->buffer(i)->mData);
	vDSP_vaddi(destination, 1, reinterpret_cast<int32_t*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
	break;
	}
	case AudioStreamDescription::Float32: {
	float* destination = static_cast<float*>(m_bufferList->buffer(i)->mData);
	vDSP_vadd(destination, 1, reinterpret_cast<float*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
	break;
	}
	case AudioStreamDescription::Float64: {
	double* destination = static_cast<double*>(m_bufferList->buffer(i)->mData);
	vDSP_vaddD(destination, 1, reinterpret_cast<double*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
	break;
	}
	case AudioStreamDescription::None:
	ASSERT_NOT_REACHED();
	break;
	}
	}

	return 0;
	}

	OSStatus AudioSampleBufferList::copyFrom(const AudioSampleBufferList& source, size_t frameCount)
	{
	ASSERT(source.streamDescription() == streamDescription());

	if (source.streamDescription() != streamDescription())
	return kAudio_ParamError;

	if (frameCount > source.sampleCount())
	frameCount = source.sampleCount();

	if (frameCount > m_sampleCapacity)
	return kAudio_ParamError;

	m_sampleCount = frameCount;

	for (uint32_t i = 0; i < m_bufferList->bufferCount(); i++) {
	uint8_t* sourceData = static_cast<uint8_t*>(source.bufferList().buffer(i)->mData);
	uint8_t* destination = static_cast<uint8_t*>(m_bufferList->buffer(i)->mData);
	memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
	}

	return 0;
	}

	OSStatus AudioSampleBufferList::copyTo(AudioBufferList& buffer, size_t frameCount)
	{
	if (frameCount > m_sampleCount)
	return kAudio_ParamError;
	if (buffer.mNumberBuffers > m_bufferList->bufferCount())
	return kAudio_ParamError;

	for (uint32_t i = 0; i < buffer.mNumberBuffers; i++) {
	uint8_t* sourceData = static_cast<uint8_t*>(m_bufferList->buffer(i)->mData);
	uint8_t* destination = static_cast<uint8_t*>(buffer.mBuffers[i].mData);
	memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
	}

	return 0;
	}

	void AudioSampleBufferList::reset()
	{
	m_sampleCount = 0;
	m_timestamp = 0;
	m_hostTime = -1;

	m_bufferList->reset();
	}

	void AudioSampleBufferList::zero()
	{
	zeroABL(m_bufferList.get(), m_internalFormat->bytesPerPacket() * m_sampleCapacity);
	}

	void AudioSampleBufferList::zeroABL(AudioBufferList& buffer, size_t byteCount)
	{
	for (uint32_t i = 0; i < buffer.mNumberBuffers; ++i)
	memset(buffer.mBuffers[i].mData, 0, byteCount);
	}

	struct AudioConverterFromABLContext {
	const AudioBufferList& buffer;
	size_t packetsAvailableToConvert;
	size_t bytesPerPacket;
	};

	static const OSStatus kRanOutOfInputDataStatus = '!mor';

	static OSStatus audioConverterFromABLCallback(AudioConverterRef, UInt32* ioNumberDataPackets, AudioBufferList* ioData, AudioStreamPacketDescription*, void inRefCon)
	{
	if (!ioNumberDataPackets \|\| !ioData \|\| !inRefCon) {
	LOG_ERROR("AudioSampleBufferList::audioConverterCallback() invalid input to AudioConverterInput");
	return kAudioConverterErr_UnspecifiedError;
	}

	auto& context = static_cast<AudioConverterFromABLContext>(inRefCon);
	if (!context.packetsAvailableToConvert) {
	*ioNumberDataPackets = 0;
	return kRanOutOfInputDataStatus;
	}

	*ioNumberDataPackets = static_cast<UInt32>(context.packetsAvailableToConvert);

	for (uint32_t i = 0; i < ioData->mNumberBuffers; ++i) {
	ioData->mBuffers[i].mData = context.buffer.mBuffers[i].mData;
	ioData->mBuffers[i].mDataByteSize = context.packetsAvailableToConvert * context.bytesPerPacket;
	}
	context.packetsAvailableToConvert = 0;

	return 0;
	}

	OSStatus AudioSampleBufferList::copyFrom(const AudioBufferList& source, size_t frameCount, AudioConverterRef converter)
	{
	reset();

	AudioStreamBasicDescription inputFormat;
	UInt32 propertyDataSize = sizeof(inputFormat);
	AudioConverterGetProperty(converter, kAudioConverterCurrentInputStreamDescription, &propertyDataSize, &inputFormat);
	ASSERT(frameCount <= source.mBuffers[0].mDataByteSize / inputFormat.mBytesPerPacket);

	AudioConverterFromABLContext context { source, frameCount, inputFormat.mBytesPerPacket };

	#if !LOG_DISABLED
	AudioStreamBasicDescription outputFormat;
	propertyDataSize = sizeof(outputFormat);
	AudioConverterGetProperty(converter, kAudioConverterCurrentOutputStreamDescription, &propertyDataSize, &outputFormat);

	ASSERT(CAAudioStreamDescription(outputFormat).numberOfChannelStreams() == m_bufferList->bufferCount());
	for (uint32_t i = 0; i < m_bufferList->bufferCount(); ++i) {
	ASSERT(m_bufferList->buffer(i)->mData);
	ASSERT(m_bufferList->buffer(i)->mDataByteSize);
	}
	#endif

	UInt32 samplesConverted = m_sampleCapacity;
	OSStatus err = AudioConverterFillComplexBuffer(converter, audioConverterFromABLCallback, &context, &samplesConverted, m_bufferList->list(), nullptr);
	if (!err \|\| err == kRanOutOfInputDataStatus) {
	m_sampleCount = samplesConverted;
	return 0;
	}

	LOG_ERROR("AudioSampleBufferList::copyFrom(%p) AudioConverterFillComplexBuffer returned error %d (%.4s)", this, (int)err, (char*)&err);
	m_sampleCount = std::min(m_sampleCapacity, static_cast<size_t>(samplesConverted));
	zero();
	return err;
	}

	OSStatus AudioSampleBufferList::copyFrom(AudioSampleBufferList& source, size_t frameCount, AudioConverterRef converter)
	{
	return copyFrom(source.bufferList(), frameCount, converter);
	}

	OSStatus AudioSampleBufferList::copyFrom(CARingBuffer& ringBuffer, size_t sampleCount, uint64_t startFrame, CARingBuffer::FetchMode mode)
	{
	reset();
	if (ringBuffer.fetch(bufferList().list(), sampleCount, startFrame, mode) != CARingBuffer::Ok)
	return kAudio_ParamError;

	m_sampleCount = sampleCount;
	return 0;
	}

	} // namespace WebCore

	#endif // ENABLE(MEDIA_STREAM)