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#include "soundio/sounddevice.h"
#include <QtDebug>
#include <cstring> // for memcpy and strcmp
#include "soundio/soundmanager.h"
#include "soundio/soundmanagerutil.h"
#include "util/debug.h"
#include "util/defs.h"
#include "util/sample.h"
SoundDevice::SoundDevice(UserSettingsPointer config, SoundManager* sm)
: m_pConfig(config),
m_pSoundManager(sm),
m_strDisplayName("Unknown Soundcard"),
m_iNumOutputChannels(2),
m_iNumInputChannels(2),
m_dSampleRate(44100.0),
m_hostAPI("Unknown API"),
m_framesPerBuffer(0) {
}
int SoundDevice::getNumInputChannels() const {
return m_iNumInputChannels;
}
int SoundDevice::getNumOutputChannels() const {
return m_iNumOutputChannels;
}
void SoundDevice::setSampleRate(double sampleRate) {
if (sampleRate <= 0.0) {
// this is the default value used elsewhere in this file
sampleRate = 44100.0;
}
m_dSampleRate = sampleRate;
}
void SoundDevice::setFramesPerBuffer(unsigned int framesPerBuffer) {
if (framesPerBuffer * 2 > MAX_BUFFER_LEN) {
// framesPerBuffer * 2 because a frame will generally end up
// being 2 samples and MAX_BUFFER_LEN is a number of samples
// this isn't checked elsewhere, so...
reportFatalErrorAndQuit("framesPerBuffer too big in "
"SoundDevice::setFramesPerBuffer(uint)");
}
m_framesPerBuffer = framesPerBuffer;
}
SoundDeviceError SoundDevice::addOutput(const AudioOutputBuffer &out) {
// Check if the output channels are already used
foreach (AudioOutputBuffer myOut, m_audioOutputs) {
if (out.channelsClash(myOut)) {
return SOUNDDEVICE_ERROR_DUPLICATE_OUTPUT_CHANNEL;
}
}
if (out.getChannelGroup().getChannelBase()
+ out.getChannelGroup().getChannelCount() > getNumOutputChannels()) {
return SOUNDDEVICE_ERROR_EXCESSIVE_OUTPUT_CHANNEL;
}
m_audioOutputs.append(out);
return SOUNDDEVICE_ERROR_OK;
}
void SoundDevice::clearOutputs() {
m_audioOutputs.clear();
}
SoundDeviceError SoundDevice::addInput(const AudioInputBuffer &in) {
// DON'T check if the input channels are already used, there's no reason
// we can't send the same inputted samples to different places in mixxx.
// -- bkgood 20101108
if (in.getChannelGroup().getChannelBase()
+ in.getChannelGroup().getChannelCount() > getNumInputChannels()) {
return SOUNDDEVICE_ERROR_EXCESSIVE_INPUT_CHANNEL;
}
m_audioInputs.append(in);
return SOUNDDEVICE_ERROR_OK;
}
void SoundDevice::clearInputs() {
m_audioInputs.clear();
}
bool SoundDevice::operator==(const SoundDevice &other) const {
return m_deviceId == other.getDeviceId();
}
void SoundDevice::composeOutputBuffer(CSAMPLE* outputBuffer,
const SINT framesToCompose,
const SINT framesReadOffset,
const int iFrameSize) {
//qDebug() << "SoundDevice::composeOutputBuffer()"
// << device->getInternalName()
// << framesToCompose << iFrameSize;
// Interlace Audio data onto portaudio buffer. We iterate through the
// source list to find out what goes in the buffer data is interlaced in
// the order of the list
if (iFrameSize == 2 && m_audioOutputs.size() == 1 &&
m_audioOutputs.at(0).getChannelGroup().getChannelCount() == 2) {
// Special case for one stereo device only
const AudioOutputBuffer& out = m_audioOutputs.at(0);
const CSAMPLE* pAudioOutputBuffer = out.getBuffer(); // Always Stereo
pAudioOutputBuffer = &pAudioOutputBuffer[framesReadOffset*2];
SampleUtil::copyClampBuffer(outputBuffer, pAudioOutputBuffer,
framesToCompose * 2);
} else {
// Reset sample for each open channel
SampleUtil::clear(outputBuffer, framesToCompose * iFrameSize);
for (QList<AudioOutputBuffer>::iterator i = m_audioOutputs.begin(),
e = m_audioOutputs.end(); i != e; ++i) {
AudioOutputBuffer& out = *i;
const ChannelGroup outChans = out.getChannelGroup();
const int iChannelCount = outChans.getChannelCount();
const int iChannelBase = outChans.getChannelBase();
const CSAMPLE* pAudioOutputBuffer = out.getBuffer();
// advanced to offset; pAudioOutputBuffer is always stereo
pAudioOutputBuffer = &pAudioOutputBuffer[framesReadOffset*2];
if (iChannelCount == 1) {
// All AudioOutputs are stereo as of Mixxx 1.12.0. If we have a mono
// output then we need to downsample.
for (SINT iFrameNo = 0; iFrameNo < framesToCompose; ++iFrameNo) {
// iFrameBase is the "base sample" in a frame (ie. the first
// sample in a frame)
const SINT iFrameBase = iFrameNo * iFrameSize;
outputBuffer[iFrameBase + iChannelBase] = SampleUtil::clampSample(
(pAudioOutputBuffer[iFrameNo * 2] +
pAudioOutputBuffer[iFrameNo * 2 + 1]) / 2.0f);
}
} else {
for (SINT iFrameNo = 0; iFrameNo < framesToCompose; ++iFrameNo) {
// iFrameBase is the "base sample" in a frame (ie. the first
// sample in a frame)
const SINT iFrameBase = iFrameNo * iFrameSize;
const SINT iLocalFrameBase = iFrameNo * iChannelCount;
// this will make sure a sample from each channel is copied
for (int iChannel = 0; iChannel < iChannelCount; ++iChannel) {
outputBuffer[iFrameBase + iChannelBase + iChannel] =
SampleUtil::clampSample(
pAudioOutputBuffer[iLocalFrameBase + iChannel]);
// Input audio pass-through (useful for debugging)
//if (in)
// output[iFrameBase + src.channelBase + iChannel] =
// in[iFrameBase + src.channelBase + iChannel];
}
}
}
}
}
}
void SoundDevice::composeInputBuffer(const CSAMPLE* inputBuffer,
const SINT framesToPush,
const SINT framesWriteOffset,
const int iFrameSize) {
//qDebug() << "SoundManager::pushBuffer"
// << framesToPush << framesWriteOffset << iFrameSize;
// This function is called a *lot* and is a big source of CPU usage.
// It needs to be very fast.
// If the framesize is only 2, then we only have one pair of input channels
// That means we don't have to do any deinterlacing, and we can pass
// the audio on to its intended destination.
if (iFrameSize == 1 && m_audioInputs.size() == 1 &&
m_audioInputs.at(0).getChannelGroup().getChannelCount() == 1) {
// One mono device only
const AudioInputBuffer& in = m_audioInputs.at(0);
CSAMPLE* pInputBuffer = in.getBuffer(); // Always Stereo
pInputBuffer = &pInputBuffer[framesWriteOffset * 2];
for (SINT iFrameNo = 0; iFrameNo < framesToPush; ++iFrameNo) {
pInputBuffer[iFrameNo * 2] =
inputBuffer[iFrameNo];
pInputBuffer[iFrameNo * 2 + 1] =
inputBuffer[iFrameNo];
}
} else if (iFrameSize == 2 && m_audioInputs.size() == 1 &&
m_audioInputs.at(0).getChannelGroup().getChannelCount() == 2) {
// One stereo device only
const AudioInputBuffer& in = m_audioInputs.at(0);
CSAMPLE* pInputBuffer = in.getBuffer(); // Always Stereo
pInputBuffer = &pInputBuffer[framesWriteOffset * 2];
SampleUtil::copy(pInputBuffer, inputBuffer, framesToPush * 2);
} else {
// Non Stereo input (iFrameSize != 2)
// Do crazy deinterleaving of the audio into the correct m_inputBuffers.
for (auto i = m_audioInputs.constBegin(), e = m_audioInputs.constEnd(); i != e; ++i) {
const AudioInputBuffer& in = *i;
ChannelGroup chanGroup = in.getChannelGroup();
int iChannelCount = chanGroup.getChannelCount();
int iChannelBase = chanGroup.getChannelBase();
CSAMPLE* pInputBuffer = in.getBuffer();
pInputBuffer = &pInputBuffer[framesWriteOffset * 2];
for (SINT iFrameNo = 0; iFrameNo < framesToPush; ++iFrameNo) {
// iFrameBase is the "base sample" in a frame (ie. the first
// sample in a frame)
SINT iFrameBase = iFrameNo * iFrameSize;
SINT iLocalFrameBase = iFrameNo * 2;
if (iChannelCount == 1) {
pInputBuffer[iLocalFrameBase] =
inputBuffer[iFrameBase + iChannelBase];
pInputBuffer[iLocalFrameBase + 1] =
inputBuffer[iFrameBase + iChannelBase];
} else if (iChannelCount > 1) {
pInputBuffer[iLocalFrameBase] =
inputBuffer[iFrameBase + iChannelBase];
pInputBuffer[iLocalFrameBase + 1] =
inputBuffer[iFrameBase + iChannelBase + 1];
}
}
}
}
}
void SoundDevice::clearInputBuffer(const SINT framesToPush,
const SINT framesWriteOffset) {
for (auto i = m_audioInputs.constBegin(), e = m_audioInputs.constEnd(); i != e; ++i) {
const AudioInputBuffer& in = *i;
CSAMPLE* pInputBuffer</
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