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audacity/src/AudioIOBase.cpp
Yuri Chornoivan d1ada5f08c Fix minor typos
2020-04-11 10:06:24 +01:00

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/**********************************************************************
Audacity: A Digital Audio Editor
AudioIOBase.cpp
Paul Licameli split from AudioIO.cpp
**********************************************************************/
#include "Audacity.h"
#include "AudioIOBase.h"
#include "Experimental.h"
#include <wx/sstream.h>
#include <wx/txtstrm.h>
#include "Envelope.h"
#include "Prefs.h"
#include "prefs/RecordingPrefs.h"
#include "widgets/MeterPanelBase.h"
#if USE_PORTMIXER
#include "portmixer.h"
#endif
#ifdef EXPERIMENTAL_MIDI_OUT
#include "../lib-src/portmidi/pm_common/portmidi.h"
#endif
int AudioIOBase::mCachedPlaybackIndex = -1;
std::vector<long> AudioIOBase::mCachedPlaybackRates;
int AudioIOBase::mCachedCaptureIndex = -1;
std::vector<long> AudioIOBase::mCachedCaptureRates;
std::vector<long> AudioIOBase::mCachedSampleRates;
double AudioIOBase::mCachedBestRateIn = 0.0;
const int AudioIOBase::StandardRates[] = {
8000,
11025,
16000,
22050,
32000,
44100,
48000,
88200,
96000,
176400,
192000,
352800,
384000
};
const int AudioIOBase::NumStandardRates = WXSIZEOF(AudioIOBase::StandardRates);
const int AudioIOBase::RatesToTry[] = {
8000,
9600,
11025,
12000,
15000,
16000,
22050,
24000,
32000,
44100,
48000,
88200,
96000,
176400,
192000,
352800,
384000
};
const int AudioIOBase::NumRatesToTry = WXSIZEOF(AudioIOBase::RatesToTry);
wxString AudioIOBase::DeviceName(const PaDeviceInfo* info)
{
wxString infoName = wxSafeConvertMB2WX(info->name);
return infoName;
}
wxString AudioIOBase::HostName(const PaDeviceInfo* info)
{
wxString hostapiName = wxSafeConvertMB2WX(Pa_GetHostApiInfo(info->hostApi)->name);
return hostapiName;
}
std::unique_ptr<AudioIOBase> AudioIOBase::ugAudioIO;
AudioIOBase *AudioIOBase::Get()
{
return ugAudioIO.get();
}
AudioIOBase::~AudioIOBase() = default;
void AudioIOBase::SetMixer(int inputSource)
{
#if defined(USE_PORTMIXER)
int oldRecordSource = Px_GetCurrentInputSource(mPortMixer);
if ( inputSource != oldRecordSource )
Px_SetCurrentInputSource(mPortMixer, inputSource);
#endif
}
void AudioIOBase::HandleDeviceChange()
{
// This should not happen, but it would screw things up if it did.
// Vaughan, 2010-10-08: But it *did* happen, due to a bug, and nobody
// caught it because this method just returned. Added wxASSERT().
wxASSERT(!IsStreamActive());
if (IsStreamActive())
return;
// get the selected record and playback devices
const int playDeviceNum = getPlayDevIndex();
const int recDeviceNum = getRecordDevIndex();
// If no change needed, return
if (mCachedPlaybackIndex == playDeviceNum &&
mCachedCaptureIndex == recDeviceNum)
return;
// cache playback/capture rates
mCachedPlaybackRates = GetSupportedPlaybackRates(playDeviceNum);
mCachedCaptureRates = GetSupportedCaptureRates(recDeviceNum);
mCachedSampleRates = GetSupportedSampleRates(playDeviceNum, recDeviceNum);
mCachedPlaybackIndex = playDeviceNum;
mCachedCaptureIndex = recDeviceNum;
mCachedBestRateIn = 0.0;
#if defined(USE_PORTMIXER)
// if we have a PortMixer object, close it down
if (mPortMixer) {
#if __WXMAC__
// on the Mac we must make sure that we restore the hardware playthrough
// state of the sound device to what it was before, because there isn't
// a UI for this (!)
if (Px_SupportsPlaythrough(mPortMixer) && mPreviousHWPlaythrough >= 0.0)
Px_SetPlaythrough(mPortMixer, mPreviousHWPlaythrough);
mPreviousHWPlaythrough = -1.0;
#endif
Px_CloseMixer(mPortMixer);
mPortMixer = NULL;
}
// that might have given us no rates whatsoever, so we have to guess an
// answer to do the next bit
int numrates = mCachedSampleRates.size();
int highestSampleRate;
if (numrates > 0)
{
highestSampleRate = mCachedSampleRates[numrates - 1];
}
else
{ // we don't actually have any rates that work for Rec and Play. Guess one
// to use for messing with the mixer, which doesn't actually do either
highestSampleRate = 44100;
// mCachedSampleRates is still empty, but it's not used again, so
// can ignore
}
mInputMixerWorks = false;
mEmulateMixerOutputVol = true;
mMixerOutputVol = 1.0;
int error;
// This tries to open the device with the samplerate worked out above, which
// will be the highest available for play and record on the device, or
// 44.1kHz if the info cannot be fetched.
PaStream *stream;
PaStreamParameters playbackParameters;
playbackParameters.device = playDeviceNum;
playbackParameters.sampleFormat = paFloat32;
playbackParameters.hostApiSpecificStreamInfo = NULL;
playbackParameters.channelCount = 1;
if (Pa_GetDeviceInfo(playDeviceNum))
playbackParameters.suggestedLatency =
Pa_GetDeviceInfo(playDeviceNum)->defaultLowOutputLatency;
else
playbackParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0;
PaStreamParameters captureParameters;
captureParameters.device = recDeviceNum;
captureParameters.sampleFormat = paFloat32;;
captureParameters.hostApiSpecificStreamInfo = NULL;
captureParameters.channelCount = 1;
if (Pa_GetDeviceInfo(recDeviceNum))
captureParameters.suggestedLatency =
Pa_GetDeviceInfo(recDeviceNum)->defaultLowInputLatency;
else
captureParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0;
// try opening for record and playback
// Not really doing I/O so pass nullptr for the callback function
error = Pa_OpenStream(&stream,
&captureParameters, &playbackParameters,
highestSampleRate, paFramesPerBufferUnspecified,
paClipOff | paDitherOff,
nullptr, NULL);
if (!error) {
// Try portmixer for this stream
mPortMixer = Px_OpenMixer(stream, 0);
if (!mPortMixer) {
Pa_CloseStream(stream);
error = true;
}
}
// if that failed, try just for record
if( error ) {
error = Pa_OpenStream(&stream,
&captureParameters, NULL,
highestSampleRate, paFramesPerBufferUnspecified,
paClipOff | paDitherOff,
nullptr, NULL);
if (!error) {
mPortMixer = Px_OpenMixer(stream, 0);
if (!mPortMixer) {
Pa_CloseStream(stream);
error = true;
}
}
}
// finally, try just for playback
if ( error ) {
error = Pa_OpenStream(&stream,
NULL, &playbackParameters,
highestSampleRate, paFramesPerBufferUnspecified,
paClipOff | paDitherOff,
nullptr, NULL);
if (!error) {
mPortMixer = Px_OpenMixer(stream, 0);
if (!mPortMixer) {
Pa_CloseStream(stream);
error = true;
}
}
}
// FIXME: TRAP_ERR errors in HandleDeviceChange not reported.
// if it's still not working, give up
if( error )
return;
// Set input source
#if USE_PORTMIXER
int sourceIndex;
if (gPrefs->Read(wxT("/AudioIO/RecordingSourceIndex"), &sourceIndex)) {
if (sourceIndex >= 0) {
//the current index of our source may be different because the stream
//is a combination of two devices, so update it.
sourceIndex = getRecordSourceIndex(mPortMixer);
if (sourceIndex >= 0)
SetMixer(sourceIndex);
}
}
#endif
// Determine mixer capabilities - if it doesn't support control of output
// signal level, we emulate it (by multiplying this value by all outgoing
// samples)
mMixerOutputVol = Px_GetPCMOutputVolume(mPortMixer);
mEmulateMixerOutputVol = false;
Px_SetPCMOutputVolume(mPortMixer, 0.0);
if (Px_GetPCMOutputVolume(mPortMixer) > 0.1)
mEmulateMixerOutputVol = true;
Px_SetPCMOutputVolume(mPortMixer, 0.2f);
if (Px_GetPCMOutputVolume(mPortMixer) < 0.1 ||
Px_GetPCMOutputVolume(mPortMixer) > 0.3)
mEmulateMixerOutputVol = true;
Px_SetPCMOutputVolume(mPortMixer, mMixerOutputVol);
float inputVol = Px_GetInputVolume(mPortMixer);
mInputMixerWorks = true; // assume it works unless proved wrong
Px_SetInputVolume(mPortMixer, 0.0);
if (Px_GetInputVolume(mPortMixer) > 0.1)
mInputMixerWorks = false; // can't set to zero
Px_SetInputVolume(mPortMixer, 0.2f);
if (Px_GetInputVolume(mPortMixer) < 0.1 ||
Px_GetInputVolume(mPortMixer) > 0.3)
mInputMixerWorks = false; // can't set level accurately
Px_SetInputVolume(mPortMixer, inputVol);
Pa_CloseStream(stream);
#if 0
wxPrintf("PortMixer: Playback: %s Recording: %s\n",
mEmulateMixerOutputVol? "emulated": "native",
mInputMixerWorks? "hardware": "no control");
#endif
mMixerOutputVol = 1.0;
#endif // USE_PORTMIXER
}
void AudioIOBase::SetCaptureMeter(AudacityProject *project, MeterPanelBase *meter)
{
if (( mOwningProject ) && ( mOwningProject != project))
return;
if (meter)
{
mInputMeter = meter;
mInputMeter->Reset(mRate, true);
}
else
mInputMeter.Release();
}
void AudioIOBase::SetPlaybackMeter(AudacityProject *project, MeterPanelBase *meter)
{
if (( mOwningProject ) && ( mOwningProject != project))
return;
if (meter)
{
mOutputMeter = meter;
mOutputMeter->Reset(mRate, true);
}
else
mOutputMeter.Release();
}
bool AudioIOBase::IsPaused() const
{
return mPaused;
}
bool AudioIOBase::IsBusy() const
{
if (mStreamToken != 0)
return true;
return false;
}
bool AudioIOBase::IsStreamActive() const
{
bool isActive = false;
// JKC: Not reporting any Pa error, but that looks OK.
if( mPortStreamV19 )
isActive = (Pa_IsStreamActive( mPortStreamV19 ) > 0);
#ifdef EXPERIMENTAL_MIDI_OUT
if( mMidiStreamActive && !mMidiOutputComplete )
isActive = true;
#endif
return isActive;
}
bool AudioIOBase::IsStreamActive(int token) const
{
return (this->IsStreamActive() && this->IsAudioTokenActive(token));
}
bool AudioIOBase::IsAudioTokenActive(int token) const
{
return ( token > 0 && token == mStreamToken );
}
bool AudioIOBase::IsMonitoring() const
{
return ( mPortStreamV19 && mStreamToken==0 );
}
void AudioIOBase::PlaybackSchedule::Init(
const double t0, const double t1,
const AudioIOStartStreamOptions &options,
const RecordingSchedule *pRecordingSchedule )
{
if ( pRecordingSchedule )
// It does not make sense to apply the time warp during overdub recording,
// which defeats the purpose of making the recording synchronized with
// the existing audio. (Unless we figured out the inverse warp of the
// captured samples in real time.)
// So just quietly ignore the time track.
mEnvelope = nullptr;
else
mEnvelope = options.envelope;
mT0 = t0;
if (pRecordingSchedule)
mT0 -= pRecordingSchedule->mPreRoll;
mT1 = t1;
if (pRecordingSchedule)
// adjust mT1 so that we don't give paComplete too soon to fill up the
// desired length of recording
mT1 -= pRecordingSchedule->mLatencyCorrection;
// Main thread's initialization of mTime
SetTrackTime( mT0 );
mPlayMode = options.playLooped
? PlaybackSchedule::PLAY_LOOPED
: PlaybackSchedule::PLAY_STRAIGHT;
mCutPreviewGapStart = options.cutPreviewGapStart;
mCutPreviewGapLen = options.cutPreviewGapLen;
#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
bool scrubbing = (options.pScrubbingOptions != nullptr);
// Scrubbing is not compatible with looping or recording or a time track!
if (scrubbing)
{
const auto &scrubOptions = *options.pScrubbingOptions;
if (pRecordingSchedule ||
Looping() ||
mEnvelope ||
scrubOptions.maxSpeed < ScrubbingOptions::MinAllowedScrubSpeed()) {
wxASSERT(false);
scrubbing = false;
}
else {
if (scrubOptions.isPlayingAtSpeed)
mPlayMode = PLAY_AT_SPEED;
else if (scrubOptions.isKeyboardScrubbing)
mPlayMode = PLAY_KEYBOARD_SCRUB;
else
mPlayMode = PLAY_SCRUB;
}
}
#endif
mWarpedTime = 0.0;
#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
if (Scrubbing())
mWarpedLength = 0.0f;
else
#endif
mWarpedLength = RealDuration(mT1);
}
double AudioIOBase::PlaybackSchedule::LimitTrackTime() const
{
// Track time readout for the main thread
// Allows for forward or backward play
return ClampTrackTime( GetTrackTime() );
}
double AudioIOBase::PlaybackSchedule::ClampTrackTime( double trackTime ) const
{
if (ReversedTime())
return std::max(mT1, std::min(mT0, trackTime));
else
return std::max(mT0, std::min(mT1, trackTime));
}
double AudioIOBase::PlaybackSchedule::NormalizeTrackTime() const
{
// Track time readout for the main thread
// dmazzoni: This function is needed for two reasons:
// One is for looped-play mode - this function makes sure that the
// position indicator keeps wrapping around. The other reason is
// more subtle - it's because PortAudio can query the hardware for
// the current stream time, and this query is not always accurate.
// Sometimes it's a little behind or ahead, and so this function
// makes sure that at least we clip it to the selection.
//
// msmeyer: There is also the possibility that we are using "cut preview"
// mode. In this case, we should jump over a defined "gap" in the
// audio.
double absoluteTime;
#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
// Limit the time between t0 and t1 if not scrubbing.
// Should the limiting be necessary in any play mode if there are no bugs?
if (Interactive())
absoluteTime = GetTrackTime();
else
#endif
absoluteTime = LimitTrackTime();
if (mCutPreviewGapLen > 0)
{
// msmeyer: We're in cut preview mode, so if we are on the right
// side of the gap, we jump over it.
if (absoluteTime > mCutPreviewGapStart)
absoluteTime += mCutPreviewGapLen;
}
return absoluteTime;
}
double AudioIOBase::GetStreamTime()
{
// Track time readout for the main thread
if( !IsStreamActive() )
return BAD_STREAM_TIME;
return mPlaybackSchedule.NormalizeTrackTime();
}
std::vector<long> AudioIOBase::GetSupportedPlaybackRates(int devIndex, double rate)
{
if (devIndex == -1)
{ // weren't given a device index, get the prefs / default one
devIndex = getPlayDevIndex();
}
// Check if we can use the cached rates
if (mCachedPlaybackIndex != -1 && devIndex == mCachedPlaybackIndex
&& (rate == 0.0 || make_iterator_range(mCachedPlaybackRates).contains(rate)))
{
return mCachedPlaybackRates;
}
std::vector<long> supported;
int irate = (int)rate;
const PaDeviceInfo* devInfo = NULL;
int i;
devInfo = Pa_GetDeviceInfo(devIndex);
if (!devInfo)
{
wxLogDebug(wxT("GetSupportedPlaybackRates() Could not get device info!"));
return supported;
}
// LLL: Remove when a proper method of determining actual supported
// DirectSound rate is devised.
const PaHostApiInfo* hostInfo = Pa_GetHostApiInfo(devInfo->hostApi);
bool isDirectSound = (hostInfo && hostInfo->type == paDirectSound);
PaStreamParameters pars;
pars.device = devIndex;
pars.channelCount = 1;
pars.sampleFormat = paFloat32;
pars.suggestedLatency = devInfo->defaultHighOutputLatency;
pars.hostApiSpecificStreamInfo = NULL;
// JKC: PortAudio Errors handled OK here. No need to report them
for (i = 0; i < NumRatesToTry; i++)
{
// LLL: Remove when a proper method of determining actual supported
// DirectSound rate is devised.
if (!(isDirectSound && RatesToTry[i] > 200000)){
if (Pa_IsFormatSupported(NULL, &pars, RatesToTry[i]) == 0)
supported.push_back(RatesToTry[i]);
Pa_Sleep( 10 );// There are ALSA drivers that don't like being probed
// too quickly.
}
}
if (irate != 0 && !make_iterator_range(supported).contains(irate))
{
// LLL: Remove when a proper method of determining actual supported
// DirectSound rate is devised.
if (!(isDirectSound && RatesToTry[i] > 200000))
if (Pa_IsFormatSupported(NULL, &pars, irate) == 0)
supported.push_back(irate);
}
return supported;
}
std::vector<long> AudioIOBase::GetSupportedCaptureRates(int devIndex, double rate)
{
if (devIndex == -1)
{ // not given a device, look up in prefs / default
devIndex = getRecordDevIndex();
}
// Check if we can use the cached rates
if (mCachedCaptureIndex != -1 && devIndex == mCachedCaptureIndex
&& (rate == 0.0 || make_iterator_range(mCachedCaptureRates).contains(rate)))
{
return mCachedCaptureRates;
}
std::vector<long> supported;
int irate = (int)rate;
const PaDeviceInfo* devInfo = NULL;
int i;
devInfo = Pa_GetDeviceInfo(devIndex);
if (!devInfo)
{
wxLogDebug(wxT("GetSupportedCaptureRates() Could not get device info!"));
return supported;
}
double latencyDuration = DEFAULT_LATENCY_DURATION;
long recordChannels = 1;
gPrefs->Read(wxT("/AudioIO/LatencyDuration"), &latencyDuration);
gPrefs->Read(wxT("/AudioIO/RecordChannels"), &recordChannels);
// LLL: Remove when a proper method of determining actual supported
// DirectSound rate is devised.
const PaHostApiInfo* hostInfo = Pa_GetHostApiInfo(devInfo->hostApi);
bool isDirectSound = (hostInfo && hostInfo->type == paDirectSound);
PaStreamParameters pars;
pars.device = devIndex;
pars.channelCount = recordChannels;
pars.sampleFormat = paFloat32;
pars.suggestedLatency = latencyDuration / 1000.0;
pars.hostApiSpecificStreamInfo = NULL;
for (i = 0; i < NumRatesToTry; i++)
{
// LLL: Remove when a proper method of determining actual supported
// DirectSound rate is devised.
if (!(isDirectSound && RatesToTry[i] > 200000))
{
if (Pa_IsFormatSupported(&pars, NULL, RatesToTry[i]) == 0)
supported.push_back(RatesToTry[i]);
Pa_Sleep( 10 );// There are ALSA drivers that don't like being probed
// too quickly.
}
}
if (irate != 0 && !make_iterator_range(supported).contains(irate))
{
// LLL: Remove when a proper method of determining actual supported
// DirectSound rate is devised.
if (!(isDirectSound && RatesToTry[i] > 200000))
if (Pa_IsFormatSupported(&pars, NULL, irate) == 0)
supported.push_back(irate);
}
return supported;
}
std::vector<long> AudioIOBase::GetSupportedSampleRates(
int playDevice, int recDevice, double rate)
{
// Not given device indices, look up prefs
if (playDevice == -1) {
playDevice = getPlayDevIndex();
}
if (recDevice == -1) {
recDevice = getRecordDevIndex();
}
// Check if we can use the cached rates
if (mCachedPlaybackIndex != -1 && mCachedCaptureIndex != -1 &&
playDevice == mCachedPlaybackIndex &&
recDevice == mCachedCaptureIndex &&
(rate == 0.0 || make_iterator_range(mCachedSampleRates).contains(rate)))
{
return mCachedSampleRates;
}
auto playback = GetSupportedPlaybackRates(playDevice, rate);
auto capture = GetSupportedCaptureRates(recDevice, rate);
int i;
// Return only sample rates which are in both arrays
std::vector<long> result;
for (i = 0; i < (int)playback.size(); i++)
if (make_iterator_range(capture).contains(playback[i]))
result.push_back(playback[i]);
// If this yields no results, use the default sample rates nevertheless
/* if (result.empty())
{
for (i = 0; i < NumStandardRates; i++)
result.push_back(StandardRates[i]);
}*/
return result;
}
/** \todo: should this take into account PortAudio's value for
* PaDeviceInfo::defaultSampleRate? In principal this should let us work out
* which rates are "real" and which resampled in the drivers, and so prefer
* the real rates. */
int AudioIOBase::GetOptimalSupportedSampleRate()
{
auto rates = GetSupportedSampleRates();
if (make_iterator_range(rates).contains(44100))
return 44100;
if (make_iterator_range(rates).contains(48000))
return 48000;
// if there are no supported rates, the next bit crashes. So check first,
// and give them a "sensible" value if there are no valid values. They
// will still get an error later, but with any luck may have changed
// something by then. It's no worse than having an invalid default rate
// stored in the preferences, which we don't check for
if (rates.empty()) return 44100;
return rates.back();
}
#if USE_PORTMIXER
int AudioIOBase::getRecordSourceIndex(PxMixer *portMixer)
{
int i;
wxString sourceName = gPrefs->Read(wxT("/AudioIO/RecordingSource"), wxT(""));
int numSources = Px_GetNumInputSources(portMixer);
for (i = 0; i < numSources; i++) {
if (sourceName == wxString(wxSafeConvertMB2WX(Px_GetInputSourceName(portMixer, i))))
return i;
}
return -1;
}
#endif
int AudioIOBase::getPlayDevIndex(const wxString &devNameArg)
{
wxString devName(devNameArg);
// if we don't get given a device, look up the preferences
if (devName.empty())
{
devName = gPrefs->Read(wxT("/AudioIO/PlaybackDevice"), wxT(""));
}
wxString hostName = gPrefs->Read(wxT("/AudioIO/Host"), wxT(""));
PaHostApiIndex hostCnt = Pa_GetHostApiCount();
PaHostApiIndex hostNum;
for (hostNum = 0; hostNum < hostCnt; hostNum++)
{
const PaHostApiInfo *hinfo = Pa_GetHostApiInfo(hostNum);
if (hinfo && wxString(wxSafeConvertMB2WX(hinfo->name)) == hostName)
{
for (PaDeviceIndex hostDevice = 0; hostDevice < hinfo->deviceCount; hostDevice++)
{
PaDeviceIndex deviceNum = Pa_HostApiDeviceIndexToDeviceIndex(hostNum, hostDevice);
const PaDeviceInfo *dinfo = Pa_GetDeviceInfo(deviceNum);
if (dinfo && DeviceName(dinfo) == devName && dinfo->maxOutputChannels > 0 )
{
// this device name matches the stored one, and works.
// So we say this is the answer and return it
return deviceNum;
}
}
// The device wasn't found so use the default for this host.
// LL: At this point, preferences and active no longer match.
return hinfo->defaultOutputDevice;
}
}
// The host wasn't found, so use the default output device.
// FIXME: TRAP_ERR PaErrorCode not handled well (this code is similar to input code
// and the input side has more comments.)
PaDeviceIndex deviceNum = Pa_GetDefaultOutputDevice();
// Sometimes PortAudio returns -1 if it cannot find a suitable default
// device, so we just use the first one available
//
// LL: At this point, preferences and active no longer match
//
// And I can't imagine how far we'll get specifying an "invalid" index later
// on...are we certain "0" even exists?
if (deviceNum < 0) {
wxASSERT(false);
deviceNum = 0;
}
return deviceNum;
}
int AudioIOBase::getRecordDevIndex(const wxString &devNameArg)
{
wxString devName(devNameArg);
// if we don't get given a device, look up the preferences
if (devName.empty())
{
devName = gPrefs->Read(wxT("/AudioIO/RecordingDevice"), wxT(""));
}
wxString hostName = gPrefs->Read(wxT("/AudioIO/Host"), wxT(""));
PaHostApiIndex hostCnt = Pa_GetHostApiCount();
PaHostApiIndex hostNum;
for (hostNum = 0; hostNum < hostCnt; hostNum++)
{
const PaHostApiInfo *hinfo = Pa_GetHostApiInfo(hostNum);
if (hinfo && wxString(wxSafeConvertMB2WX(hinfo->name)) == hostName)
{
for (PaDeviceIndex hostDevice = 0; hostDevice < hinfo->deviceCount; hostDevice++)
{
PaDeviceIndex deviceNum = Pa_HostApiDeviceIndexToDeviceIndex(hostNum, hostDevice);
const PaDeviceInfo *dinfo = Pa_GetDeviceInfo(deviceNum);
if (dinfo && DeviceName(dinfo) == devName && dinfo->maxInputChannels > 0 )
{
// this device name matches the stored one, and works.
// So we say this is the answer and return it
return deviceNum;
}
}
// The device wasn't found so use the default for this host.
// LL: At this point, preferences and active no longer match.
return hinfo->defaultInputDevice;
}
}
// The host wasn't found, so use the default input device.
// FIXME: TRAP_ERR PaErrorCode not handled well in getRecordDevIndex()
PaDeviceIndex deviceNum = Pa_GetDefaultInputDevice();
// Sometimes PortAudio returns -1 if it cannot find a suitable default
// device, so we just use the first one available
// PortAudio has an error reporting function. We should log/report the error?
//
// LL: At this point, preferences and active no longer match
//
// And I can't imagine how far we'll get specifying an "invalid" index later
// on...are we certain "0" even exists?
if (deviceNum < 0) {
// JKC: This ASSERT will happen if you run with no config file
// This happens once. Config file will exist on the next run.
// TODO: Look into this a bit more. Could be relevant to blank Device Toolbar.
wxASSERT(false);
deviceNum = 0;
}
return deviceNum;
}
wxString AudioIOBase::GetDeviceInfo()
{
wxStringOutputStream o;
wxTextOutputStream s(o, wxEOL_UNIX);
if (IsStreamActive()) {
return XO("Stream is active ... unable to gather information.\n")
.Translation();
}
// FIXME: TRAP_ERR PaErrorCode not handled. 3 instances in GetDeviceInfo().
int recDeviceNum = Pa_GetDefaultInputDevice();
int playDeviceNum = Pa_GetDefaultOutputDevice();
int cnt = Pa_GetDeviceCount();
// PRL: why only into the log?
wxLogDebug(wxT("Portaudio reports %d audio devices"),cnt);
s << wxT("==============================\n");
s << XO("Default recording device number: %d\n").Format( recDeviceNum );
s << XO("Default playback device number: %d\n").Format( playDeviceNum);
wxString recDevice = gPrefs->Read(wxT("/AudioIO/RecordingDevice"), wxT(""));
wxString playDevice = gPrefs->Read(wxT("/AudioIO/PlaybackDevice"), wxT(""));
int j;
// This gets info on all available audio devices (input and output)
if (cnt <= 0) {
s << XO("No devices found\n");
return o.GetString();
}
const PaDeviceInfo* info;
for (j = 0; j < cnt; j++) {
s << wxT("==============================\n");
info = Pa_GetDeviceInfo(j);
if (!info) {
s << XO("Device info unavailable for: %d\n").Format( j );
continue;
}
wxString name = DeviceName(info);
s << XO("Device ID: %d\n").Format( j );
s << XO("Device name: %s\n").Format( name );
s << XO("Host name: %s\n").Format( HostName(info) );
s << XO("Recording channels: %d\n").Format( info->maxInputChannels );
s << XO("Playback channels: %d\n").Format( info->maxOutputChannels );
s << XO("Low Recording Latency: %g\n").Format( info->defaultLowInputLatency );
s << XO("Low Playback Latency: %g\n").Format( info->defaultLowOutputLatency );
s << XO("High Recording Latency: %g\n").Format( info->defaultHighInputLatency );
s << XO("High Playback Latency: %g\n").Format( info->defaultHighOutputLatency );
auto rates = GetSupportedPlaybackRates(j, 0.0);
/* i18n-hint: Supported, meaning made available by the system */
s << XO("Supported Rates:\n");
for (int k = 0; k < (int) rates.size(); k++) {
s << wxT(" ") << (int)rates[k] << wxT("\n");
}
if (name == playDevice && info->maxOutputChannels > 0)
playDeviceNum = j;
if (name == recDevice && info->maxInputChannels > 0)
recDeviceNum = j;
// Sometimes PortAudio returns -1 if it cannot find a suitable default
// device, so we just use the first one available
if (recDeviceNum < 0 && info->maxInputChannels > 0){
recDeviceNum = j;
}
if (playDeviceNum < 0 && info->maxOutputChannels > 0){
playDeviceNum = j;
}
}
bool haveRecDevice = (recDeviceNum >= 0);
bool havePlayDevice = (playDeviceNum >= 0);
s << wxT("==============================\n");
if (haveRecDevice)
s << XO("Selected recording device: %d - %s\n").Format( recDeviceNum, recDevice );
else
s << XO("No recording device found for '%s'.\n").Format( recDevice );
if (havePlayDevice)
s << XO("Selected playback device: %d - %s\n").Format( playDeviceNum, playDevice );
else
s << XO("No playback device found for '%s'.\n").Format( playDevice );
std::vector<long> supportedSampleRates;
if (havePlayDevice && haveRecDevice) {
supportedSampleRates = GetSupportedSampleRates(playDeviceNum, recDeviceNum);
s << XO("Supported Rates:\n");
for (int k = 0; k < (int) supportedSampleRates.size(); k++) {
s << wxT(" ") << (int)supportedSampleRates[k] << wxT("\n");
}
}
else {
s << XO("Cannot check mutual sample rates without both devices.\n");
return o.GetString();
}
#if defined(USE_PORTMIXER)
if (supportedSampleRates.size() > 0)
{
int highestSampleRate = supportedSampleRates.back();
bool EmulateMixerInputVol = true;
bool EmulateMixerOutputVol = true;
float MixerInputVol = 1.0;
float MixerOutputVol = 1.0;
int error;
PaStream *stream;
PaStreamParameters playbackParameters;
playbackParameters.device = playDeviceNum;
playbackParameters.sampleFormat = paFloat32;
playbackParameters.hostApiSpecificStreamInfo = NULL;
playbackParameters.channelCount = 1;
if (Pa_GetDeviceInfo(playDeviceNum)){
playbackParameters.suggestedLatency =
Pa_GetDeviceInfo(playDeviceNum)->defaultLowOutputLatency;
}
else{
playbackParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0;
}
PaStreamParameters captureParameters;
captureParameters.device = recDeviceNum;
captureParameters.sampleFormat = paFloat32;;
captureParameters.hostApiSpecificStreamInfo = NULL;
captureParameters.channelCount = 1;
if (Pa_GetDeviceInfo(recDeviceNum)){
captureParameters.suggestedLatency =
Pa_GetDeviceInfo(recDeviceNum)->defaultLowInputLatency;
}else{
captureParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0;
}
// Not really doing I/O so pass nullptr for the callback function
error = Pa_OpenStream(&stream,
&captureParameters, &playbackParameters,
highestSampleRate, paFramesPerBufferUnspecified,
paClipOff | paDitherOff,
nullptr, NULL);
if (error) {
error = Pa_OpenStream(&stream,
&captureParameters, NULL,
highestSampleRate, paFramesPerBufferUnspecified,
paClipOff | paDitherOff,
nullptr, NULL);
}
if (error) {
s << XO("Received %d while opening devices\n").Format( error );
return o.GetString();
}
PxMixer *PortMixer = Px_OpenMixer(stream, 0);
if (!PortMixer) {
s << XO("Unable to open Portmixer\n");
Pa_CloseStream(stream);
return o.GetString();
}
s << wxT("==============================\n");
s << XO("Available mixers:\n");
// FIXME: ? PortMixer errors on query not reported in GetDeviceInfo
cnt = Px_GetNumMixers(stream);
for (int i = 0; i < cnt; i++) {
wxString name = wxSafeConvertMB2WX(Px_GetMixerName(stream, i));
s << XO("%d - %s\n").Format( i, name );
}
s << wxT("==============================\n");
s << XO("Available recording sources:\n");
cnt = Px_GetNumInputSources(PortMixer);
for (int i = 0; i < cnt; i++) {
wxString name = wxSafeConvertMB2WX(Px_GetInputSourceName(PortMixer, i));
s << XO("%d - %s\n").Format( i, name );
}
s << wxT("==============================\n");
s << XO("Available playback volumes:\n");
cnt = Px_GetNumOutputVolumes(PortMixer);
for (int i = 0; i < cnt; i++) {
wxString name = wxSafeConvertMB2WX(Px_GetOutputVolumeName(PortMixer, i));
s << XO("%d - %s\n").Format( i, name );
}
// Determine mixer capabilities - if it doesn't support either
// input or output, we emulate them (by multiplying this value
// by all incoming/outgoing samples)
MixerOutputVol = Px_GetPCMOutputVolume(PortMixer);
EmulateMixerOutputVol = false;
Px_SetPCMOutputVolume(PortMixer, 0.0);
if (Px_GetPCMOutputVolume(PortMixer) > 0.1)
EmulateMixerOutputVol = true;
Px_SetPCMOutputVolume(PortMixer, 0.2f);
if (Px_GetPCMOutputVolume(PortMixer) < 0.1 ||
Px_GetPCMOutputVolume(PortMixer) > 0.3)
EmulateMixerOutputVol = true;
Px_SetPCMOutputVolume(PortMixer, MixerOutputVol);
MixerInputVol = Px_GetInputVolume(PortMixer);
EmulateMixerInputVol = false;
Px_SetInputVolume(PortMixer, 0.0);
if (Px_GetInputVolume(PortMixer) > 0.1)
EmulateMixerInputVol = true;
Px_SetInputVolume(PortMixer, 0.2f);
if (Px_GetInputVolume(PortMixer) < 0.1 ||
Px_GetInputVolume(PortMixer) > 0.3)
EmulateMixerInputVol = true;
Px_SetInputVolume(PortMixer, MixerInputVol);
Pa_CloseStream(stream);
s << wxT("==============================\n");
s << ( EmulateMixerInputVol
? XO("Recording volume is emulated\n")
: XO("Recording volume is native\n") );
s << ( EmulateMixerOutputVol
? XO("Playback volume is emulated\n")
: XO("Playback volume is native\n") );
Px_CloseMixer(PortMixer);
} //end of massive if statement if a valid sample rate has been found
#endif
return o.GetString();
}
#ifdef EXPERIMENTAL_MIDI_OUT
// FIXME: When EXPERIMENTAL_MIDI_IN is added (eventually) this should also be enabled -- Poke
wxString AudioIOBase::GetMidiDeviceInfo()
{
wxStringOutputStream o;
wxTextOutputStream s(o, wxEOL_UNIX);
if (IsStreamActive()) {
return XO("Stream is active ... unable to gather information.\n")
.Translation();
}
// XXX: May need to trap errors as with the normal device info
int recDeviceNum = Pm_GetDefaultInputDeviceID();
int playDeviceNum = Pm_GetDefaultOutputDeviceID();
int cnt = Pm_CountDevices();
// PRL: why only into the log?
wxLogDebug(wxT("PortMidi reports %d MIDI devices"), cnt);
s << wxT("==============================\n");
s << XO("Default recording device number: %d\n").Format( recDeviceNum );
s << XO("Default playback device number: %d\n").Format( playDeviceNum );
wxString recDevice = gPrefs->Read(wxT("/MidiIO/RecordingDevice"), wxT(""));
wxString playDevice = gPrefs->Read(wxT("/MidiIO/PlaybackDevice"), wxT(""));
// This gets info on all available audio devices (input and output)
if (cnt <= 0) {
s << XO("No devices found\n");
return o.GetString();
}
for (int i = 0; i < cnt; i++) {
s << wxT("==============================\n");
const PmDeviceInfo* info = Pm_GetDeviceInfo(i);
if (!info) {
s << XO("Device info unavailable for: %d\n").Format( i );
continue;
}
wxString name = wxSafeConvertMB2WX(info->name);
wxString hostName = wxSafeConvertMB2WX(info->interf);
s << XO("Device ID: %d\n").Format( i );
s << XO("Device name: %s\n").Format( name );
s << XO("Host name: %s\n").Format( hostName );
/* i18n-hint: Supported, meaning made available by the system */
s << XO("Supports output: %d\n").Format( info->output );
/* i18n-hint: Supported, meaning made available by the system */
s << XO("Supports input: %d\n").Format( info->input );
s << XO("Opened: %d\n").Format( info->opened );
if (name == playDevice && info->output)
playDeviceNum = i;
if (name == recDevice && info->input)
recDeviceNum = i;
// XXX: This is only done because the same was applied with PortAudio
// If PortMidi returns -1 for the default device, use the first one
if (recDeviceNum < 0 && info->input){
recDeviceNum = i;
}
if (playDeviceNum < 0 && info->output){
playDeviceNum = i;
}
}
bool haveRecDevice = (recDeviceNum >= 0);
bool havePlayDevice = (playDeviceNum >= 0);
s << wxT("==============================\n");
if (haveRecDevice)
s << XO("Selected MIDI recording device: %d - %s\n").Format( recDeviceNum, recDevice );
else
s << XO("No MIDI recording device found for '%s'.\n").Format( recDevice );
if (havePlayDevice)
s << XO("Selected MIDI playback device: %d - %s\n").Format( playDeviceNum, playDevice );
else
s << XO("No MIDI playback device found for '%s'.\n").Format( playDevice );
// Mention our conditional compilation flags for Alpha only
#ifdef IS_ALPHA
// Not internationalizing these alpha-only messages
s << wxT("==============================\n");
#ifdef EXPERIMENTAL_MIDI_OUT
s << wxT("EXPERIMENTAL_MIDI_OUT is enabled\n");
#else
s << wxT("EXPERIMENTAL_MIDI_OUT is NOT enabled\n");
#endif
#ifdef EXPERIMENTAL_MIDI_IN
s << wxT("EXPERIMENTAL_MIDI_IN is enabled\n");
#else
s << wxT("EXPERIMENTAL_MIDI_IN is NOT enabled\n");
#endif
#endif
return o.GetString();
}
#endif
bool AudioIOBase::PlaybackSchedule::PassIsComplete() const
{
// Test mTime within the PortAudio callback
if (Scrubbing())
return false; // but may be true if playing at speed
return Overruns( GetTrackTime() );
}
bool AudioIOBase::PlaybackSchedule::Overruns( double trackTime ) const
{
return (ReversedTime() ? trackTime <= mT1 : trackTime >= mT1);
}
namespace
{
/** @brief Compute the duration (in seconds at playback) of the specified region of the track.
*
* Takes a region of the time track (specified by the unwarped time points in the project), and
* calculates how long it will actually take to play this region back, taking the time track's
* warping effects into account.
* @param t0 unwarped time to start calculation from
* @param t1 unwarped time to stop calculation at
* @return the warped duration in seconds
*/
double ComputeWarpedLength(const Envelope &env, double t0, double t1)
{
return env.IntegralOfInverse(t0, t1);
}
/** @brief Compute how much unwarped time must have elapsed if length seconds of warped time has
* elapsed
*
* @param t0 The unwarped time (seconds from project start) at which to start
* @param length How many seconds of warped time went past.
* @return The end point (in seconds from project start) as unwarped time
*/
double SolveWarpedLength(const Envelope &env, double t0, double length)
{
return env.SolveIntegralOfInverse(t0, length);
}
}
double AudioIOBase::PlaybackSchedule::AdvancedTrackTime(
double time, double realElapsed, double speed ) const
{
if (ReversedTime())
realElapsed *= -1.0;
// Defense against cases that might cause loops not to terminate
if ( fabs(mT0 - mT1) < 1e-9 )
return mT0;
if (mEnvelope) {
wxASSERT( speed == 1.0 );
double total=0.0;
bool foundTotal = false;
do {
auto oldTime = time;
if (foundTotal && fabs(realElapsed) > fabs(total))
// Avoid SolveWarpedLength
time = mT1;
else
time = SolveWarpedLength(*mEnvelope, time, realElapsed);
if (!Looping() || !Overruns( time ))
break;
// Bug1922: The part of the time track outside the loop should not
// influence the result
double delta;
if (foundTotal && oldTime == mT0)
// Avoid integrating again
delta = total;
else {
delta = ComputeWarpedLength(*mEnvelope, oldTime, mT1);
if (oldTime == mT0)
foundTotal = true, total = delta;
}
realElapsed -= delta;
time = mT0;
} while ( true );
}
else {
time += realElapsed * fabs(speed);
// Wrap to start if looping
if (Looping()) {
while ( Overruns( time ) ) {
// LL: This is not exactly right, but I'm at my wits end trying to
// figure it out. Feel free to fix it. :-)
// MB: it's much easier than you think, mTime isn't warped at all!
time -= mT1 - mT0;
}
}
}
return time;
}
void AudioIOBase::PlaybackSchedule::TrackTimeUpdate(double realElapsed)
{
// Update mTime within the PortAudio callback
if (Interactive())
return;
auto time = GetTrackTime();
auto newTime = AdvancedTrackTime( time, realElapsed, 1.0 );
SetTrackTime( newTime );
}
double AudioIOBase::PlaybackSchedule::TrackDuration(double realElapsed) const
{
if (mEnvelope)
return SolveWarpedLength(*mEnvelope, mT0, realElapsed);
else
return realElapsed;
}
double AudioIOBase::PlaybackSchedule::RealDuration(double trackTime1) const
{
double duration;
if (mEnvelope)
duration = ComputeWarpedLength(*mEnvelope, mT0, trackTime1);
else
duration = trackTime1 - mT0;
return fabs(duration);
}
double AudioIOBase::PlaybackSchedule::RealTimeRemaining() const
{
return mWarpedLength - mWarpedTime;
}
void AudioIOBase::PlaybackSchedule::RealTimeAdvance( double increment )
{
mWarpedTime += increment;
}
void AudioIOBase::PlaybackSchedule::RealTimeInit( double trackTime )
{
if (Scrubbing())
mWarpedTime = 0.0;
else
mWarpedTime = RealDuration( trackTime );
}
void AudioIOBase::PlaybackSchedule::RealTimeRestart()
{
mWarpedTime = 0;
}
double AudioIOBase::RecordingSchedule::ToConsume() const
{
return mDuration - Consumed();
}
double AudioIOBase::RecordingSchedule::Consumed() const
{
return std::max( 0.0, mPosition + TotalCorrection() );
}
double AudioIOBase::RecordingSchedule::ToDiscard() const
{
return std::max(0.0, -( mPosition + TotalCorrection() ) );
}
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