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Roger's fixes for MIDI timing on Alsa, as adapted by Paul Licameli

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Paul Licameli 2017-09-25 03:03:21 -04:00
commit f2d9ff59bb
2 changed files with 465 additions and 135 deletions
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541
src/AudioIO.cpp
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@ -81,6 +81,78 @@
speed at mTime. This effectively integrates speed to get position.
Negative speeds are allowed too, for instance in scrubbing.
\par The Big Picture
@verbatim
Sample
Time (in seconds, = total_sample_count / sample_rate)
^
| / /
| y=x-mSystemTimeMinusAudioTime / /
| / # /
| / /
| / # <- callbacks (#) showing
| /# / lots of timing jitter.
| top line is "full buffer" / / Some are later,
| condition / / indicating buffer is
| / / getting low. Plot
| / # / shows sample time
| / # / (based on how many
| / # / samples previously
| / / *written*) vs. real
| / # / time.
| /<------->/ audio latency
| /# v/
| / / bottom line is "empty buffer"
| / # / condition = DAC output time =
| / /
| / # <-- rapid callbacks as buffer is filled
| / /
0 +...+---------#---------------------------------------------------->
0 ^ | | real time
| | first callback time
| mSystemMinusAudioTime
|
Probably the actual real times shown in this graph are very large
in practice (> 350,000 sec.), so the X "origin" might be when
the computer was booted or 1970 or something.
@endverbatim
To estimate the true DAC time (needed to synchronize MIDI), we need
a mapping from track time to DAC time. The estimate is the theoretical
time of the full buffer (top diagonal line) + audio latency. To
estimate the top diagonal line, we "draw" the line to be at least
as high as any sample time corresponding to a callback (#), and we
slowly lower the line in case the sample clock is slow or the system
clock is fast, preventing the estimated line from drifting too far
from the actual callback observations. The line is occasionally
"bumped" up by new callback observations, but continuously
"lowered" at a very low rate. All adjustment is accomplished
by changing mSystemMinusAudioTime, shown here as the X-intercept.\n
theoreticalFullBufferTime = realTime - mSystemMinusAudioTime\n
To estimate audio latency, notice that the first callback happens on
an empty buffer, but the buffer soon fills up. This will cause a rapid
re-estimation of mSystemMinusAudioTime. (The first estimate of
mSystemMinusAudioTime will simply be the real time of the first
callback time.) By watching these changes, which happen within ms of
starting, we can estimate the buffer size and thus audio latency.
So, to map from track time to real time, we compute:\n
DACoutputTime = trackTime + mSystemMinusAudioTime\n
There are some additional details to avoid counting samples while
paused or while waiting for initialization, MIDI latency, etc.
Also, in the code, track time is measured with respect to the track
origin, so there's an extra term to add (mT0) if you start somewhere
in the middle of the track.
Finally, when a callback occurs, you might expect there is room in
the output buffer for the requested frames, so maybe the "full buffer"
sample time should be based not on the first sample of the callback, but
the last sample time + 1 sample. I suspect, at least on Linux, that the
callback occurs as soon as the last callback completes, so the buffer is
really full, and the callback thread is going to block waiting for space
in the output buffer.
\par Midi Time
MIDI is not warped according to the speed control. This might be
something that should be changed. (Editorial note: Wouldn't it
@ -95,33 +167,61 @@
\par
Therefore, we define the following interface for MIDI timing:
\li \c AudioTime() is the time based on all samples written so far, including zeros output during pauses. AudioTime() is based on the start location mT0, not zero.
\li \c PauseTime() is the amount of time spent paused, based on a count of zero samples output.
\li \c MidiTime() is an estimate in milliseconds of the current audio output time + 1s. In other words, what audacity track time corresponds to the audio (including pause insertions) at the output?
\li \c PauseTime() is the amount of time spent paused, based on a count of zero-padding samples output.
\li \c MidiTime() is an estimate in milliseconds of the current audio output time + 1s. In other words, what audacity track time corresponds to the audio (plus pause insertions) at the DAC output?
\par AudioTime() and PauseTime() computation
AudioTime() is simply mT0 + mNumFrames / mRate.
mNumFrames is incremented in each audio callback. Similarly, PauseTime()
is mNumPauseFrames / mRate. mNumPauseFrames is also incremented in
each audio callback when a pause is in effect.
each audio callback when a pause is in effect or audio output is ready to start.
\par MidiTime() computation
MidiTime() is computed based on information from PortAudio's callback,
which estimates the system time at which the current audio buffer will
be output. Consider the (unimplemented) function RealToTrack() that
maps real time to track time. If outputTime is PortAudio's time
estimate for the most recent output buffer, then \n
RealToTrack(outputTime) = AudioTime() - PauseTime() - bufferDuration \n
We want to know RealToTrack of the current time, so we use this
approximation for small d: \n
RealToTrack(t + d) = RealToTrack(t) + d \n
Letting t = outputTime and d = (systemTime - outputTime), we can
maps real audio write time to track time. If writeTime is the system
time for the first sample of the current output buffer, and
if we are in the callback, so AudioTime() also refers to the first sample
of the buffer, then \n
RealToTrack(writeTime) = AudioTime() - PauseTime()\n
We want to know RealToTrack of the current time (when we are not in the
callback, so we use this approximation for small d: \n
RealToTrack(t + d) = RealToTrack(t) + d, or \n
Letting t = writeTime and d = (systemTime - writeTime), we can
substitute to get:\n
RealToTrack(systemTime) = AudioTime() - PauseTime() - bufferduration + (systemTime - outputTime) \n
MidiTime() should include pause time, so add PauseTime() to both sides of
the equation. Also MidiTime() is offset by 1 second to avoid negative
time at startup, so add 1 to both sides:
MidiTime() in seconds = RealToTrack(systemTime) + PauseTime() + 1 = \n
AudioTime() - bufferduration + (systemTime - outputTime) + 1
RealToTrack(systemTime)
= RealToTrack(writeTime) + systemTime - writeTime\n
= AudioTime() - PauseTime() + (systemTime - writeTime) \n
MidiTime() should include pause time, so that it increases smoothly,
and audioLatency so that MidiTime() corresponds to the time of audio
output rather than audio write times. Also MidiTime() is offset by 1
second to avoid negative time at startup, so add 1: \n
MidiTime(systemTime) in seconds\n
= RealToTrack(systemTime) + PauseTime() - audioLatency + 1 \n
= AudioTime() + (systemTime - writeTime) - audioLatency + 1 \n
(Note that audioLatency is called mAudioOutLatency in the code.)
When we schedule a MIDI event with track time TT, we need
to map TT to a PortMidi timestamp. The PortMidi timestamp is exactly
MidiTime(systemTime) in ms units, and \n
MidiTime(x) = RealToTrack(x) + PauseTime() + 1, so \n
timestamp = TT + PauseTime() + 1 - midiLatency \n
Note 1: The timestamp is incremented by the PortMidi stream latency
(midiLatency) so we subtract midiLatency here for the timestamp
passed to PortMidi. \n
Note 2: Here, we're setting x to the time at which RealToTrack(x) = TT,
so then MidiTime(x) is the desired timestamp. To be completely
correct, we should assume that MidiTime(x + d) = MidiTime(x) + d,
and consider that we compute MidiTime(systemTime) based on the
*current* system time, but we really want the MidiTime(x) for some
future time corresponding when MidiTime(x) = TT.)
\par
Also, we should assume PortMidi was opened with mMidiLatency, and that
MIDI messages become sound with a delay of mSynthLatency. Therefore,
the final timestamp calculation is: \n
timestamp = TT + PauseTime() + 1 - (mMidiLatency + mSynthLatency) \n
(All units here are seconds; some conversion is needed in the code.)
\par
The difference AudioTime() - PauseTime() is the time "cursor" for
@ -129,34 +229,92 @@
unsynchronized. In particular, MIDI will not be synchronized with
the visual cursor, which moves with scaled time reported in mTime.
\par Midi Synchronization
The goal of MIDI playback is to deliver MIDI messages synchronized to
audio (assuming no speed variation for now). If a midi event has time
tmidi, then the timestamp for that message should be \n
timestamp (in seconds) = tmidi + PauseTime() + 1.0 - latency.\n
(This is actually off by 1ms; see "PortMidi Latency Parameter" below for
more detail.)
Notice the extra 1.0, added because MidiTime() is offset by 1s to avoid
starting at a negative value. Also notice that we subtract latency.
The user must set device latency using preferences. Some software
synthesizers have very high latency (on the order of 100ms), so unless
we lower timestamps and send messages early, the final output will not
be synchronized.
This timestamp is interpreted by PortMidi relative to MidiTime(), which
is synchronized to audio output. So the only thing we need to do is
output Midi messages shortly before they will be played with the correct
timestamp. We will take "shortly before" to mean "at about the same time
as corresponding audio". Based on this, output the event when
AudioTime() - PauseTime() > mtime - latency,
adjusting the event time by adding PauseTime() + 1 - latency.
This gives at least mAudioOutputLatency for
the MIDI output to be generated (we want to generate MIDI output before
the actual output time because events generated early are accurately timed
according to their timestamp). However, the MIDI thread sleeps for
MIDI_SLEEP in its polling loop, so the worst case is really
mAudioOutputLatency + MIDI_SLEEP. In case the audio output latency is
very low, we will output events when
AudioTime() + MIDI_SLEEP - PauseTime() > mtime - latency.
\par Timing in Linux
It seems we cannot get much info from Linux. We can read the time
when we get a callback, and we get a variable frame count (it changes
from one callback to the next). Returning to the RealToTrack()
equations above: \n
RealToTrack(outputTime) = AudioTime() - PauseTime() - bufferDuration \n
where outputTime should be PortAudio's estimate for the most recent output
buffer, but at least on my Dell Latitude E7450, PortAudio is getting zero
from ALSA, so we need to find a proxy for this.
\par Estimating outputTime (Plan A, assuming double-buffered, fixed-size buffers, please skip to Plan B)
One can expect the audio callback to happen as soon as there is room in
the output for another block of samples, so we could just measure system
time at the top of the callback. Then we could add the maximum delay
buffered in the system. E.g. if there is simple double buffering and the
callback is computing one of the buffers, the callback happens just as
one of the buffers empties, meaning the other buffer is full, so we have
exactly one buffer delay before the next computed sample is output.
If computation falls behind a bit, the callback will be later, so the
delay to play the next computed sample will be less. I think a reasonable
way to estimate the actual output time is to assume that the computer is
mostly keeping up and that *most* callbacks will occur immediately when
there is space. Note that the most likely reason for the high-priority
audio thread to fall behind is the callback itself, but the start of the
callback should be pretty consistently keeping up.
Also, we do not have to have a perfect estimate of the time. Suppose we
estimate a linear mapping from sample count to system time by saying
that the sample count maps to the system time at the most recent callback,
and set the slope to 1% slower than real time (as if the sample clock is
slow). Now, at each callback, if the callback seems to occur earlier than
expected, we can adjust the mapping to be earlier. The earlier the
callback, the more accurate it must be. On the other hand, if the callback
is later than predicted, it must be a delayed callback (or else the
sample clock is more than 1% slow, which is really a hardware problem.)
How bad can this be? Assuming callbacks every 30ms (this seems to be what
I'm observing in a default setup), you'll be a maximum of 1ms off even if
2 out of 3 callbacks are late. This is pretty reasonable given that
PortMIDI clock precision is 1ms. If buffers are larger and callback timing
is more erratic, errors will be larger, but even a few ms error is
probably OK.
\par Estimating outputTime (Plan B, variable framesPerBuffer in callback, please skip to Plan C)
ALSA is complicated because we get varying values of
framesPerBuffer from callback to callback. Assume you get more frames
when the callback is later (because there is more accumulated input to
deliver and more more accumulated room in the output buffers). So take
the current time and subtract the duration of the frame count in the
current callback. This should be a time position that is relatively
jitter free (because we estimated the lateness by frame count and
subtracted that out). This time position intuitively represents the
current ADC time, or if no input, the time of the tail of the output
buffer. If we wanted DAC time, we'd have to add the total output
buffer duration, which should be reported by PortAudio. (If PortAudio
is wrong, we'll be systematically shifted in time by the error.)
Since there is still bound to be jitter, we can smooth these estimates.
First, we will assume a linear mapping from system time to audio time
with slope = 1, so really it's just the offset we need, which is going
to be a double that we can read/write atomically without locks or
anything fancy. (Maybe it should be "volatile".)
To improve the estimate, we get a new offset every callback, so we can
create a "smooth" offset by using a simple regression model (also
this could be seen as a first order filter). The following formula
updates smooth_offset with a new offset estimate in the callback:
smooth_offset = smooth_offset * 0.9 + new_offset_estimate * 0.1
Since this is smooth, we'll have to be careful to give it a good initial
value to avoid a long convergence.
\par Estimating outputTime (Plan C)
ALSA is complicated because we get varying values of
framesPerBuffer from callback to callback. It seems there is a lot
of variation in callback times and buffer space. One solution would
be to go to fixed size double buffer, but Audacity seems to work
better as is, so Plan C is to rely on one invariant which is that
the output buffer cannot overflow, so there's a limit to how far
ahead of the DAC time we can be writing samples into the
buffer. Therefore, we'll assume that the audio clock runs slow by
about 0.2% and we'll assume we're computing at that rate. If the
actual output position is ever ahead of the computed position, we'll
increase the computed position to the actual position. Thus whenever
the buffer is less than near full, we'll stay ahead of DAC time,
falling back at a rate of about 0.2% until eventually there's
another near-full buffer callback that will push the time back ahead.
\par Interaction between MIDI, Audio, and Pause
When Pause is used, PauseTime() will increase at the same rate as
@ -283,6 +441,14 @@ writing audio.
#ifdef EXPERIMENTAL_MIDI_OUT
#define MIDI_SLEEP 10 /* milliseconds */
// how long do we think the thread that fills MIDI buffers,
// if it is separate from the portaudio thread,
// might be delayed due to other threads?
#ifdef USE_MIDI_THREAD
#define THREAD_LATENCY 10 /* milliseconds */
#else
#define THREAD_LATENCY 0 /* milliseconds */
#endif
#define ROUND(x) (int) ((x)+0.5)
//#include <string.h>
#include "../lib-src/portmidi/pm_common/portmidi.h"
@ -797,6 +963,26 @@ private:
};
#endif
// return the system time as a double
static double streamStartTime = 0; // bias system time to small number
static double SystemTime(bool usingAlsa)
{
#ifdef __WXGTK__
if (usingAlsa) {
struct timespec now;
// CLOCK_MONOTONIC_RAW is unaffected by NTP or adj-time
clock_gettime(CLOCK_MONOTONIC_RAW, &now);
//return now.tv_sec + now.tv_nsec * 0.000000001;
return (now.tv_sec + now.tv_nsec * 0.000000001) - streamStartTime;
}
#else
WXUNUSED(usingAlsa);
#endif
return PaUtil_GetTime() - streamStartTime;
}
const int AudioIO::StandardRates[] = {
8000,
11025,
@ -1444,6 +1630,17 @@ bool AudioIO::StartPortAudioStream(double sampleRate,
#ifdef EXPERIMENTAL_MIDI_OUT
mNumFrames = 0;
mNumPauseFrames = 0;
// we want this initial value to be way high. It should be
// sufficient to assume AudioTime is zero and therefore
// mSystemMinusAudioTime is SystemTime(), but we'll add 1000s
// for good measure. On the first callback, this should be
// reduced to SystemTime() - mT0, and note that mT0 is always
// positive.
mSystemMinusAudioTimePlusLatency =
mSystemMinusAudioTime = SystemTime(mUsingAlsa) + 1000;
mAudioOutLatency = 0.0; // set when stream is opened
mCallbackCount = 0;
mAudioFramesPerBuffer = 0;
#endif
mOwningProject = GetActiveProject();
mInputMeter = NULL;
@ -1554,9 +1751,6 @@ bool AudioIO::StartPortAudioStream(double sampleRate,
int userData = 24;
int* lpUserData = (captureFormat_saved == int24Sample) ? &userData : NULL;
#ifndef USE_TIME_INFO
mMidiTimeCorrection = 0;
#endif
mLastPaError = Pa_OpenStream( &mPortStreamV19,
useCapture ? &captureParameters : NULL,
usePlayback ? &playbackParameters : NULL,
@ -1571,13 +1765,6 @@ bool AudioIO::StartPortAudioStream(double sampleRate,
#endif
if (mPortStreamV19 != NULL && mLastPaError == paNoError) {
#ifndef USE_TIME_INFO
auto info = Pa_GetStreamInfo(mPortStreamV19);
if (info)
mMidiTimeCorrection =
info->outputLatency - (MIDI_MINIMAL_LATENCY_MS / 1000.0);
#endif
#ifdef __WXMAC__
if (mPortMixer) {
if (Px_SupportsPlaythrough(mPortMixer)) {
@ -1597,6 +1784,15 @@ bool AudioIO::StartPortAudioStream(double sampleRate,
}
#endif
// We use audio latency to estimate how far ahead of DACS we are writing
if (mPortStreamV19 != NULL && mLastPaError == paNoError) {
const PaStreamInfo* info = Pa_GetStreamInfo(mPortStreamV19);
// this is an initial guess, but for PA/Linux/ALSA it's wrong and will be
// updated with a better value:
mAudioOutLatency = info->outputLatency;
mSystemMinusAudioTimePlusLatency += mAudioOutLatency;
}
return (mLastPaError == paNoError);
}
@ -1618,6 +1814,7 @@ void AudioIO::StartMonitoring(double sampleRate)
// FIXME: TRAP_ERR StartPortAudioStream (a PaError may be present)
// but StartPortAudioStream function only returns true or false.
mUsingAlsa = false;
success = StartPortAudioStream(sampleRate, (unsigned int)playbackChannels,
(unsigned int)captureChannels,
captureFormat);
@ -1678,6 +1875,12 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
wxMilliSleep( 50 );
}
#ifdef __WXGTK__
// Detect whether ALSA is the chosen host, and do the various involved MIDI
// timing compensations only then.
mUsingAlsa = (gPrefs->Read(wxT("/AudioIO/Host"), wxT("")) == "ALSA");
#endif
gPrefs->Read(wxT("/AudioIO/SWPlaythrough"), &mSoftwarePlaythrough, false);
gPrefs->Read(wxT("/AudioIO/SoundActivatedRecord"), &mPauseRec, false);
int silenceLevelDB;
@ -1716,6 +1919,9 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
double playbackTime = 4.0;
streamStartTime = 0;
streamStartTime = SystemTime(mUsingAlsa);
#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
bool scrubbing = (options.pScrubbingOptions != nullptr);
@ -2033,18 +2239,35 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
// (Which we should be able to determine from fields of
// PaStreamCallbackTimeInfo, but that seems not to work as documented with
// ALSA.)
wxString hostName = gPrefs->Read(wxT("/AudioIO/Host"), wxT(""));
if (hostName == "ALSA")
if (mUsingAlsa)
// Perhaps we should do this only if also playing MIDI ?
PaAlsa_EnableRealtimeScheduling( mPortStreamV19, 1 );
#endif
//
// Generate a unique value each time, to be returned to
// clients accessing the AudioIO API, so they can query if they
// are the ones who have reserved AudioIO or not.
//
// It is important to set this before setting the portaudio stream in
// motion -- otherwise it may play an unspecified number of leading
// zeroes.
mStreamToken = (++mNextStreamToken);
// This affects the AudioThread (not the portaudio callback).
// Probably not needed so urgently before portaudio thread start for usual
// playback, since our ring buffers have been primed already with 4 sec
// of audio, but then we might be scrubbing, so do it.
mAudioThreadFillBuffersLoopRunning = true;
// Now start the PortAudio stream!
PaError err;
err = Pa_StartStream( mPortStreamV19 );
if( err != paNoError )
{
mStreamToken = 0;
mAudioThreadFillBuffersLoopRunning = false;
if (mListener && mNumCaptureChannels > 0)
mListener->OnAudioIOStopRecording();
StartStreamCleanup();
@ -2075,18 +2298,9 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
wxTheApp->ProcessEvent(e);
}
mAudioThreadFillBuffersLoopRunning = true;
// Enable warning popups for unfound aliased blockfiles.
wxGetApp().SetMissingAliasedFileWarningShouldShow(true);
//
// Generate an unique value each time, to be returned to
// clients accessing the AudioIO API, so they can query if
// are the ones who have reserved AudioIO or not.
//
mStreamToken = (++mNextStreamToken);
return mStreamToken;
}
@ -2203,6 +2417,7 @@ bool AudioIO::StartPortMidiStream()
mMidiPaused = false;
mMidiLoopPasses = 0;
mMidiOutputComplete = false;
mMaxMidiTimestamp = 0;
PrepareMidiIterator();
// It is ok to call this now, but do not send timestamped midi
@ -2391,9 +2606,10 @@ void AudioIO::StopStream()
// respond to these messages. This is probably a bug in PortMidi
// if the All Off messages do not get out, but for security,
// delay a bit so that messages can be delivered before closing
// the stream. It should take about 16ms to send All Off messages,
// so this will add 24ms latency.
wxMilliSleep(40); // deliver the all-off messages
// the stream. Add 2ms of "padding" to avoid any rounding errors.
while (mMaxMidiTimestamp + 2 > MidiTime()) {
wxMilliSleep(1); // deliver the all-off messages
}
Pm_Close(mMidiStream);
mMidiStream = NULL;
mIterator->end();
@ -3859,11 +4075,12 @@ void AudioIO::OutputEvent()
if (time < 0 || mSendMidiState) time = 0;
PmTimestamp timestamp = (PmTimestamp) (time * 1000); /* s to ms */
// The special event gAllNotesOffEvent means "end of playback, send
// The special event gAllNotesOff means "end of playback, send
// all notes off on all channels"
if (mNextEvent == &gAllNotesOff) {
AllNotesOff();
if (mPlayMode == gAudioIO->PLAY_LOOPED) {
bool looping = (mPlayMode == gAudioIO->PLAY_LOOPED);
AllNotesOff(looping);
if (looping) {
// jump back to beginning of loop
++mMidiLoopPasses;
PrepareMidiIterator(false, MidiLoopOffset());
@ -3970,6 +4187,10 @@ void AudioIO::OutputEvent()
}
}
if (command != -1) {
// keep track of greatest timestamp used
if (timestamp > mMaxMidiTimestamp) {
mMaxMidiTimestamp = timestamp;
}
Pm_WriteShort(mMidiStream, timestamp,
Pm_Message((int) (command + channel),
(long) data1, (long) data2));
@ -4046,10 +4267,20 @@ void AudioIO::FillMidiBuffers()
break;
}
SetHasSolo(hasSolo);
double time = AudioTime();
// If we compute until mNextEventTime > current audio track time,
// we would have a built-in compute-ahead of mAudioOutLatency, and
// it's probably good to compute MIDI when we compute audio (so when
// we stop, both stop about the same time).
double time = AudioTime(); // compute to here
// But if mAudioOutLatency is very low, we might need some extra
// compute-ahead to deal with mSynthLatency or even this thread.
double actual_latency = (MIDI_SLEEP + THREAD_LATENCY +
MIDI_MINIMAL_LATENCY_MS + mSynthLatency) * 0.001;
if (actual_latency > mAudioOutLatency) {
time += actual_latency - mAudioOutLatency;
}
while (mNextEvent &&
UncorrectedMidiEventTime() <
time + ((MIDI_SLEEP + mSynthLatency) * 0.001)) {
UncorrectedMidiEventTime() < time) {
OutputEvent();
GetNextEvent();
}
@ -4091,51 +4322,73 @@ double AudioIO::PauseTime()
}
// MidiTime() is an estimate in milliseconds of the current audio
// output (DAC) time + 1s. In other words, what audacity track time
// corresponds to the audio (including pause insertions) at the output?
//
PmTimestamp AudioIO::MidiTime()
{
//printf("AudioIO:MidiTime: PaUtil_GetTime() %g mAudioCallbackOutputDacTime %g time - outputTime %g\n",
// PaUtil_GetTime(), mAudioCallbackOutputDacTime, PaUtil_GetTime() - mAudioCallbackOutputDacTime);
// note: the extra 0.0005 is for rounding. Round down by casting to
// unsigned long, then convert to PmTimeStamp (currently signed)
// See long comments at the top of the file for the explanation of this
// calculation; we must use PaUtil_GetTime() here and also in the audio
// callback, to change the origin of times from portaudio, so the diffence of
// now and then is small, as the long comment assumes.
// PRL: the time correction is really Midi latency achieved by different
// means than specifiying it to Pm_OpenStream. The use of the accumulated
// means than specifying it to Pm_OpenStream. The use of the accumulated
// sample count generated by the audio callback (in AudioTime()) might also
// have the virtue of keeping the Midi output synched with audio, even though
// pmlinuxalsa.c does not implement any synchronization of its own.
// have the virtue of keeping the Midi output synched with audio.
#ifdef USE_TIME_INFO
auto offset = mAudioCallbackOutputDacTime - mAudioCallbackOutputCurrentTime;
#else
// We are now using mMidiTimeCorrection, computed once after opening the
// portaudio stream, rather than the difference of dac and current
// times reported to the audio callback; because on Linux with ALSA,
// the dac and current times were not reliable, and that caused irregular
// timing of Midi playback because latency was effectively zero.
auto offset = mMidiTimeCorrection;
#endif
auto clockChange = PaUtil_GetTime() - mAudioCallbackClockTime;
// auto offset = mAudioCallbackOutputDacTime - mAudioCallbackOutputCurrentTime;
return (PmTimestamp) ((unsigned long) (1000 * (
AudioTime() + 1.0005 -
mAudioFramesPerBuffer / mRate +
clockChange - offset
)));
PmTimestamp ts;
// subtract latency here because mSystemMinusAudioTime gets us
// to the current *write* time, but we're writing ahead by audio output
// latency (mAudioOutLatency).
double now = SystemTime(mUsingAlsa);
ts = (PmTimestamp) ((unsigned long)
(1000 * (now + 1.0005 -
mSystemMinusAudioTimePlusLatency)));
// printf("AudioIO::MidiTime() %d time %g sys-aud %g\n",
// ts, now, mSystemMinusAudioTime);
return ts + MIDI_MINIMAL_LATENCY_MS;
}
void AudioIO::AllNotesOff()
void AudioIO::AllNotesOff(bool looping)
{
#ifdef __WXGTK__
bool doDelay = !looping;
#else
bool doDelay = false;
WXUNUSED(looping);
#endif
// to keep track of when MIDI should all be delivered,
// update mMaxMidiTimestamp to now:
PmTimestamp now = MidiTime();
if (mMaxMidiTimestamp < now) {
mMaxMidiTimestamp = now;
}
#ifdef AUDIO_IO_GB_MIDI_WORKAROUND
// PRL:
// Send individual note-off messages for each note-on not yet paired.
// RBD:
// Even this did not work as planned. My guess is ALSA does not use
// a "stable sort" for timed messages, so that when a note-off is
// added later at the same time as a future note-on, the order is
// not respected, and the note-off can go first, leaving a stuck note.
// The workaround here is to use mMaxMidiTimestamp to ensure that
// note-offs come at least 1ms later than any previous message
// PRL:
// I think we should do that only when stopping or pausing, not when looping
// Note that on Linux, MIDI always uses ALSA, no matter whether portaudio
// uses some other host api.
mMaxMidiTimestamp += 1;
for (const auto &pair : mPendingNotesOff) {
Pm_WriteShort(mMidiStream, 0, Pm_Message(
Pm_WriteShort(mMidiStream,
(doDelay ? mMaxMidiTimestamp : 0),
Pm_Message(
0x90 + pair.first, pair.second, 0));
mMaxMidiTimestamp++; // allow 1ms per note-off
}
mPendingNotesOff.clear();
@ -4143,7 +4396,10 @@ void AudioIO::AllNotesOff()
#endif
for (int chan = 0; chan < 16; chan++) {
Pm_WriteShort(mMidiStream, 0, Pm_Message(0xB0 + chan, 0x7B, 0));
Pm_WriteShort(mMidiStream,
(doDelay ? mMaxMidiTimestamp : 0),
Pm_Message(0xB0 + chan, 0x7B, 0));
mMaxMidiTimestamp++; // allow 1ms per all-notes-off
}
}
@ -4357,19 +4613,78 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
(float *)alloca(framesPerBuffer*numPlaybackChannels * sizeof(float)) :
(float *)outputBuffer;
if (gAudioIO->mCallbackCount++ == 0) {
// This is effectively mSystemMinusAudioTime when the buffer is empty:
gAudioIO->mStartTime = SystemTime(gAudioIO->mUsingAlsa) - gAudioIO->mT0;
// later, mStartTime - mSystemMinusAudioTime will tell us latency
}
#ifdef EXPERIMENTAL_MIDI_OUT
/* GSW: Save timeInfo in case MidiPlayback needs it */
gAudioIO->mAudioCallbackClockTime = PaUtil_GetTime();
#ifdef USE_TIME_INFO
gAudioIO->mAudioCallbackOutputDacTime = timeInfo->outputBufferDacTime;
gAudioIO->mAudioCallbackOutputCurrentTime = timeInfo->currentTime;
#endif
/* for Linux, estimate a smooth audio time as a slowly-changing
offset from system time */
// rnow is system time as a double to simplify math
double rnow = SystemTime(gAudioIO->mUsingAlsa);
// anow is next-sample-to-be-computed audio time as a double
double anow = gAudioIO->AudioTime();
if (gAudioIO->mUsingAlsa) {
// timeInfo's fields are not all reliable.
// enow is audio time estimated from our clock synchronization protocol,
// which produces mSystemMinusAudioTime. But we want the estimate
// to drift low, so we steadily increase mSystemMinusAudioTime to
// simulate a fast system clock or a slow audio clock. If anow > enow,
// we'll update mSystemMinusAudioTime to keep in sync. (You might think
// we could just use anow as the "truth", but it has a lot of jitter,
// so we are using enow to smooth out this jitter, in fact to < 1ms.)
// Add worst-case clock drift using previous framesPerBuffer:
const auto increase =
gAudioIO->mAudioFramesPerBuffer * 0.0002 / gAudioIO->mRate;
gAudioIO->mSystemMinusAudioTime += increase;
gAudioIO->mSystemMinusAudioTimePlusLatency += increase;
double enow = rnow - gAudioIO->mSystemMinusAudioTime;
// now, use anow instead if it is ahead of enow
if (anow > enow) {
gAudioIO->mSystemMinusAudioTime = rnow - anow;
// Update our mAudioOutLatency estimate during the first 20 callbacks.
// During this period, the buffer should fill. Once we have a good
// estimate of mSystemMinusAudioTime (expected in fewer than 20 callbacks)
// we want to stop the updating in case there is clock drift, which would
// cause the mAudioOutLatency estimation to drift as well. The clock drift
// in the first 20 callbacks should be negligible, however.
if (gAudioIO->mCallbackCount < 20) {
gAudioIO->mAudioOutLatency = gAudioIO->mStartTime -
gAudioIO->mSystemMinusAudioTime;
}
gAudioIO->mSystemMinusAudioTimePlusLatency =
gAudioIO->mSystemMinusAudioTime + gAudioIO->mAudioOutLatency;
}
}
else {
// If not using Alsa, rely on timeInfo to have meaningful values that are
// more precise than the output latency value reported at stream start.
gAudioIO->mSystemMinusAudioTime = rnow - anow;
gAudioIO->mSystemMinusAudioTimePlusLatency =
gAudioIO->mSystemMinusAudioTime +
(timeInfo->outputBufferDacTime - timeInfo->currentTime);
}
// printf("in callback, mAudioCallbackOutputDacTime %g\n", gAudioIO->mAudioCallbackOutputDacTime); //DBG
gAudioIO->mAudioFramesPerBuffer = framesPerBuffer;
if(gAudioIO->IsPaused())
if (gAudioIO->IsPaused()
// PRL: Why was this added? Was it only because of the mysterious
// initial leading zeroes, now solved by setting mStreamToken early?
|| gAudioIO->mStreamToken <= 0
)
gAudioIO->mNumPauseFrames += framesPerBuffer;
// PRL: Note that when there is a separate MIDI thread, it is effectively
// blocked until the first visit to this line during a playback, and will
// not read gAudioIO->mSystemMinusAudioTimePlusLatency sooner:
gAudioIO->mNumFrames += framesPerBuffer;
#ifndef USE_MIDI_THREAD
@ -4678,6 +4993,8 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
if ((gAudioIO->ReversedTime()
? gAudioIO->mTime <= gAudioIO->mT1
: gAudioIO->mTime >= gAudioIO->mT1))
// PRL: singalling MIDI output complete is necessary if
// not USE_MIDI_THREAD, otherwise it's harmlessly redundant
gAudioIO->mMidiOutputComplete = true,
callbackReturn = paComplete;
}
@ -4739,6 +5056,8 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
? gAudioIO->mTime <= gAudioIO->mT1
: gAudioIO->mTime >= gAudioIO->mT1)) {
// PRL: singalling MIDI output complete is necessary if
// not USE_MIDI_THREAD, otherwise it's harmlessly redundant
gAudioIO->mMidiOutputComplete = true,
callbackReturn = paComplete;
}

59
src/AudioIO.h
View File

@ -93,18 +93,15 @@ DECLARE_EXPORTED_EVENT_TYPE(AUDACITY_DLL_API, EVT_AUDIOIO_PLAYBACK, -1);
DECLARE_EXPORTED_EVENT_TYPE(AUDACITY_DLL_API, EVT_AUDIOIO_CAPTURE, -1);
DECLARE_EXPORTED_EVENT_TYPE(AUDACITY_DLL_API, EVT_AUDIOIO_MONITOR, -1);
// PRL:
// If we always run a portaudio output stream (even just to produce silence)
// whenever we play Midi, then we can use just one thread for both, which
// simplifies synchronization problems and avoids the rush of notes at start of
// play. PRL.
#undef USE_MIDI_THREAD
// Whether we trust all of the time info passed to audacityAudioCallback
#ifdef __WXGTK__
#undef USE_TIME_INFO
#else
#define USE_TIME_INFO
#endif
// whenever we play Midi, then we might use just one thread for both.
// I thought this would improve MIDI synch problems on Linux/ALSA, but RBD
// convinced me it was neither a necessary nor sufficient fix. Perhaps too the
// MIDI thread might block in some error situations but we should then not
// also block the audio thread.
// So leave the separate thread ENABLED.
#define USE_MIDI_THREAD
struct ScrubbingOptions;
@ -460,7 +457,7 @@ private:
void GetNextEvent();
double AudioTime() { return mT0 + mNumFrames / mRate; }
double PauseTime();
void AllNotesOff();
void AllNotesOff(bool looping = false);
#endif
/** \brief Get the number of audio samples free in all of the playback
@ -542,10 +539,6 @@ private:
PmStream *mMidiStream;
PmError mLastPmError;
#ifndef USE_TIME_INFO
PaTime mMidiTimeCorrection; // seconds
#endif
/// Latency of MIDI synthesizer
long mSynthLatency; // ms
@ -554,13 +547,6 @@ private:
/// PortAudio's clock time
volatile double mAudioCallbackClockTime;
#ifdef USE_TIME_INFO
/// PortAudio's currentTime -- its origin is unspecified!
volatile double mAudioCallbackOutputCurrentTime;
/// PortAudio's outTime
volatile double mAudioCallbackOutputDacTime;
#endif
/// Number of frames output, including pauses
volatile long mNumFrames;
/// How many frames of zeros were output due to pauses?
@ -573,6 +559,30 @@ private:
/// Used by Midi process to record that pause has begun,
/// so that AllNotesOff() is only delivered once
volatile bool mMidiPaused;
/// The largest timestamp written so far, used to delay
/// stream closing until last message has been delivered
PmTimestamp mMaxMidiTimestamp;
/// Offset from ideal sample computation time to system time,
/// where "ideal" means when we would get the callback if there
/// were no scheduling delays or computation time
double mSystemMinusAudioTime;
/// audio output latency reported by PortAudio
/// (initially; for Alsa, we adjust it to the largest "observed" value)
double mAudioOutLatency;
// Next two are used to adjust the previous two, if
// PortAudio does not provide the info (using ALSA):
/// time of first callback
/// used to find "observed" latency
double mStartTime;
/// number of callbacks since stream start
long mCallbackCount;
/// Make just one variable to communicate from audio to MIDI thread,
/// to avoid problems of atomicity of updates
volatile double mSystemMinusAudioTimePlusLatency;
Alg_seq_ptr mSeq;
std::unique_ptr<Alg_iterator> mIterator;
@ -727,6 +737,8 @@ private:
const TimeTrack *mTimeTrack;
bool mUsingAlsa { false };
// For cacheing supported sample rates
static int mCachedPlaybackIndex;
static wxArrayLong mCachedPlaybackRates;
@ -786,4 +798,3 @@ private:
};
#endif