Replace float/UInt64 durations by a more precise Time class (#388)

Improve Clock class with atomic RestartIfOver method and allows to choose required precision
This commit is contained in:
Jérôme Leclercq
2022-12-29 21:31:46 +01:00
committed by GitHub
parent 1de5f65536
commit dd421a6385
84 changed files with 1278 additions and 663 deletions

View File

@@ -39,6 +39,27 @@ namespace Nz
return m_pitch;
}
Time DummyAudioSource::GetPlayingOffset() const
{
if (m_status == SoundStatus::Stopped)
return Time::Zero(); //< Always return 0 when stopped, to mimic OpenAL behavior
Time bufferTime = UpdateTime();
Time playingOffset = Time::Zero();
// All processed buffers count
for (const auto& processedBuffer : m_processedBuffers)
playingOffset += processedBuffer->GetDuration();
if (!m_queuedBuffers.empty())
{
auto& frontBuffer = m_queuedBuffers.front();
playingOffset += std::min(bufferTime, frontBuffer->GetDuration());
}
return playingOffset;
}
Vector3f DummyAudioSource::GetPosition() const
{
return m_position;
@@ -49,7 +70,7 @@ namespace Nz
if (m_status == SoundStatus::Stopped)
return 0; //< Always return 0 when stopped, to mimic OpenAL behavior
UInt64 bufferTime = UpdateTime();
Time bufferTime = UpdateTime();
UInt64 sampleOffset = 0;
// All processed buffers count in sample offset
@@ -59,7 +80,7 @@ namespace Nz
if (!m_queuedBuffers.empty())
{
auto& frontBuffer = m_queuedBuffers.front();
UInt64 bufferOffset = bufferTime * frontBuffer->GetSampleRate() / 1000;
UInt64 bufferOffset = bufferTime.AsMicroseconds() * frontBuffer->GetSampleRate() / 1'000'000ll;
UInt64 bufferDuration = frontBuffer->GetSampleCount() / GetChannelCount(frontBuffer->GetAudioFormat());
sampleOffset += std::min(bufferOffset, bufferDuration);
@@ -72,7 +93,7 @@ namespace Nz
{
OffsetWithLatency info;
info.sampleOffset = GetSampleOffset() * 1000;
info.sourceLatency = 0;
info.sourceLatency = Time::Zero();
return info;
}
@@ -120,19 +141,19 @@ namespace Nz
void DummyAudioSource::Play()
{
if (m_status == SoundStatus::Paused)
m_playClock.Unpause();
else
if (m_status != SoundStatus::Paused)
{
// playing or stopped, restart
RequeueBuffers();
// special case, we are stopped but SetSampleOffset has been called
if (m_status == SoundStatus::Stopped && m_playClock.GetMilliseconds() != 0)
if (m_status == SoundStatus::Stopped && m_playClock.GetElapsedTime() != Time::Zero())
m_playClock.Unpause();
else
m_playClock.Restart(); //< already playing or stopped, restart from beginning
}
else
m_playClock.Unpause();
m_status = SoundStatus::Playing;
}
@@ -161,6 +182,13 @@ namespace Nz
m_pitch = pitch;
}
void DummyAudioSource::SetPlayingOffset(Time offset)
{
// Next UpdateTime call will handle this properly
RequeueBuffers();
m_playClock.Restart(offset, m_playClock.IsPaused());
}
void DummyAudioSource::SetPosition(const Vector3f& position)
{
m_position = position;
@@ -187,7 +215,7 @@ namespace Nz
if (!m_queuedBuffers.empty())
{
UInt64 timeOffset = 1'000'000ULL * offset / m_queuedBuffers.front()->GetSampleRate();
Time timeOffset = Time::Microseconds(1'000'000ll * offset / m_queuedBuffers.front()->GetSampleRate());
m_playClock.Restart(timeOffset, m_playClock.IsPaused());
}
else
@@ -206,7 +234,7 @@ namespace Nz
void DummyAudioSource::Stop()
{
m_playClock.Restart(0, true);
m_playClock.Restart(Time::Zero(), true);
m_status = SoundStatus::Stopped;
}
@@ -246,9 +274,10 @@ namespace Nz
}
}
UInt64 DummyAudioSource::UpdateTime() const
Time DummyAudioSource::UpdateTime() const
{
UInt64 currentTime = m_playClock.GetMilliseconds();
Time currentTime = m_playClock.GetElapsedTime();
bool isPaused = m_playClock.IsPaused();
while (!m_queuedBuffers.empty() && currentTime >= m_queuedBuffers.front()->GetDuration())
{
@@ -278,10 +307,14 @@ namespace Nz
}
}
else
{
m_status = SoundStatus::Stopped;
currentTime = Time::Zero();
isPaused = m_playClock.IsPaused();
}
}
m_playClock.Restart(currentTime * 1000, m_playClock.IsPaused()); //< Adjust time
m_playClock.Restart(currentTime, isPaused); //< Adjust time
return currentTime;
}
}