diff options
| -rw-r--r-- | examples/alffplay.cpp | 59 |
1 files changed, 26 insertions, 33 deletions
diff --git a/examples/alffplay.cpp b/examples/alffplay.cpp index 28b968e2..c51c5119 100644 --- a/examples/alffplay.cpp +++ b/examples/alffplay.cpp @@ -91,16 +91,17 @@ inline constexpr int64_t operator "" _i64(unsigned long long int n) noexcept { r #define M_PI (3.14159265358979323846) #endif +using fixed32 = std::chrono::duration<int64_t,std::ratio<1,(1_i64<<32)>>; using nanoseconds = std::chrono::nanoseconds; using microseconds = std::chrono::microseconds; using milliseconds = std::chrono::milliseconds; using seconds = std::chrono::seconds; using seconds_d64 = std::chrono::duration<double>; -const std::string AppName("alffplay"); +const std::string AppName{"alffplay"}; -bool EnableDirectOut = false; -bool EnableWideStereo = false; +bool EnableDirectOut{false}; +bool EnableWideStereo{false}; LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT; LPALCGETINTEGER64VSOFT alcGetInteger64vSOFT; @@ -115,20 +116,20 @@ LPALEVENTCONTROLSOFT alEventControlSOFT; LPALEVENTCALLBACKSOFT alEventCallbackSOFT; #endif -const seconds AVNoSyncThreshold(10); +const seconds AVNoSyncThreshold{10}; const milliseconds VideoSyncThreshold(10); #define VIDEO_PICTURE_QUEUE_SIZE 16 -const seconds_d64 AudioSyncThreshold(0.03); -const milliseconds AudioSampleCorrectionMax(50); +const seconds_d64 AudioSyncThreshold{0.03}; +const milliseconds AudioSampleCorrectionMax{50}; /* Averaging filter coefficient for audio sync. */ #define AUDIO_DIFF_AVG_NB 20 -const double AudioAvgFilterCoeff = std::pow(0.01, 1.0/AUDIO_DIFF_AVG_NB); +const double AudioAvgFilterCoeff{std::pow(0.01, 1.0/AUDIO_DIFF_AVG_NB)}; /* Per-buffer size, in time */ -const milliseconds AudioBufferTime(20); +const milliseconds AudioBufferTime{20}; /* Buffer total size, in time (should be divisible by the buffer time) */ -const milliseconds AudioBufferTotalTime(800); +const milliseconds AudioBufferTotalTime{800}; #define MAX_QUEUE_SIZE (15 * 1024 * 1024) /* Bytes of compressed data to keep queued */ @@ -419,15 +420,15 @@ nanoseconds AudioState::getClockNoLock() // Get the current device clock time and latency. auto device = alcGetContextsDevice(alcGetCurrentContext()); - ALCint64SOFT devtimes[2] = {0,0}; + ALCint64SOFT devtimes[2]{0,0}; alcGetInteger64vSOFT(device, ALC_DEVICE_CLOCK_LATENCY_SOFT, 2, devtimes); - auto latency = nanoseconds(devtimes[1]); - auto device_time = nanoseconds(devtimes[0]); + auto latency = nanoseconds{devtimes[1]}; + auto device_time = nanoseconds{devtimes[0]}; // The clock is simply the current device time relative to the recorded // start time. We can also subtract the latency to get more a accurate // position of where the audio device actually is in the output stream. - return device_time - mDeviceStartTime - latency; + std::max(device_time - mDeviceStartTime - latency, nanoseconds::zero()); } /* The source-based clock is based on 4 components: @@ -445,12 +446,10 @@ nanoseconds AudioState::getClockNoLock() * sample at OpenAL's current position, and subtracting the source latency * from that gives the timestamp of the sample currently at the DAC. */ - nanoseconds pts = mCurrentPts; + nanoseconds pts{mCurrentPts}; if(mSource) { ALint64SOFT offset[2]; - ALint queued; - ALint status; /* NOTE: The source state must be checked last, in case an underrun * occurs and the source stops between retrieving the offset+latency @@ -461,9 +460,10 @@ nanoseconds AudioState::getClockNoLock() { ALint ioffset; alGetSourcei(mSource, AL_SAMPLE_OFFSET, &ioffset); - offset[0] = static_cast<ALint64SOFT>(ioffset) << 32; + offset[0] = ALint64SOFT{ioffset} << 32; offset[1] = 0; } + ALint queued, status; alGetSourcei(mSource, AL_BUFFERS_QUEUED, &queued); alGetSourcei(mSource, AL_SOURCE_STATE, &status); @@ -473,16 +473,13 @@ nanoseconds AudioState::getClockNoLock() * when it starts recovery. */ if(status != AL_STOPPED) { - using fixed32 = std::chrono::duration<int64_t,std::ratio<1,(1ll<<32)>>; - pts -= AudioBufferTime*queued; pts += std::chrono::duration_cast<nanoseconds>( - fixed32(offset[0] / mCodecCtx->sample_rate) - ); + fixed32{offset[0] / mCodecCtx->sample_rate}); } /* Don't offset by the latency if the source isn't playing. */ if(status == AL_PLAYING) - pts -= nanoseconds(offset[1]); + pts -= nanoseconds{offset[1]}; } return std::max(pts, nanoseconds::zero()); @@ -503,16 +500,14 @@ void AudioState::startPlayback() alSourcePlay(mSource); if(alcGetInteger64vSOFT) { - using fixed32 = std::chrono::duration<int64_t,std::ratio<1,(1ll<<32)>>; - // Subtract the total buffer queue time from the current pts to get the // pts of the start of the queue. - nanoseconds startpts = mCurrentPts - AudioBufferTotalTime; - int64_t srctimes[2]={0,0}; + nanoseconds startpts{mCurrentPts - AudioBufferTotalTime}; + int64_t srctimes[2]{0,0}; alGetSourcei64vSOFT(mSource, AL_SAMPLE_OFFSET_CLOCK_SOFT, srctimes); - auto device_time = nanoseconds(srctimes[1]); - auto src_offset = std::chrono::duration_cast<nanoseconds>(fixed32(srctimes[0])) / - mCodecCtx->sample_rate; + auto device_time = nanoseconds{srctimes[1]}; + auto src_offset = std::chrono::duration_cast<nanoseconds>(fixed32{srctimes[0]}) / + mCodecCtx->sample_rate; // The mixer may have ticked and incremented the device time and sample // offset, so subtract the source offset from the device time to get @@ -1068,10 +1063,8 @@ int AudioState::handler() */ int64_t devtime{}; alcGetInteger64vSOFT(alcGetContextsDevice(alcGetCurrentContext()), - ALC_DEVICE_CLOCK_SOFT, 1, &devtime); - auto device_time = nanoseconds{devtime}; - - mDeviceStartTime = device_time - mCurrentPts + AudioBufferTotalTime; + ALC_DEVICE_CLOCK_SOFT, 1, &devtime); + mDeviceStartTime = nanoseconds{devtime} - mCurrentPts; } continue; } |