2 * An example showing how to play a stream sync'd to video, using ffmpeg.
7 #include <condition_variable>
33 _Pragma("GCC diagnostic push")
34 _Pragma("GCC diagnostic ignored \"-Wconversion\"")
35 _Pragma("GCC diagnostic ignored \"-Wold-style-cast\"")
37 #include "libavcodec/avcodec.h"
38 #include "libavformat/avformat.h"
39 #include "libavformat/avio.h"
40 #include "libavformat/version.h"
41 #include "libavutil/avutil.h"
42 #include "libavutil/error.h"
43 #include "libavutil/frame.h"
44 #include "libavutil/mem.h"
45 #include "libavutil/pixfmt.h"
46 #include "libavutil/rational.h"
47 #include "libavutil/samplefmt.h"
48 #include "libavutil/time.h"
49 #include "libavutil/version.h"
50 #include "libavutil/channel_layout.h"
51 #include "libswscale/swscale.h"
52 #include "libswresample/swresample.h"
54 constexpr auto AVNoPtsValue
= AV_NOPTS_VALUE
;
55 constexpr auto AVErrorEOF
= AVERROR_EOF
;
59 _Pragma("GCC diagnostic pop")
69 #include "common/alhelpers.h"
72 /* Undefine this to disable use of experimental extensions. Don't use for
73 * production code! Interfaces and behavior may change prior to being
76 #define ALLOW_EXPERIMENTAL_EXTS
78 #ifdef ALLOW_EXPERIMENTAL_EXTS
79 #ifndef AL_SOFT_callback_buffer
80 #define AL_SOFT_callback_buffer
81 typedef unsigned int ALbitfieldSOFT
;
82 #define AL_BUFFER_CALLBACK_FUNCTION_SOFT 0x19A0
83 #define AL_BUFFER_CALLBACK_USER_PARAM_SOFT 0x19A1
84 typedef ALsizei (AL_APIENTRY
*ALBUFFERCALLBACKTYPESOFT
)(ALvoid
*userptr
, ALvoid
*sampledata
, ALsizei numbytes
);
85 typedef void (AL_APIENTRY
*LPALBUFFERCALLBACKSOFT
)(ALuint buffer
, ALenum format
, ALsizei freq
, ALBUFFERCALLBACKTYPESOFT callback
, ALvoid
*userptr
, ALbitfieldSOFT flags
);
86 typedef void (AL_APIENTRY
*LPALGETBUFFERPTRSOFT
)(ALuint buffer
, ALenum param
, ALvoid
**value
);
87 typedef void (AL_APIENTRY
*LPALGETBUFFER3PTRSOFT
)(ALuint buffer
, ALenum param
, ALvoid
**value1
, ALvoid
**value2
, ALvoid
**value3
);
88 typedef void (AL_APIENTRY
*LPALGETBUFFERPTRVSOFT
)(ALuint buffer
, ALenum param
, ALvoid
**values
);
92 #define AL_FORMAT_UHJ2CHN8_SOFT 0x19A2
93 #define AL_FORMAT_UHJ2CHN16_SOFT 0x19A3
94 #define AL_FORMAT_UHJ2CHN_FLOAT32_SOFT 0x19A4
95 #define AL_FORMAT_UHJ3CHN8_SOFT 0x19A5
96 #define AL_FORMAT_UHJ3CHN16_SOFT 0x19A6
97 #define AL_FORMAT_UHJ3CHN_FLOAT32_SOFT 0x19A7
98 #define AL_FORMAT_UHJ4CHN8_SOFT 0x19A8
99 #define AL_FORMAT_UHJ4CHN16_SOFT 0x19A9
100 #define AL_FORMAT_UHJ4CHN_FLOAT32_SOFT 0x19AA
101 #define AL_STEREO_MODE_SOFT 0x19B0
102 #define AL_NORMAL_SOFT 0x0000
103 #define AL_SUPER_STEREO_SOFT 0x0001
104 #define AL_SUPER_STEREO_WIDTH_SOFT 0x19B1
106 #endif /* ALLOW_EXPERIMENTAL_EXTS */
111 inline constexpr int64_t operator "" _i64(unsigned long long int n
) noexcept
{ return static_cast<int64_t>(n
); }
114 #define M_PI (3.14159265358979323846)
117 using fixed32
= std::chrono::duration
<int64_t,std::ratio
<1,(1_i64
<<32)>>;
118 using nanoseconds
= std::chrono::nanoseconds
;
119 using microseconds
= std::chrono::microseconds
;
120 using milliseconds
= std::chrono::milliseconds
;
121 using seconds
= std::chrono::seconds
;
122 using seconds_d64
= std::chrono::duration
<double>;
123 using std::chrono::duration_cast
;
125 const std::string AppName
{"alffplay"};
127 ALenum DirectOutMode
{AL_FALSE
};
128 bool EnableWideStereo
{false};
129 bool EnableSuperStereo
{false};
130 bool DisableVideo
{false};
131 LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT
;
132 LPALCGETINTEGER64VSOFT alcGetInteger64vSOFT
;
133 LPALEVENTCONTROLSOFT alEventControlSOFT
;
134 LPALEVENTCALLBACKSOFT alEventCallbackSOFT
;
136 #ifdef AL_SOFT_callback_buffer
137 LPALBUFFERCALLBACKSOFT alBufferCallbackSOFT
;
139 ALenum FormatStereo8
{AL_FORMAT_STEREO8
};
140 ALenum FormatStereo16
{AL_FORMAT_STEREO16
};
141 ALenum FormatStereo32F
{AL_FORMAT_STEREO_FLOAT32
};
143 const seconds AVNoSyncThreshold
{10};
145 #define VIDEO_PICTURE_QUEUE_SIZE 24
147 const seconds_d64 AudioSyncThreshold
{0.03};
148 const milliseconds AudioSampleCorrectionMax
{50};
149 /* Averaging filter coefficient for audio sync. */
150 #define AUDIO_DIFF_AVG_NB 20
151 const double AudioAvgFilterCoeff
{std::pow(0.01, 1.0/AUDIO_DIFF_AVG_NB
)};
152 /* Per-buffer size, in time */
153 constexpr milliseconds AudioBufferTime
{20};
154 /* Buffer total size, in time (should be divisible by the buffer time) */
155 constexpr milliseconds AudioBufferTotalTime
{800};
156 constexpr auto AudioBufferCount
= AudioBufferTotalTime
/ AudioBufferTime
;
159 FF_MOVIE_DONE_EVENT
= SDL_USEREVENT
162 enum class SyncMaster
{
171 inline microseconds
get_avtime()
172 { return microseconds
{av_gettime()}; }
174 /* Define unique_ptrs to auto-cleanup associated ffmpeg objects. */
175 struct AVIOContextDeleter
{
176 void operator()(AVIOContext
*ptr
) { avio_closep(&ptr
); }
178 using AVIOContextPtr
= std::unique_ptr
<AVIOContext
,AVIOContextDeleter
>;
180 struct AVFormatCtxDeleter
{
181 void operator()(AVFormatContext
*ptr
) { avformat_close_input(&ptr
); }
183 using AVFormatCtxPtr
= std::unique_ptr
<AVFormatContext
,AVFormatCtxDeleter
>;
185 struct AVCodecCtxDeleter
{
186 void operator()(AVCodecContext
*ptr
) { avcodec_free_context(&ptr
); }
188 using AVCodecCtxPtr
= std::unique_ptr
<AVCodecContext
,AVCodecCtxDeleter
>;
190 struct AVPacketDeleter
{
191 void operator()(AVPacket
*pkt
) { av_packet_free(&pkt
); }
193 using AVPacketPtr
= std::unique_ptr
<AVPacket
,AVPacketDeleter
>;
195 struct AVFrameDeleter
{
196 void operator()(AVFrame
*ptr
) { av_frame_free(&ptr
); }
198 using AVFramePtr
= std::unique_ptr
<AVFrame
,AVFrameDeleter
>;
200 struct SwrContextDeleter
{
201 void operator()(SwrContext
*ptr
) { swr_free(&ptr
); }
203 using SwrContextPtr
= std::unique_ptr
<SwrContext
,SwrContextDeleter
>;
205 struct SwsContextDeleter
{
206 void operator()(SwsContext
*ptr
) { sws_freeContext(ptr
); }
208 using SwsContextPtr
= std::unique_ptr
<SwsContext
,SwsContextDeleter
>;
211 template<size_t SizeLimit
>
213 std::mutex mPacketMutex
, mFrameMutex
;
214 std::condition_variable mPacketCond
;
215 std::condition_variable mInFrameCond
, mOutFrameCond
;
217 std::deque
<AVPacketPtr
> mPackets
;
218 size_t mTotalSize
{0};
219 bool mFinished
{false};
221 AVPacketPtr
getPacket()
223 std::unique_lock
<std::mutex
> plock
{mPacketMutex
};
224 while(mPackets
.empty() && !mFinished
)
225 mPacketCond
.wait(plock
);
229 auto ret
= std::move(mPackets
.front());
230 mPackets
.pop_front();
231 mTotalSize
-= static_cast<unsigned int>(ret
->size
);
236 int sendPacket(AVCodecContext
*codecctx
)
238 AVPacketPtr packet
{getPacket()};
242 std::unique_lock
<std::mutex
> flock
{mFrameMutex
};
243 while((ret
=avcodec_send_packet(codecctx
, packet
.get())) == AVERROR(EAGAIN
))
244 mInFrameCond
.wait_for(flock
, milliseconds
{50});
246 mOutFrameCond
.notify_one();
250 if(!ret
) return AVErrorEOF
;
251 std::cerr
<< "Failed to send flush packet: "<<ret
<<std::endl
;
255 std::cerr
<< "Failed to send packet: "<<ret
<<std::endl
;
259 int receiveFrame(AVCodecContext
*codecctx
, AVFrame
*frame
)
263 std::unique_lock
<std::mutex
> flock
{mFrameMutex
};
264 while((ret
=avcodec_receive_frame(codecctx
, frame
)) == AVERROR(EAGAIN
))
265 mOutFrameCond
.wait_for(flock
, milliseconds
{50});
267 mInFrameCond
.notify_one();
274 std::lock_guard
<std::mutex
> _
{mPacketMutex
};
277 mPacketCond
.notify_one();
283 std::lock_guard
<std::mutex
> _
{mPacketMutex
};
289 mPacketCond
.notify_one();
292 bool put(const AVPacket
*pkt
)
295 std::unique_lock
<std::mutex
> lock
{mPacketMutex
};
296 if(mTotalSize
>= SizeLimit
|| mFinished
)
299 mPackets
.push_back(AVPacketPtr
{av_packet_alloc()});
300 if(av_packet_ref(mPackets
.back().get(), pkt
) != 0)
306 mTotalSize
+= static_cast<unsigned int>(mPackets
.back()->size
);
308 mPacketCond
.notify_one();
319 AVStream
*mStream
{nullptr};
320 AVCodecCtxPtr mCodecCtx
;
322 DataQueue
<2*1024*1024> mQueue
;
324 /* Used for clock difference average computation */
325 seconds_d64 mClockDiffAvg
{0};
327 /* Time of the next sample to be buffered */
328 nanoseconds mCurrentPts
{0};
330 /* Device clock time that the stream started at. */
331 nanoseconds mDeviceStartTime
{nanoseconds::min()};
333 /* Decompressed sample frame, and swresample context for conversion */
334 AVFramePtr mDecodedFrame
;
335 SwrContextPtr mSwresCtx
;
337 /* Conversion format, for what gets fed to OpenAL */
338 uint64_t mDstChanLayout
{0};
339 AVSampleFormat mDstSampleFmt
{AV_SAMPLE_FMT_NONE
};
341 /* Storage of converted samples */
342 uint8_t *mSamples
{nullptr};
343 int mSamplesLen
{0}; /* In samples */
347 std::unique_ptr
<uint8_t[]> mBufferData
;
348 size_t mBufferDataSize
{0};
349 std::atomic
<size_t> mReadPos
{0};
350 std::atomic
<size_t> mWritePos
{0};
353 ALenum mFormat
{AL_NONE
};
354 ALuint mFrameSize
{0};
356 std::mutex mSrcMutex
;
357 std::condition_variable mSrcCond
;
358 std::atomic_flag mConnected
;
360 std::array
<ALuint
,AudioBufferCount
> mBuffers
{};
361 ALuint mBufferIdx
{0};
363 AudioState(MovieState
&movie
) : mMovie(movie
)
364 { mConnected
.test_and_set(std::memory_order_relaxed
); }
368 alDeleteSources(1, &mSource
);
370 alDeleteBuffers(static_cast<ALsizei
>(mBuffers
.size()), mBuffers
.data());
375 static void AL_APIENTRY
eventCallbackC(ALenum eventType
, ALuint object
, ALuint param
,
376 ALsizei length
, const ALchar
*message
, void *userParam
)
377 { static_cast<AudioState
*>(userParam
)->eventCallback(eventType
, object
, param
, length
, message
); }
378 void eventCallback(ALenum eventType
, ALuint object
, ALuint param
, ALsizei length
,
379 const ALchar
*message
);
381 #ifdef AL_SOFT_callback_buffer
382 static ALsizei AL_APIENTRY
bufferCallbackC(void *userptr
, void *data
, ALsizei size
)
383 { return static_cast<AudioState
*>(userptr
)->bufferCallback(data
, size
); }
384 ALsizei
bufferCallback(void *data
, ALsizei size
);
387 nanoseconds
getClockNoLock();
388 nanoseconds
getClock()
390 std::lock_guard
<std::mutex
> lock
{mSrcMutex
};
391 return getClockNoLock();
394 bool startPlayback();
398 bool readAudio(uint8_t *samples
, unsigned int length
, int &sample_skip
);
399 bool readAudio(int sample_skip
);
407 AVStream
*mStream
{nullptr};
408 AVCodecCtxPtr mCodecCtx
;
410 DataQueue
<14*1024*1024> mQueue
;
412 /* The pts of the currently displayed frame, and the time (av_gettime) it
413 * was last updated - used to have running video pts
415 nanoseconds mDisplayPts
{0};
416 microseconds mDisplayPtsTime
{microseconds::min()};
417 std::mutex mDispPtsMutex
;
419 /* Swscale context for format conversion */
420 SwsContextPtr mSwscaleCtx
;
424 nanoseconds mPts
{nanoseconds::min()};
426 std::array
<Picture
,VIDEO_PICTURE_QUEUE_SIZE
> mPictQ
;
427 std::atomic
<size_t> mPictQRead
{0u}, mPictQWrite
{1u};
428 std::mutex mPictQMutex
;
429 std::condition_variable mPictQCond
;
431 SDL_Texture
*mImage
{nullptr};
432 int mWidth
{0}, mHeight
{0}; /* Full texture size */
433 bool mFirstUpdate
{true};
435 std::atomic
<bool> mEOS
{false};
436 std::atomic
<bool> mFinalUpdate
{false};
438 VideoState(MovieState
&movie
) : mMovie(movie
) { }
442 SDL_DestroyTexture(mImage
);
446 nanoseconds
getClock();
448 void display(SDL_Window
*screen
, SDL_Renderer
*renderer
, AVFrame
*frame
);
449 void updateVideo(SDL_Window
*screen
, SDL_Renderer
*renderer
, bool redraw
);
454 AVIOContextPtr mIOContext
;
455 AVFormatCtxPtr mFormatCtx
;
457 SyncMaster mAVSyncType
{SyncMaster::Default
};
459 microseconds mClockBase
{microseconds::min()};
461 std::atomic
<bool> mQuit
{false};
466 std::mutex mStartupMutex
;
467 std::condition_variable mStartupCond
;
468 bool mStartupDone
{false};
470 std::thread mParseThread
;
471 std::thread mAudioThread
;
472 std::thread mVideoThread
;
474 std::string mFilename
;
476 MovieState(std::string fname
)
477 : mAudio(*this), mVideo(*this), mFilename(std::move(fname
))
482 if(mParseThread
.joinable())
486 static int decode_interrupt_cb(void *ctx
);
488 void setTitle(SDL_Window
*window
);
491 nanoseconds
getClock();
493 nanoseconds
getMasterClock();
495 nanoseconds
getDuration();
497 int streamComponentOpen(unsigned int stream_index
);
502 nanoseconds
AudioState::getClockNoLock()
504 // The audio clock is the timestamp of the sample currently being heard.
505 if(alcGetInteger64vSOFT
)
507 // If device start time = min, we aren't playing yet.
508 if(mDeviceStartTime
== nanoseconds::min())
509 return nanoseconds::zero();
511 // Get the current device clock time and latency.
512 auto device
= alcGetContextsDevice(alcGetCurrentContext());
513 ALCint64SOFT devtimes
[2]{0,0};
514 alcGetInteger64vSOFT(device
, ALC_DEVICE_CLOCK_LATENCY_SOFT
, 2, devtimes
);
515 auto latency
= nanoseconds
{devtimes
[1]};
516 auto device_time
= nanoseconds
{devtimes
[0]};
518 // The clock is simply the current device time relative to the recorded
519 // start time. We can also subtract the latency to get more a accurate
520 // position of where the audio device actually is in the output stream.
521 return device_time
- mDeviceStartTime
- latency
;
524 if(mBufferDataSize
> 0)
526 if(mDeviceStartTime
== nanoseconds::min())
527 return nanoseconds::zero();
529 /* With a callback buffer and no device clock, mDeviceStartTime is
530 * actually the timestamp of the first sample frame played. The audio
531 * clock, then, is that plus the current source offset.
533 ALint64SOFT offset
[2];
534 if(alGetSourcei64vSOFT
)
535 alGetSourcei64vSOFT(mSource
, AL_SAMPLE_OFFSET_LATENCY_SOFT
, offset
);
539 alGetSourcei(mSource
, AL_SAMPLE_OFFSET
, &ioffset
);
540 offset
[0] = ALint64SOFT
{ioffset
} << 32;
543 /* NOTE: The source state must be checked last, in case an underrun
544 * occurs and the source stops between getting the state and retrieving
545 * the offset+latency.
548 alGetSourcei(mSource
, AL_SOURCE_STATE
, &status
);
551 if(status
== AL_PLAYING
|| status
== AL_PAUSED
)
552 pts
= mDeviceStartTime
- nanoseconds
{offset
[1]} +
553 duration_cast
<nanoseconds
>(fixed32
{offset
[0] / mCodecCtx
->sample_rate
});
556 /* If the source is stopped, the pts of the next sample to be heard
557 * is the pts of the next sample to be buffered, minus the amount
558 * already in the buffer ready to play.
560 const size_t woffset
{mWritePos
.load(std::memory_order_acquire
)};
561 const size_t roffset
{mReadPos
.load(std::memory_order_relaxed
)};
562 const size_t readable
{((woffset
>= roffset
) ? woffset
: (mBufferDataSize
+woffset
)) -
565 pts
= mCurrentPts
- nanoseconds
{seconds
{readable
/mFrameSize
}}/mCodecCtx
->sample_rate
;
571 /* The source-based clock is based on 4 components:
572 * 1 - The timestamp of the next sample to buffer (mCurrentPts)
573 * 2 - The length of the source's buffer queue
574 * (AudioBufferTime*AL_BUFFERS_QUEUED)
575 * 3 - The offset OpenAL is currently at in the source (the first value
576 * from AL_SAMPLE_OFFSET_LATENCY_SOFT)
577 * 4 - The latency between OpenAL and the DAC (the second value from
578 * AL_SAMPLE_OFFSET_LATENCY_SOFT)
580 * Subtracting the length of the source queue from the next sample's
581 * timestamp gives the timestamp of the sample at the start of the source
582 * queue. Adding the source offset to that results in the timestamp for the
583 * sample at OpenAL's current position, and subtracting the source latency
584 * from that gives the timestamp of the sample currently at the DAC.
586 nanoseconds pts
{mCurrentPts
};
589 ALint64SOFT offset
[2];
590 if(alGetSourcei64vSOFT
)
591 alGetSourcei64vSOFT(mSource
, AL_SAMPLE_OFFSET_LATENCY_SOFT
, offset
);
595 alGetSourcei(mSource
, AL_SAMPLE_OFFSET
, &ioffset
);
596 offset
[0] = ALint64SOFT
{ioffset
} << 32;
599 ALint queued
, status
;
600 alGetSourcei(mSource
, AL_BUFFERS_QUEUED
, &queued
);
601 alGetSourcei(mSource
, AL_SOURCE_STATE
, &status
);
603 /* If the source is AL_STOPPED, then there was an underrun and all
604 * buffers are processed, so ignore the source queue. The audio thread
605 * will put the source into an AL_INITIAL state and clear the queue
606 * when it starts recovery.
608 if(status
!= AL_STOPPED
)
610 pts
-= AudioBufferTime
*queued
;
611 pts
+= duration_cast
<nanoseconds
>(fixed32
{offset
[0] / mCodecCtx
->sample_rate
});
613 /* Don't offset by the latency if the source isn't playing. */
614 if(status
== AL_PLAYING
)
615 pts
-= nanoseconds
{offset
[1]};
618 return std::max(pts
, nanoseconds::zero());
621 bool AudioState::startPlayback()
623 const size_t woffset
{mWritePos
.load(std::memory_order_acquire
)};
624 const size_t roffset
{mReadPos
.load(std::memory_order_relaxed
)};
625 const size_t readable
{((woffset
>= roffset
) ? woffset
: (mBufferDataSize
+woffset
)) -
628 if(mBufferDataSize
> 0)
632 if(!alcGetInteger64vSOFT
)
633 mDeviceStartTime
= mCurrentPts
-
634 nanoseconds
{seconds
{readable
/mFrameSize
}}/mCodecCtx
->sample_rate
;
639 alGetSourcei(mSource
, AL_BUFFERS_QUEUED
, &queued
);
640 if(queued
== 0) return false;
643 alSourcePlay(mSource
);
644 if(alcGetInteger64vSOFT
)
646 /* Subtract the total buffer queue time from the current pts to get the
647 * pts of the start of the queue.
649 int64_t srctimes
[2]{0,0};
650 alGetSourcei64vSOFT(mSource
, AL_SAMPLE_OFFSET_CLOCK_SOFT
, srctimes
);
651 auto device_time
= nanoseconds
{srctimes
[1]};
652 auto src_offset
= duration_cast
<nanoseconds
>(fixed32
{srctimes
[0]}) /
653 mCodecCtx
->sample_rate
;
655 /* The mixer may have ticked and incremented the device time and sample
656 * offset, so subtract the source offset from the device time to get
657 * the device time the source started at. Also subtract startpts to get
658 * the device time the stream would have started at to reach where it
661 if(mBufferDataSize
> 0)
663 nanoseconds startpts
{mCurrentPts
-
664 nanoseconds
{seconds
{readable
/mFrameSize
}}/mCodecCtx
->sample_rate
};
665 mDeviceStartTime
= device_time
- src_offset
- startpts
;
669 nanoseconds startpts
{mCurrentPts
- AudioBufferTotalTime
};
670 mDeviceStartTime
= device_time
- src_offset
- startpts
;
676 int AudioState::getSync()
678 if(mMovie
.mAVSyncType
== SyncMaster::Audio
)
681 auto ref_clock
= mMovie
.getMasterClock();
682 auto diff
= ref_clock
- getClockNoLock();
684 if(!(diff
< AVNoSyncThreshold
&& diff
> -AVNoSyncThreshold
))
686 /* Difference is TOO big; reset accumulated average */
687 mClockDiffAvg
= seconds_d64::zero();
691 /* Accumulate the diffs */
692 mClockDiffAvg
= mClockDiffAvg
*AudioAvgFilterCoeff
+ diff
;
693 auto avg_diff
= mClockDiffAvg
*(1.0 - AudioAvgFilterCoeff
);
694 if(avg_diff
< AudioSyncThreshold
/2.0 && avg_diff
> -AudioSyncThreshold
)
697 /* Constrain the per-update difference to avoid exceedingly large skips */
698 diff
= std::min
<nanoseconds
>(diff
, AudioSampleCorrectionMax
);
699 return static_cast<int>(duration_cast
<seconds
>(diff
*mCodecCtx
->sample_rate
).count());
702 int AudioState::decodeFrame()
705 while(int ret
{mQueue
.receiveFrame(mCodecCtx
.get(), mDecodedFrame
.get())})
707 if(ret
== AVErrorEOF
) return 0;
708 std::cerr
<< "Failed to receive frame: "<<ret
<<std::endl
;
710 } while(mDecodedFrame
->nb_samples
<= 0);
712 /* If provided, update w/ pts */
713 if(mDecodedFrame
->best_effort_timestamp
!= AVNoPtsValue
)
714 mCurrentPts
= duration_cast
<nanoseconds
>(seconds_d64
{av_q2d(mStream
->time_base
) *
715 static_cast<double>(mDecodedFrame
->best_effort_timestamp
)});
717 if(mDecodedFrame
->nb_samples
> mSamplesMax
)
720 av_samples_alloc(&mSamples
, nullptr, mCodecCtx
->channels
, mDecodedFrame
->nb_samples
,
722 mSamplesMax
= mDecodedFrame
->nb_samples
;
724 /* Return the amount of sample frames converted */
725 int data_size
{swr_convert(mSwresCtx
.get(), &mSamples
, mDecodedFrame
->nb_samples
,
726 const_cast<const uint8_t**>(mDecodedFrame
->data
), mDecodedFrame
->nb_samples
)};
728 av_frame_unref(mDecodedFrame
.get());
732 /* Duplicates the sample at in to out, count times. The frame size is a
733 * multiple of the template type size.
736 static void sample_dup(uint8_t *out
, const uint8_t *in
, size_t count
, size_t frame_size
)
738 auto *sample
= reinterpret_cast<const T
*>(in
);
739 auto *dst
= reinterpret_cast<T
*>(out
);
740 if(frame_size
== sizeof(T
))
741 std::fill_n(dst
, count
, *sample
);
744 /* NOTE: frame_size is a multiple of sizeof(T). */
745 size_t type_mult
{frame_size
/ sizeof(T
)};
747 std::generate_n(dst
, count
*type_mult
,
748 [sample
,type_mult
,&i
]() -> T
759 bool AudioState::readAudio(uint8_t *samples
, unsigned int length
, int &sample_skip
)
761 unsigned int audio_size
{0};
763 /* Read the next chunk of data, refill the buffer, and queue it
765 length
/= mFrameSize
;
766 while(mSamplesLen
> 0 && audio_size
< length
)
768 unsigned int rem
{length
- audio_size
};
771 const auto len
= static_cast<unsigned int>(mSamplesLen
- mSamplesPos
);
772 if(rem
> len
) rem
= len
;
773 std::copy_n(mSamples
+ static_cast<unsigned int>(mSamplesPos
)*mFrameSize
,
774 rem
*mFrameSize
, samples
);
778 rem
= std::min(rem
, static_cast<unsigned int>(-mSamplesPos
));
780 /* Add samples by copying the first sample */
781 if((mFrameSize
&7) == 0)
782 sample_dup
<uint64_t>(samples
, mSamples
, rem
, mFrameSize
);
783 else if((mFrameSize
&3) == 0)
784 sample_dup
<uint32_t>(samples
, mSamples
, rem
, mFrameSize
);
785 else if((mFrameSize
&1) == 0)
786 sample_dup
<uint16_t>(samples
, mSamples
, rem
, mFrameSize
);
788 sample_dup
<uint8_t>(samples
, mSamples
, rem
, mFrameSize
);
792 mCurrentPts
+= nanoseconds
{seconds
{rem
}} / mCodecCtx
->sample_rate
;
793 samples
+= rem
*mFrameSize
;
796 while(mSamplesPos
>= mSamplesLen
)
798 mSamplesLen
= decodeFrame();
799 mSamplesPos
= std::min(mSamplesLen
, sample_skip
);
800 if(mSamplesLen
<= 0) break;
802 sample_skip
-= mSamplesPos
;
804 // Adjust the device start time and current pts by the amount we're
805 // skipping/duplicating, so that the clock remains correct for the
806 // current stream position.
807 auto skip
= nanoseconds
{seconds
{mSamplesPos
}} / mCodecCtx
->sample_rate
;
808 mDeviceStartTime
-= skip
;
816 if(audio_size
< length
)
818 const unsigned int rem
{length
- audio_size
};
819 std::fill_n(samples
, rem
*mFrameSize
,
820 (mDstSampleFmt
== AV_SAMPLE_FMT_U8
) ? 0x80 : 0x00);
821 mCurrentPts
+= nanoseconds
{seconds
{rem
}} / mCodecCtx
->sample_rate
;
827 bool AudioState::readAudio(int sample_skip
)
829 size_t woffset
{mWritePos
.load(std::memory_order_acquire
)};
830 while(mSamplesLen
> 0)
832 const size_t roffset
{mReadPos
.load(std::memory_order_relaxed
)};
836 size_t rem
{(((roffset
> woffset
) ? roffset
-1
837 : ((roffset
== 0) ? mBufferDataSize
-1
838 : mBufferDataSize
)) - woffset
) / mFrameSize
};
839 rem
= std::min
<size_t>(rem
, static_cast<ALuint
>(-mSamplesPos
));
842 auto *splout
{&mBufferData
[woffset
]};
843 if((mFrameSize
&7) == 0)
844 sample_dup
<uint64_t>(splout
, mSamples
, rem
, mFrameSize
);
845 else if((mFrameSize
&3) == 0)
846 sample_dup
<uint32_t>(splout
, mSamples
, rem
, mFrameSize
);
847 else if((mFrameSize
&1) == 0)
848 sample_dup
<uint16_t>(splout
, mSamples
, rem
, mFrameSize
);
850 sample_dup
<uint8_t>(splout
, mSamples
, rem
, mFrameSize
);
851 woffset
+= rem
* mFrameSize
;
852 if(woffset
== mBufferDataSize
)
854 mWritePos
.store(woffset
, std::memory_order_release
);
855 mSamplesPos
+= static_cast<int>(rem
);
856 mCurrentPts
+= nanoseconds
{seconds
{rem
}} / mCodecCtx
->sample_rate
;
860 const size_t boffset
{static_cast<ALuint
>(mSamplesPos
) * size_t{mFrameSize
}};
861 const size_t nbytes
{static_cast<ALuint
>(mSamplesLen
)*size_t{mFrameSize
} -
863 if(roffset
> woffset
)
865 const size_t writable
{roffset
-woffset
-1};
866 if(writable
< nbytes
) break;
868 memcpy(&mBufferData
[woffset
], mSamples
+boffset
, nbytes
);
873 const size_t writable
{mBufferDataSize
+roffset
-woffset
-1};
874 if(writable
< nbytes
) break;
876 const size_t todo1
{std::min
<size_t>(nbytes
, mBufferDataSize
-woffset
)};
877 const size_t todo2
{nbytes
- todo1
};
879 memcpy(&mBufferData
[woffset
], mSamples
+boffset
, todo1
);
881 if(woffset
== mBufferDataSize
)
886 memcpy(&mBufferData
[woffset
], mSamples
+boffset
+todo1
, todo2
);
891 mWritePos
.store(woffset
, std::memory_order_release
);
892 mCurrentPts
+= nanoseconds
{seconds
{mSamplesLen
-mSamplesPos
}} / mCodecCtx
->sample_rate
;
895 mSamplesLen
= decodeFrame();
896 mSamplesPos
= std::min(mSamplesLen
, sample_skip
);
897 if(mSamplesLen
<= 0) return false;
899 sample_skip
-= mSamplesPos
;
901 auto skip
= nanoseconds
{seconds
{mSamplesPos
}} / mCodecCtx
->sample_rate
;
902 mDeviceStartTime
-= skip
;
904 } while(mSamplesPos
>= mSamplesLen
);
911 void AL_APIENTRY
AudioState::eventCallback(ALenum eventType
, ALuint object
, ALuint param
,
912 ALsizei length
, const ALchar
*message
)
914 if(eventType
== AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT
)
916 /* Temporarily lock the source mutex to ensure it's not between
917 * checking the processed count and going to sleep.
919 std::unique_lock
<std::mutex
>{mSrcMutex
}.unlock();
920 mSrcCond
.notify_one();
924 std::cout
<< "\n---- AL Event on AudioState "<<this<<" ----\nEvent: ";
927 case AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT
: std::cout
<< "Buffer completed"; break;
928 case AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT
: std::cout
<< "Source state changed"; break;
929 case AL_EVENT_TYPE_DISCONNECTED_SOFT
: std::cout
<< "Disconnected"; break;
931 std::cout
<< "0x"<<std::hex
<<std::setw(4)<<std::setfill('0')<<eventType
<<std::dec
<<
932 std::setw(0)<<std::setfill(' '); break;
935 "Object ID: "<<object
<<"\n"
936 "Parameter: "<<param
<<"\n"
937 "Message: "<<std::string
{message
, static_cast<ALuint
>(length
)}<<"\n----"<<
940 if(eventType
== AL_EVENT_TYPE_DISCONNECTED_SOFT
)
943 std::lock_guard
<std::mutex
> lock
{mSrcMutex
};
944 mConnected
.clear(std::memory_order_release
);
946 mSrcCond
.notify_one();
950 #ifdef AL_SOFT_callback_buffer
951 ALsizei
AudioState::bufferCallback(void *data
, ALsizei size
)
955 size_t roffset
{mReadPos
.load(std::memory_order_acquire
)};
958 const size_t woffset
{mWritePos
.load(std::memory_order_relaxed
)};
959 if(woffset
== roffset
) break;
961 size_t todo
{((woffset
< roffset
) ? mBufferDataSize
: woffset
) - roffset
};
962 todo
= std::min
<size_t>(todo
, static_cast<ALuint
>(size
-got
));
964 memcpy(data
, &mBufferData
[roffset
], todo
);
965 data
= static_cast<ALbyte
*>(data
) + todo
;
966 got
+= static_cast<ALsizei
>(todo
);
969 if(roffset
== mBufferDataSize
)
972 mReadPos
.store(roffset
, std::memory_order_release
);
978 int AudioState::handler()
980 std::unique_lock
<std::mutex
> srclock
{mSrcMutex
, std::defer_lock
};
981 milliseconds sleep_time
{AudioBufferTime
/ 3};
983 struct EventControlManager
{
984 const std::array
<ALenum
,3> evt_types
{{
985 AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT
, AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT
,
986 AL_EVENT_TYPE_DISCONNECTED_SOFT
}};
988 EventControlManager(milliseconds
&sleep_time
)
990 if(alEventControlSOFT
)
992 alEventControlSOFT(static_cast<ALsizei
>(evt_types
.size()), evt_types
.data(),
994 alEventCallbackSOFT(&AudioState::eventCallbackC
, this);
995 sleep_time
= AudioBufferTotalTime
;
998 ~EventControlManager()
1000 if(alEventControlSOFT
)
1002 alEventControlSOFT(static_cast<ALsizei
>(evt_types
.size()), evt_types
.data(),
1004 alEventCallbackSOFT(nullptr, nullptr);
1008 EventControlManager event_controller
{sleep_time
};
1010 const bool has_bfmt_ex
{alIsExtensionPresent("AL_SOFT_bformat_ex") != AL_FALSE
};
1011 ALenum ambi_layout
{AL_FUMA_SOFT
};
1012 ALenum ambi_scale
{AL_FUMA_SOFT
};
1014 std::unique_ptr
<uint8_t[]> samples
;
1015 ALsizei buffer_len
{0};
1017 /* Find a suitable format for OpenAL. */
1020 if((mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_FLT
|| mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_FLTP
1021 || mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_DBL
1022 || mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_DBLP
1023 || mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_S32
1024 || mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_S32P
1025 || mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_S64
1026 || mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_S64P
)
1027 && alIsExtensionPresent("AL_EXT_FLOAT32"))
1029 mDstSampleFmt
= AV_SAMPLE_FMT_FLT
;
1031 if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
1033 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_7POINT1
)
1035 mDstChanLayout
= mCodecCtx
->channel_layout
;
1037 mFormat
= alGetEnumValue("AL_FORMAT_71CHN32");
1039 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_5POINT1
1040 || mCodecCtx
->channel_layout
== AV_CH_LAYOUT_5POINT1_BACK
)
1042 mDstChanLayout
= mCodecCtx
->channel_layout
;
1044 mFormat
= alGetEnumValue("AL_FORMAT_51CHN32");
1046 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_QUAD
)
1048 mDstChanLayout
= mCodecCtx
->channel_layout
;
1050 mFormat
= alGetEnumValue("AL_FORMAT_QUAD32");
1053 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_MONO
)
1055 mDstChanLayout
= mCodecCtx
->channel_layout
;
1057 mFormat
= AL_FORMAT_MONO_FLOAT32
;
1059 /* Assume 3D B-Format (ambisonics) if the channel layout is blank and
1060 * there's 4 or more channels. FFmpeg/libavcodec otherwise seems to
1061 * have no way to specify if the source is actually B-Format (let alone
1062 * if it's 2D or 3D).
1064 if(mCodecCtx
->channel_layout
== 0 && mCodecCtx
->channels
>= 4
1065 && alIsExtensionPresent("AL_EXT_BFORMAT"))
1067 /* Calculate what should be the ambisonic order from the number of
1068 * channels, and confirm that's the number of channels. Opus allows
1069 * an optional non-diegetic stereo stream with the B-Format stream,
1070 * which we can ignore, so check for that too.
1072 auto order
= static_cast<int>(std::sqrt(mCodecCtx
->channels
)) - 1;
1073 int channels
{(order
+1) * (order
+1)};
1074 if(channels
== mCodecCtx
->channels
|| channels
+2 == mCodecCtx
->channels
)
1076 /* OpenAL only supports first-order with AL_EXT_BFORMAT, which
1077 * is 4 channels for 3D buffers.
1080 mFormat
= alGetEnumValue("AL_FORMAT_BFORMAT3D_FLOAT32");
1083 if(!mFormat
|| mFormat
== -1)
1085 mDstChanLayout
= AV_CH_LAYOUT_STEREO
;
1087 mFormat
= FormatStereo32F
;
1090 if(mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_U8
|| mCodecCtx
->sample_fmt
== AV_SAMPLE_FMT_U8P
)
1092 mDstSampleFmt
= AV_SAMPLE_FMT_U8
;
1094 if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
1096 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_7POINT1
)
1098 mDstChanLayout
= mCodecCtx
->channel_layout
;
1100 mFormat
= alGetEnumValue("AL_FORMAT_71CHN8");
1102 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_5POINT1
1103 || mCodecCtx
->channel_layout
== AV_CH_LAYOUT_5POINT1_BACK
)
1105 mDstChanLayout
= mCodecCtx
->channel_layout
;
1107 mFormat
= alGetEnumValue("AL_FORMAT_51CHN8");
1109 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_QUAD
)
1111 mDstChanLayout
= mCodecCtx
->channel_layout
;
1113 mFormat
= alGetEnumValue("AL_FORMAT_QUAD8");
1116 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_MONO
)
1118 mDstChanLayout
= mCodecCtx
->channel_layout
;
1120 mFormat
= AL_FORMAT_MONO8
;
1122 if(mCodecCtx
->channel_layout
== 0 && mCodecCtx
->channels
>= 4
1123 && alIsExtensionPresent("AL_EXT_BFORMAT"))
1125 auto order
= static_cast<int>(std::sqrt(mCodecCtx
->channels
)) - 1;
1126 int channels
{(order
+1) * (order
+1)};
1127 if(channels
== mCodecCtx
->channels
|| channels
+2 == mCodecCtx
->channels
)
1130 mFormat
= alGetEnumValue("AL_FORMAT_BFORMAT3D_8");
1133 if(!mFormat
|| mFormat
== -1)
1135 mDstChanLayout
= AV_CH_LAYOUT_STEREO
;
1137 mFormat
= FormatStereo8
;
1140 if(!mFormat
|| mFormat
== -1)
1142 mDstSampleFmt
= AV_SAMPLE_FMT_S16
;
1144 if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
1146 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_7POINT1
)
1148 mDstChanLayout
= mCodecCtx
->channel_layout
;
1150 mFormat
= alGetEnumValue("AL_FORMAT_71CHN16");
1152 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_5POINT1
1153 || mCodecCtx
->channel_layout
== AV_CH_LAYOUT_5POINT1_BACK
)
1155 mDstChanLayout
= mCodecCtx
->channel_layout
;
1157 mFormat
= alGetEnumValue("AL_FORMAT_51CHN16");
1159 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_QUAD
)
1161 mDstChanLayout
= mCodecCtx
->channel_layout
;
1163 mFormat
= alGetEnumValue("AL_FORMAT_QUAD16");
1166 if(mCodecCtx
->channel_layout
== AV_CH_LAYOUT_MONO
)
1168 mDstChanLayout
= mCodecCtx
->channel_layout
;
1170 mFormat
= AL_FORMAT_MONO16
;
1172 if(mCodecCtx
->channel_layout
== 0 && mCodecCtx
->channels
>= 4
1173 && alIsExtensionPresent("AL_EXT_BFORMAT"))
1175 auto order
= static_cast<int>(std::sqrt(mCodecCtx
->channels
)) - 1;
1176 int channels
{(order
+1) * (order
+1)};
1177 if(channels
== mCodecCtx
->channels
|| channels
+2 == mCodecCtx
->channels
)
1180 mFormat
= alGetEnumValue("AL_FORMAT_BFORMAT3D_16");
1183 if(!mFormat
|| mFormat
== -1)
1185 mDstChanLayout
= AV_CH_LAYOUT_STEREO
;
1187 mFormat
= FormatStereo16
;
1196 mDecodedFrame
.reset(av_frame_alloc());
1199 std::cerr
<< "Failed to allocate audio frame" <<std::endl
;
1205 /* OpenAL only supports first-order ambisonics with AL_EXT_BFORMAT, so
1206 * we have to drop any extra channels.
1208 mSwresCtx
.reset(swr_alloc_set_opts(nullptr,
1209 (1_i64
<<4)-1, mDstSampleFmt
, mCodecCtx
->sample_rate
,
1210 (1_i64
<<mCodecCtx
->channels
)-1, mCodecCtx
->sample_fmt
, mCodecCtx
->sample_rate
,
1213 /* Note that ffmpeg/libavcodec has no method to check the ambisonic
1214 * channel order and normalization, so we can only assume AmbiX as the
1215 * defacto-standard. This is not true for .amb files, which use FuMa.
1217 std::vector
<double> mtx(64*64, 0.0);
1218 ambi_layout
= AL_ACN_SOFT
;
1219 ambi_scale
= AL_SN3D_SOFT
;
1222 /* An identity matrix that doesn't remix any channels. */
1223 std::cout
<< "Found AL_SOFT_bformat_ex" <<std::endl
;
1224 mtx
[0 + 0*64] = 1.0;
1225 mtx
[1 + 1*64] = 1.0;
1226 mtx
[2 + 2*64] = 1.0;
1227 mtx
[3 + 3*64] = 1.0;
1231 std::cout
<< "Found AL_EXT_BFORMAT" <<std::endl
;
1232 /* Without AL_SOFT_bformat_ex, OpenAL only supports FuMa channel
1233 * ordering and normalization, so a custom matrix is needed to
1234 * scale and reorder the source from AmbiX.
1236 mtx
[0 + 0*64] = std::sqrt(0.5);
1237 mtx
[3 + 1*64] = 1.0;
1238 mtx
[1 + 2*64] = 1.0;
1239 mtx
[2 + 3*64] = 1.0;
1241 swr_set_matrix(mSwresCtx
.get(), mtx
.data(), 64);
1244 mSwresCtx
.reset(swr_alloc_set_opts(nullptr,
1245 static_cast<int64_t>(mDstChanLayout
), mDstSampleFmt
, mCodecCtx
->sample_rate
,
1246 mCodecCtx
->channel_layout
? static_cast<int64_t>(mCodecCtx
->channel_layout
)
1247 : av_get_default_channel_layout(mCodecCtx
->channels
),
1248 mCodecCtx
->sample_fmt
, mCodecCtx
->sample_rate
,
1250 if(!mSwresCtx
|| swr_init(mSwresCtx
.get()) != 0)
1252 std::cerr
<< "Failed to initialize audio converter" <<std::endl
;
1256 alGenBuffers(static_cast<ALsizei
>(mBuffers
.size()), mBuffers
.data());
1257 alGenSources(1, &mSource
);
1260 alSourcei(mSource
, AL_DIRECT_CHANNELS_SOFT
, DirectOutMode
);
1261 if(EnableWideStereo
)
1263 const float angles
[2]{static_cast<float>(M_PI
/ 3.0), static_cast<float>(-M_PI
/ 3.0)};
1264 alSourcefv(mSource
, AL_STEREO_ANGLES
, angles
);
1268 for(ALuint bufid
: mBuffers
)
1270 alBufferi(bufid
, AL_AMBISONIC_LAYOUT_SOFT
, ambi_layout
);
1271 alBufferi(bufid
, AL_AMBISONIC_SCALING_SOFT
, ambi_scale
);
1275 if(EnableSuperStereo
)
1276 alSourcei(mSource
, AL_STEREO_MODE_SOFT
, AL_SUPER_STEREO_SOFT
);
1279 if(alGetError() != AL_NO_ERROR
)
1282 #ifdef AL_SOFT_callback_buffer
1283 bool callback_ok
{false};
1284 if(alBufferCallbackSOFT
)
1286 alBufferCallbackSOFT(mBuffers
[0], mFormat
, mCodecCtx
->sample_rate
, bufferCallbackC
, this,
1288 alSourcei(mSource
, AL_BUFFER
, static_cast<ALint
>(mBuffers
[0]));
1289 if(alGetError() != AL_NO_ERROR
)
1291 fprintf(stderr
, "Failed to set buffer callback\n");
1292 alSourcei(mSource
, AL_BUFFER
, 0);
1296 mBufferDataSize
= static_cast<size_t>(duration_cast
<seconds
>(mCodecCtx
->sample_rate
*
1297 AudioBufferTotalTime
).count()) * mFrameSize
;
1298 mBufferData
= std::make_unique
<uint8_t[]>(mBufferDataSize
);
1299 std::fill_n(mBufferData
.get(), mBufferDataSize
, uint8_t{});
1301 mReadPos
.store(0, std::memory_order_relaxed
);
1302 mWritePos
.store(mBufferDataSize
/mFrameSize
/2*mFrameSize
, std::memory_order_relaxed
);
1305 alcGetIntegerv(alcGetContextsDevice(alcGetCurrentContext()), ALC_REFRESH
, 1, &refresh
);
1306 sleep_time
= milliseconds
{seconds
{1}} / refresh
;
1312 buffer_len
= static_cast<int>(duration_cast
<seconds
>(mCodecCtx
->sample_rate
*
1313 AudioBufferTime
).count() * mFrameSize
);
1315 samples
= std::make_unique
<uint8_t[]>(static_cast<ALuint
>(buffer_len
));
1317 /* Prefill the codec buffer. */
1318 auto packet_sender
= [this]()
1322 const int ret
{mQueue
.sendPacket(mCodecCtx
.get())};
1323 if(ret
== AVErrorEOF
) break;
1326 auto sender
= std::async(std::launch::async
, packet_sender
);
1329 if(alcGetInteger64vSOFT
)
1332 alcGetInteger64vSOFT(alcGetContextsDevice(alcGetCurrentContext()), ALC_DEVICE_CLOCK_SOFT
,
1334 mDeviceStartTime
= nanoseconds
{devtime
} - mCurrentPts
;
1337 mSamplesLen
= decodeFrame();
1340 mSamplesPos
= std::min(mSamplesLen
, getSync());
1342 auto skip
= nanoseconds
{seconds
{mSamplesPos
}} / mCodecCtx
->sample_rate
;
1343 mDeviceStartTime
-= skip
;
1344 mCurrentPts
+= skip
;
1350 if(mBufferDataSize
> 0)
1352 alGetSourcei(mSource
, AL_SOURCE_STATE
, &state
);
1353 /* If mQuit is set, don't actually quit until we can't get more
1354 * audio, indicating we've reached the flush packet and the packet
1355 * sender will also quit.
1357 * If mQuit is not set, don't quit even if there's no more audio,
1358 * so what's buffered has a chance to play to the real end.
1360 if(!readAudio(getSync()) && mMovie
.mQuit
.load(std::memory_order_relaxed
))
1365 ALint processed
, queued
;
1367 /* First remove any processed buffers. */
1368 alGetSourcei(mSource
, AL_BUFFERS_PROCESSED
, &processed
);
1369 while(processed
> 0)
1372 alSourceUnqueueBuffers(mSource
, 1, &bid
);
1376 /* Refill the buffer queue. */
1377 int sync_skip
{getSync()};
1378 alGetSourcei(mSource
, AL_BUFFERS_QUEUED
, &queued
);
1379 while(static_cast<ALuint
>(queued
) < mBuffers
.size())
1381 /* Read the next chunk of data, filling the buffer, and queue
1384 const bool got_audio
{readAudio(samples
.get(), static_cast<ALuint
>(buffer_len
),
1388 if(mMovie
.mQuit
.load(std::memory_order_relaxed
))
1393 const ALuint bufid
{mBuffers
[mBufferIdx
]};
1394 mBufferIdx
= static_cast<ALuint
>((mBufferIdx
+1) % mBuffers
.size());
1396 alBufferData(bufid
, mFormat
, samples
.get(), buffer_len
, mCodecCtx
->sample_rate
);
1397 alSourceQueueBuffers(mSource
, 1, &bufid
);
1401 /* Check that the source is playing. */
1402 alGetSourcei(mSource
, AL_SOURCE_STATE
, &state
);
1403 if(state
== AL_STOPPED
)
1405 /* AL_STOPPED means there was an underrun. Clear the buffer
1406 * queue since this likely means we're late, and rewind the
1407 * source to get it back into an AL_INITIAL state.
1409 alSourceRewind(mSource
);
1410 alSourcei(mSource
, AL_BUFFER
, 0);
1411 if(alcGetInteger64vSOFT
)
1413 /* Also update the device start time with the current
1414 * device clock, so the decoder knows we're running behind.
1417 alcGetInteger64vSOFT(alcGetContextsDevice(alcGetCurrentContext()),
1418 ALC_DEVICE_CLOCK_SOFT
, 1, &devtime
);
1419 mDeviceStartTime
= nanoseconds
{devtime
} - mCurrentPts
;
1425 /* (re)start the source if needed, and wait for a buffer to finish */
1426 if(state
!= AL_PLAYING
&& state
!= AL_PAUSED
)
1428 if(!startPlayback())
1431 if(ALenum err
{alGetError()})
1432 std::cerr
<< "Got AL error: 0x"<<std::hex
<<err
<<std::dec
1433 << " ("<<alGetString(err
)<<")" <<std::endl
;
1435 mSrcCond
.wait_for(srclock
, sleep_time
);
1439 alSourceRewind(mSource
);
1440 alSourcei(mSource
, AL_BUFFER
, 0);
1447 nanoseconds
VideoState::getClock()
1449 /* NOTE: This returns incorrect times while not playing. */
1450 std::lock_guard
<std::mutex
> _
{mDispPtsMutex
};
1451 if(mDisplayPtsTime
== microseconds::min())
1452 return nanoseconds::zero();
1453 auto delta
= get_avtime() - mDisplayPtsTime
;
1454 return mDisplayPts
+ delta
;
1457 /* Called by VideoState::updateVideo to display the next video frame. */
1458 void VideoState::display(SDL_Window
*screen
, SDL_Renderer
*renderer
, AVFrame
*frame
)
1463 double aspect_ratio
;
1467 int frame_width
{frame
->width
- static_cast<int>(frame
->crop_left
+ frame
->crop_right
)};
1468 int frame_height
{frame
->height
- static_cast<int>(frame
->crop_top
+ frame
->crop_bottom
)};
1469 if(frame
->sample_aspect_ratio
.num
== 0)
1473 aspect_ratio
= av_q2d(frame
->sample_aspect_ratio
) * frame_width
/
1476 if(aspect_ratio
<= 0.0)
1477 aspect_ratio
= static_cast<double>(frame_width
) / frame_height
;
1479 SDL_GetWindowSize(screen
, &win_w
, &win_h
);
1481 w
= (static_cast<int>(std::rint(h
* aspect_ratio
)) + 3) & ~3;
1485 h
= (static_cast<int>(std::rint(w
/ aspect_ratio
)) + 3) & ~3;
1487 x
= (win_w
- w
) / 2;
1488 y
= (win_h
- h
) / 2;
1490 SDL_Rect src_rect
{ static_cast<int>(frame
->crop_left
), static_cast<int>(frame
->crop_top
),
1491 frame_width
, frame_height
};
1492 SDL_Rect dst_rect
{ x
, y
, w
, h
};
1493 SDL_RenderCopy(renderer
, mImage
, &src_rect
, &dst_rect
);
1494 SDL_RenderPresent(renderer
);
1497 /* Called regularly on the main thread where the SDL_Renderer was created. It
1498 * handles updating the textures of decoded frames and displaying the latest
1501 void VideoState::updateVideo(SDL_Window
*screen
, SDL_Renderer
*renderer
, bool redraw
)
1503 size_t read_idx
{mPictQRead
.load(std::memory_order_relaxed
)};
1504 Picture
*vp
{&mPictQ
[read_idx
]};
1506 auto clocktime
= mMovie
.getMasterClock();
1507 bool updated
{false};
1510 size_t next_idx
{(read_idx
+1)%mPictQ
.size()};
1511 if(next_idx
== mPictQWrite
.load(std::memory_order_acquire
))
1513 Picture
*nextvp
{&mPictQ
[next_idx
]};
1514 if(clocktime
< nextvp
->mPts
&& !mMovie
.mQuit
.load(std::memory_order_relaxed
))
1516 /* For the first update, ensure the first frame gets shown. */
1517 if(!mFirstUpdate
|| updated
)
1523 read_idx
= next_idx
;
1525 if(mMovie
.mQuit
.load(std::memory_order_relaxed
))
1528 mFinalUpdate
= true;
1529 mPictQRead
.store(read_idx
, std::memory_order_release
);
1530 std::unique_lock
<std::mutex
>{mPictQMutex
}.unlock();
1531 mPictQCond
.notify_one();
1535 AVFrame
*frame
{vp
->mFrame
.get()};
1538 mPictQRead
.store(read_idx
, std::memory_order_release
);
1539 std::unique_lock
<std::mutex
>{mPictQMutex
}.unlock();
1540 mPictQCond
.notify_one();
1542 /* allocate or resize the buffer! */
1543 bool fmt_updated
{false};
1544 if(!mImage
|| mWidth
!= frame
->width
|| mHeight
!= frame
->height
)
1548 SDL_DestroyTexture(mImage
);
1549 mImage
= SDL_CreateTexture(renderer
, SDL_PIXELFORMAT_IYUV
, SDL_TEXTUREACCESS_STREAMING
,
1550 frame
->width
, frame
->height
);
1552 std::cerr
<< "Failed to create YV12 texture!" <<std::endl
;
1553 mWidth
= frame
->width
;
1554 mHeight
= frame
->height
;
1557 int frame_width
{frame
->width
- static_cast<int>(frame
->crop_left
+ frame
->crop_right
)};
1558 int frame_height
{frame
->height
- static_cast<int>(frame
->crop_top
+ frame
->crop_bottom
)};
1559 if(mFirstUpdate
&& frame_width
> 0 && frame_height
> 0)
1561 /* For the first update, set the window size to the video size. */
1562 mFirstUpdate
= false;
1564 if(frame
->sample_aspect_ratio
.den
!= 0)
1566 double aspect_ratio
= av_q2d(frame
->sample_aspect_ratio
);
1567 if(aspect_ratio
>= 1.0)
1568 frame_width
= static_cast<int>(frame_width
*aspect_ratio
+ 0.5);
1569 else if(aspect_ratio
> 0.0)
1570 frame_height
= static_cast<int>(frame_height
/aspect_ratio
+ 0.5);
1572 SDL_SetWindowSize(screen
, frame_width
, frame_height
);
1577 void *pixels
{nullptr};
1580 if(mCodecCtx
->pix_fmt
== AV_PIX_FMT_YUV420P
)
1581 SDL_UpdateYUVTexture(mImage
, nullptr,
1582 frame
->data
[0], frame
->linesize
[0],
1583 frame
->data
[1], frame
->linesize
[1],
1584 frame
->data
[2], frame
->linesize
[2]
1586 else if(SDL_LockTexture(mImage
, nullptr, &pixels
, &pitch
) != 0)
1587 std::cerr
<< "Failed to lock texture" <<std::endl
;
1590 // Convert the image into YUV format that SDL uses
1591 int w
{frame
->width
};
1592 int h
{frame
->height
};
1593 if(!mSwscaleCtx
|| fmt_updated
)
1595 mSwscaleCtx
.reset(sws_getContext(
1596 w
, h
, mCodecCtx
->pix_fmt
,
1597 w
, h
, AV_PIX_FMT_YUV420P
, 0,
1598 nullptr, nullptr, nullptr
1602 /* point pict at the queue */
1603 uint8_t *pict_data
[3];
1604 pict_data
[0] = static_cast<uint8_t*>(pixels
);
1605 pict_data
[1] = pict_data
[0] + w
*h
;
1606 pict_data
[2] = pict_data
[1] + w
*h
/4;
1608 int pict_linesize
[3];
1609 pict_linesize
[0] = pitch
;
1610 pict_linesize
[1] = pitch
/ 2;
1611 pict_linesize
[2] = pitch
/ 2;
1613 sws_scale(mSwscaleCtx
.get(), reinterpret_cast<uint8_t**>(frame
->data
), frame
->linesize
,
1614 0, h
, pict_data
, pict_linesize
);
1615 SDL_UnlockTexture(mImage
);
1624 /* Show the picture! */
1625 display(screen
, renderer
, frame
);
1630 auto disp_time
= get_avtime();
1632 std::lock_guard
<std::mutex
> _
{mDispPtsMutex
};
1633 mDisplayPts
= vp
->mPts
;
1634 mDisplayPtsTime
= disp_time
;
1636 if(mEOS
.load(std::memory_order_acquire
))
1638 if((read_idx
+1)%mPictQ
.size() == mPictQWrite
.load(std::memory_order_acquire
))
1640 mFinalUpdate
= true;
1641 std::unique_lock
<std::mutex
>{mPictQMutex
}.unlock();
1642 mPictQCond
.notify_one();
1647 int VideoState::handler()
1649 std::for_each(mPictQ
.begin(), mPictQ
.end(),
1650 [](Picture
&pict
) -> void
1651 { pict
.mFrame
= AVFramePtr
{av_frame_alloc()}; });
1653 /* Prefill the codec buffer. */
1654 auto packet_sender
= [this]()
1658 const int ret
{mQueue
.sendPacket(mCodecCtx
.get())};
1659 if(ret
== AVErrorEOF
) break;
1662 auto sender
= std::async(std::launch::async
, packet_sender
);
1665 std::lock_guard
<std::mutex
> _
{mDispPtsMutex
};
1666 mDisplayPtsTime
= get_avtime();
1669 auto current_pts
= nanoseconds::zero();
1672 size_t write_idx
{mPictQWrite
.load(std::memory_order_relaxed
)};
1673 Picture
*vp
{&mPictQ
[write_idx
]};
1675 /* Retrieve video frame. */
1676 AVFrame
*decoded_frame
{vp
->mFrame
.get()};
1677 while(int ret
{mQueue
.receiveFrame(mCodecCtx
.get(), decoded_frame
)})
1679 if(ret
== AVErrorEOF
) goto finish
;
1680 std::cerr
<< "Failed to receive frame: "<<ret
<<std::endl
;
1683 /* Get the PTS for this frame. */
1684 if(decoded_frame
->best_effort_timestamp
!= AVNoPtsValue
)
1685 current_pts
= duration_cast
<nanoseconds
>(seconds_d64
{av_q2d(mStream
->time_base
) *
1686 static_cast<double>(decoded_frame
->best_effort_timestamp
)});
1687 vp
->mPts
= current_pts
;
1689 /* Update the video clock to the next expected PTS. */
1690 auto frame_delay
= av_q2d(mCodecCtx
->time_base
);
1691 frame_delay
+= decoded_frame
->repeat_pict
* (frame_delay
* 0.5);
1692 current_pts
+= duration_cast
<nanoseconds
>(seconds_d64
{frame_delay
});
1694 /* Put the frame in the queue to be loaded into a texture and displayed
1695 * by the rendering thread.
1697 write_idx
= (write_idx
+1)%mPictQ
.size();
1698 mPictQWrite
.store(write_idx
, std::memory_order_release
);
1700 if(write_idx
== mPictQRead
.load(std::memory_order_acquire
))
1702 /* Wait until we have space for a new pic */
1703 std::unique_lock
<std::mutex
> lock
{mPictQMutex
};
1704 while(write_idx
== mPictQRead
.load(std::memory_order_acquire
))
1705 mPictQCond
.wait(lock
);
1711 std::unique_lock
<std::mutex
> lock
{mPictQMutex
};
1712 while(!mFinalUpdate
) mPictQCond
.wait(lock
);
1718 int MovieState::decode_interrupt_cb(void *ctx
)
1720 return static_cast<MovieState
*>(ctx
)->mQuit
.load(std::memory_order_relaxed
);
1723 bool MovieState::prepare()
1725 AVIOContext
*avioctx
{nullptr};
1726 AVIOInterruptCB intcb
{decode_interrupt_cb
, this};
1727 if(avio_open2(&avioctx
, mFilename
.c_str(), AVIO_FLAG_READ
, &intcb
, nullptr))
1729 std::cerr
<< "Failed to open "<<mFilename
<<std::endl
;
1732 mIOContext
.reset(avioctx
);
1734 /* Open movie file. If avformat_open_input fails it will automatically free
1735 * this context, so don't set it onto a smart pointer yet.
1737 AVFormatContext
*fmtctx
{avformat_alloc_context()};
1738 fmtctx
->pb
= mIOContext
.get();
1739 fmtctx
->interrupt_callback
= intcb
;
1740 if(avformat_open_input(&fmtctx
, mFilename
.c_str(), nullptr, nullptr) != 0)
1742 std::cerr
<< "Failed to open "<<mFilename
<<std::endl
;
1745 mFormatCtx
.reset(fmtctx
);
1747 /* Retrieve stream information */
1748 if(avformat_find_stream_info(mFormatCtx
.get(), nullptr) < 0)
1750 std::cerr
<< mFilename
<<": failed to find stream info" <<std::endl
;
1754 /* Dump information about file onto standard error */
1755 av_dump_format(mFormatCtx
.get(), 0, mFilename
.c_str(), 0);
1757 mParseThread
= std::thread
{std::mem_fn(&MovieState::parse_handler
), this};
1759 std::unique_lock
<std::mutex
> slock
{mStartupMutex
};
1760 while(!mStartupDone
) mStartupCond
.wait(slock
);
1764 void MovieState::setTitle(SDL_Window
*window
)
1766 auto pos1
= mFilename
.rfind('/');
1767 auto pos2
= mFilename
.rfind('\\');
1768 auto fpos
= ((pos1
== std::string::npos
) ? pos2
:
1769 (pos2
== std::string::npos
) ? pos1
:
1770 std::max(pos1
, pos2
)) + 1;
1771 SDL_SetWindowTitle(window
, (mFilename
.substr(fpos
)+" - "+AppName
).c_str());
1774 nanoseconds
MovieState::getClock()
1776 if(mClockBase
== microseconds::min())
1777 return nanoseconds::zero();
1778 return get_avtime() - mClockBase
;
1781 nanoseconds
MovieState::getMasterClock()
1783 if(mAVSyncType
== SyncMaster::Video
&& mVideo
.mStream
)
1784 return mVideo
.getClock();
1785 if(mAVSyncType
== SyncMaster::Audio
&& mAudio
.mStream
)
1786 return mAudio
.getClock();
1790 nanoseconds
MovieState::getDuration()
1791 { return std::chrono::duration
<int64_t,std::ratio
<1,AV_TIME_BASE
>>(mFormatCtx
->duration
); }
1793 int MovieState::streamComponentOpen(unsigned int stream_index
)
1795 if(stream_index
>= mFormatCtx
->nb_streams
)
1798 /* Get a pointer to the codec context for the stream, and open the
1801 AVCodecCtxPtr avctx
{avcodec_alloc_context3(nullptr)};
1802 if(!avctx
) return -1;
1804 if(avcodec_parameters_to_context(avctx
.get(), mFormatCtx
->streams
[stream_index
]->codecpar
))
1807 const AVCodec
*codec
{avcodec_find_decoder(avctx
->codec_id
)};
1808 if(!codec
|| avcodec_open2(avctx
.get(), codec
, nullptr) < 0)
1810 std::cerr
<< "Unsupported codec: "<<avcodec_get_name(avctx
->codec_id
)
1811 << " (0x"<<std::hex
<<avctx
->codec_id
<<std::dec
<<")" <<std::endl
;
1815 /* Initialize and start the media type handler */
1816 switch(avctx
->codec_type
)
1818 case AVMEDIA_TYPE_AUDIO
:
1819 mAudio
.mStream
= mFormatCtx
->streams
[stream_index
];
1820 mAudio
.mCodecCtx
= std::move(avctx
);
1823 case AVMEDIA_TYPE_VIDEO
:
1824 mVideo
.mStream
= mFormatCtx
->streams
[stream_index
];
1825 mVideo
.mCodecCtx
= std::move(avctx
);
1832 return static_cast<int>(stream_index
);
1835 int MovieState::parse_handler()
1837 auto &audio_queue
= mAudio
.mQueue
;
1838 auto &video_queue
= mVideo
.mQueue
;
1840 int video_index
{-1};
1841 int audio_index
{-1};
1843 /* Find the first video and audio streams */
1844 for(unsigned int i
{0u};i
< mFormatCtx
->nb_streams
;i
++)
1846 auto codecpar
= mFormatCtx
->streams
[i
]->codecpar
;
1847 if(codecpar
->codec_type
== AVMEDIA_TYPE_VIDEO
&& !DisableVideo
&& video_index
< 0)
1848 video_index
= streamComponentOpen(i
);
1849 else if(codecpar
->codec_type
== AVMEDIA_TYPE_AUDIO
&& audio_index
< 0)
1850 audio_index
= streamComponentOpen(i
);
1854 std::unique_lock
<std::mutex
> slock
{mStartupMutex
};
1855 mStartupDone
= true;
1857 mStartupCond
.notify_all();
1859 if(video_index
< 0 && audio_index
< 0)
1861 std::cerr
<< mFilename
<<": could not open codecs" <<std::endl
;
1865 /* Set the base time 750ms ahead of the current av time. */
1866 mClockBase
= get_avtime() + milliseconds
{750};
1868 if(audio_index
>= 0)
1869 mAudioThread
= std::thread
{std::mem_fn(&AudioState::handler
), &mAudio
};
1870 if(video_index
>= 0)
1871 mVideoThread
= std::thread
{std::mem_fn(&VideoState::handler
), &mVideo
};
1873 /* Main packet reading/dispatching loop */
1874 AVPacketPtr packet
{av_packet_alloc()};
1875 while(!mQuit
.load(std::memory_order_relaxed
))
1877 if(av_read_frame(mFormatCtx
.get(), packet
.get()) < 0)
1880 /* Copy the packet into the queue it's meant for. */
1881 if(packet
->stream_index
== video_index
)
1883 while(!mQuit
.load(std::memory_order_acquire
) && !video_queue
.put(packet
.get()))
1884 std::this_thread::sleep_for(milliseconds
{100});
1886 else if(packet
->stream_index
== audio_index
)
1888 while(!mQuit
.load(std::memory_order_acquire
) && !audio_queue
.put(packet
.get()))
1889 std::this_thread::sleep_for(milliseconds
{100});
1892 av_packet_unref(packet
.get());
1894 /* Finish the queues so the receivers know nothing more is coming. */
1895 video_queue
.setFinished();
1896 audio_queue
.setFinished();
1898 /* all done - wait for it */
1899 if(mVideoThread
.joinable())
1900 mVideoThread
.join();
1901 if(mAudioThread
.joinable())
1902 mAudioThread
.join();
1905 std::unique_lock
<std::mutex
> lock
{mVideo
.mPictQMutex
};
1906 while(!mVideo
.mFinalUpdate
)
1907 mVideo
.mPictQCond
.wait(lock
);
1911 evt
.user
.type
= FF_MOVIE_DONE_EVENT
;
1912 SDL_PushEvent(&evt
);
1917 void MovieState::stop()
1920 mAudio
.mQueue
.flush();
1921 mVideo
.mQueue
.flush();
1925 // Helper class+method to print the time with human-readable formatting.
1929 std::ostream
&operator<<(std::ostream
&os
, const PrettyTime
&rhs
)
1931 using hours
= std::chrono::hours
;
1932 using minutes
= std::chrono::minutes
;
1934 seconds t
{rhs
.mTime
};
1941 // Only handle up to hour formatting
1943 os
<< duration_cast
<hours
>(t
).count() << 'h' << std::setfill('0') << std::setw(2)
1944 << (duration_cast
<minutes
>(t
).count() % 60) << 'm';
1946 os
<< duration_cast
<minutes
>(t
).count() << 'm' << std::setfill('0');
1947 os
<< std::setw(2) << (duration_cast
<seconds
>(t
).count() % 60) << 's' << std::setw(0)
1948 << std::setfill(' ');
1955 int main(int argc
, char *argv
[])
1957 std::unique_ptr
<MovieState
> movState
;
1961 std::cerr
<< "Usage: "<<argv
[0]<<" [-device <device name>] [-direct] <files...>" <<std::endl
;
1964 /* Register all formats and codecs */
1965 #if !(LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(58, 9, 100))
1968 /* Initialize networking protocols */
1969 avformat_network_init();
1971 if(SDL_Init(SDL_INIT_VIDEO
| SDL_INIT_EVENTS
))
1973 std::cerr
<< "Could not initialize SDL - <<"<<SDL_GetError() <<std::endl
;
1977 /* Make a window to put our video */
1978 SDL_Window
*screen
{SDL_CreateWindow(AppName
.c_str(), 0, 0, 640, 480, SDL_WINDOW_RESIZABLE
)};
1981 std::cerr
<< "SDL: could not set video mode - exiting" <<std::endl
;
1984 /* Make a renderer to handle the texture image surface and rendering. */
1985 Uint32 render_flags
{SDL_RENDERER_ACCELERATED
| SDL_RENDERER_PRESENTVSYNC
};
1986 SDL_Renderer
*renderer
{SDL_CreateRenderer(screen
, -1, render_flags
)};
1989 SDL_RendererInfo rinf
{};
1992 /* Make sure the renderer supports IYUV textures. If not, fallback to a
1993 * software renderer. */
1994 if(SDL_GetRendererInfo(renderer
, &rinf
) == 0)
1996 for(Uint32 i
{0u};!ok
&& i
< rinf
.num_texture_formats
;i
++)
1997 ok
= (rinf
.texture_formats
[i
] == SDL_PIXELFORMAT_IYUV
);
2001 std::cerr
<< "IYUV pixelformat textures not supported on renderer "<<rinf
.name
<<std::endl
;
2002 SDL_DestroyRenderer(renderer
);
2008 render_flags
= SDL_RENDERER_SOFTWARE
| SDL_RENDERER_PRESENTVSYNC
;
2009 renderer
= SDL_CreateRenderer(screen
, -1, render_flags
);
2013 std::cerr
<< "SDL: could not create renderer - exiting" <<std::endl
;
2016 SDL_SetRenderDrawColor(renderer
, 0, 0, 0, 255);
2017 SDL_RenderFillRect(renderer
, nullptr);
2018 SDL_RenderPresent(renderer
);
2020 /* Open an audio device */
2022 if(InitAL(&argv
, &argc
))
2024 std::cerr
<< "Failed to set up audio device" <<std::endl
;
2029 auto device
= alcGetContextsDevice(alcGetCurrentContext());
2030 if(alcIsExtensionPresent(device
, "ALC_SOFT_device_clock"))
2032 std::cout
<< "Found ALC_SOFT_device_clock" <<std::endl
;
2033 alcGetInteger64vSOFT
= reinterpret_cast<LPALCGETINTEGER64VSOFT
>(
2034 alcGetProcAddress(device
, "alcGetInteger64vSOFT")
2039 if(alIsExtensionPresent("AL_SOFT_source_latency"))
2041 std::cout
<< "Found AL_SOFT_source_latency" <<std::endl
;
2042 alGetSourcei64vSOFT
= reinterpret_cast<LPALGETSOURCEI64VSOFT
>(
2043 alGetProcAddress("alGetSourcei64vSOFT")
2046 if(alIsExtensionPresent("AL_SOFT_events"))
2048 std::cout
<< "Found AL_SOFT_events" <<std::endl
;
2049 alEventControlSOFT
= reinterpret_cast<LPALEVENTCONTROLSOFT
>(
2050 alGetProcAddress("alEventControlSOFT"));
2051 alEventCallbackSOFT
= reinterpret_cast<LPALEVENTCALLBACKSOFT
>(
2052 alGetProcAddress("alEventCallbackSOFT"));
2054 #ifdef AL_SOFT_callback_buffer
2055 if(alIsExtensionPresent("AL_SOFTX_callback_buffer"))
2057 std::cout
<< "Found AL_SOFT_callback_buffer" <<std::endl
;
2058 alBufferCallbackSOFT
= reinterpret_cast<LPALBUFFERCALLBACKSOFT
>(
2059 alGetProcAddress("alBufferCallbackSOFT"));
2064 for(;fileidx
< argc
;++fileidx
)
2066 if(strcmp(argv
[fileidx
], "-direct") == 0)
2068 if(alIsExtensionPresent("AL_SOFT_direct_channels_remix"))
2070 std::cout
<< "Found AL_SOFT_direct_channels_remix" <<std::endl
;
2071 DirectOutMode
= AL_REMIX_UNMATCHED_SOFT
;
2073 else if(alIsExtensionPresent("AL_SOFT_direct_channels"))
2075 std::cout
<< "Found AL_SOFT_direct_channels" <<std::endl
;
2076 DirectOutMode
= AL_DROP_UNMATCHED_SOFT
;
2079 std::cerr
<< "AL_SOFT_direct_channels not supported for direct output" <<std::endl
;
2081 else if(strcmp(argv
[fileidx
], "-wide") == 0)
2083 if(!alIsExtensionPresent("AL_EXT_STEREO_ANGLES"))
2084 std::cerr
<< "AL_EXT_STEREO_ANGLES not supported for wide stereo" <<std::endl
;
2087 std::cout
<< "Found AL_EXT_STEREO_ANGLES" <<std::endl
;
2088 EnableWideStereo
= true;
2091 else if(strcmp(argv
[fileidx
], "-uhj") == 0)
2094 if(!alIsExtensionPresent("AL_SOFTX_UHJ"))
2095 std::cerr
<< "AL_SOFT_UHJ not supported for UHJ decoding" <<std::endl
;
2098 std::cout
<< "Found AL_SOFT_UHJ" <<std::endl
;
2099 FormatStereo8
= AL_FORMAT_UHJ2CHN8_SOFT
;
2100 FormatStereo16
= AL_FORMAT_UHJ2CHN16_SOFT
;
2101 FormatStereo32F
= AL_FORMAT_UHJ2CHN_FLOAT32_SOFT
;
2104 std::cerr
<< "AL_SOFT_UHJ not supported for UHJ decoding" <<std::endl
;
2107 else if(strcmp(argv
[fileidx
], "-superstereo") == 0)
2110 if(!alIsExtensionPresent("AL_SOFTX_UHJ"))
2111 std::cerr
<< "AL_SOFT_UHJ not supported for Super Stereo decoding" <<std::endl
;
2114 std::cout
<< "Found AL_SOFT_UHJ (Super Stereo)" <<std::endl
;
2115 EnableSuperStereo
= true;
2118 std::cerr
<< "AL_SOFT_UHJ not supported for Super Stereo decoding" <<std::endl
;
2121 else if(strcmp(argv
[fileidx
], "-novideo") == 0)
2122 DisableVideo
= true;
2127 while(fileidx
< argc
&& !movState
)
2129 movState
= std::unique_ptr
<MovieState
>{new MovieState
{argv
[fileidx
++]}};
2130 if(!movState
->prepare()) movState
= nullptr;
2134 std::cerr
<< "Could not start a video" <<std::endl
;
2137 movState
->setTitle(screen
);
2139 /* Default to going to the next movie at the end of one. */
2140 enum class EomAction
{
2142 } eom_action
{EomAction::Next
};
2143 seconds last_time
{seconds::min()};
2146 /* SDL_WaitEventTimeout is broken, just force a 10ms sleep. */
2147 std::this_thread::sleep_for(milliseconds
{10});
2149 auto cur_time
= std::chrono::duration_cast
<seconds
>(movState
->getMasterClock());
2150 if(cur_time
!= last_time
)
2152 auto end_time
= std::chrono::duration_cast
<seconds
>(movState
->getDuration());
2153 std::cout
<< " \r "<<PrettyTime
{cur_time
}<<" / "<<PrettyTime
{end_time
} <<std::flush
;
2154 last_time
= cur_time
;
2157 bool force_redraw
{false};
2159 while(SDL_PollEvent(&event
) != 0)
2164 switch(event
.key
.keysym
.sym
)
2168 eom_action
= EomAction::Quit
;
2173 eom_action
= EomAction::Next
;
2181 case SDL_WINDOWEVENT
:
2182 switch(event
.window
.event
)
2184 case SDL_WINDOWEVENT_RESIZED
:
2185 SDL_SetRenderDrawColor(renderer
, 0, 0, 0, 255);
2186 SDL_RenderFillRect(renderer
, nullptr);
2187 force_redraw
= true;
2190 case SDL_WINDOWEVENT_EXPOSED
:
2191 force_redraw
= true;
2201 eom_action
= EomAction::Quit
;
2204 case FF_MOVIE_DONE_EVENT
:
2206 last_time
= seconds::min();
2207 if(eom_action
!= EomAction::Quit
)
2210 while(fileidx
< argc
&& !movState
)
2212 movState
= std::unique_ptr
<MovieState
>{new MovieState
{argv
[fileidx
++]}};
2213 if(!movState
->prepare()) movState
= nullptr;
2217 movState
->setTitle(screen
);
2222 /* Nothing more to play. Shut everything down and quit. */
2227 SDL_DestroyRenderer(renderer
);
2229 SDL_DestroyWindow(screen
);
2240 movState
->mVideo
.updateVideo(screen
, renderer
, force_redraw
);
2243 std::cerr
<< "SDL_WaitEvent error - "<<SDL_GetError() <<std::endl
;