/* * An example showing how to play a stream sync'd to video, using ffmpeg. * * Requires C++14. */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include extern "C" { #ifdef __GNUC__ _Pragma("GCC diagnostic push") _Pragma("GCC diagnostic ignored \"-Wconversion\"") _Pragma("GCC diagnostic ignored \"-Wold-style-cast\"") #endif #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libavformat/avio.h" #include "libavformat/version.h" #include "libavutil/avutil.h" #include "libavutil/error.h" #include "libavutil/frame.h" #include "libavutil/mem.h" #include "libavutil/pixfmt.h" #include "libavutil/rational.h" #include "libavutil/samplefmt.h" #include "libavutil/time.h" #include "libavutil/version.h" #include "libavutil/channel_layout.h" #include "libswscale/swscale.h" #include "libswresample/swresample.h" constexpr auto AVNoPtsValue = AV_NOPTS_VALUE; constexpr auto AVErrorEOF = AVERROR_EOF; struct SwsContext; #ifdef __GNUC__ _Pragma("GCC diagnostic pop") #endif } #include "SDL.h" #include "AL/alc.h" #include "AL/al.h" #include "AL/alext.h" #include "common/alhelpers.h" extern "C" { /* Undefine this to disable use of experimental extensions. Don't use for * production code! Interfaces and behavior may change prior to being * finalized. */ #define ALLOW_EXPERIMENTAL_EXTS #ifdef ALLOW_EXPERIMENTAL_EXTS #ifndef AL_SOFT_callback_buffer #define AL_SOFT_callback_buffer typedef unsigned int ALbitfieldSOFT; #define AL_BUFFER_CALLBACK_FUNCTION_SOFT 0x19A0 #define AL_BUFFER_CALLBACK_USER_PARAM_SOFT 0x19A1 typedef ALsizei (AL_APIENTRY*LPALBUFFERCALLBACKTYPESOFT)(ALvoid *userptr, ALvoid *sampledata, ALsizei numsamples); typedef void (AL_APIENTRY*LPALBUFFERCALLBACKSOFT)(ALuint buffer, ALenum format, ALsizei freq, LPALBUFFERCALLBACKTYPESOFT callback, ALvoid *userptr, ALbitfieldSOFT flags); typedef void (AL_APIENTRY*LPALGETBUFFERPTRSOFT)(ALuint buffer, ALenum param, ALvoid **value); typedef void (AL_APIENTRY*LPALGETBUFFER3PTRSOFT)(ALuint buffer, ALenum param, ALvoid **value1, ALvoid **value2, ALvoid **value3); typedef void (AL_APIENTRY*LPALGETBUFFERPTRVSOFT)(ALuint buffer, ALenum param, ALvoid **values); #endif #endif /* ALLOW_EXPERIMENTAL_EXTS */ } namespace { inline constexpr int64_t operator "" _i64(unsigned long long int n) noexcept { return static_cast(n); } #ifndef M_PI #define M_PI (3.14159265358979323846) #endif using fixed32 = std::chrono::duration>; 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; using std::chrono::duration_cast; const std::string AppName{"alffplay"}; ALenum DirectOutMode{AL_FALSE}; bool EnableWideStereo{false}; bool DisableVideo{false}; LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT; LPALCGETINTEGER64VSOFT alcGetInteger64vSOFT; #ifdef AL_SOFT_events LPALEVENTCONTROLSOFT alEventControlSOFT; LPALEVENTCALLBACKSOFT alEventCallbackSOFT; #endif #ifdef AL_SOFT_callback_buffer LPALBUFFERCALLBACKSOFT alBufferCallbackSOFT; #endif const seconds AVNoSyncThreshold{10}; #define VIDEO_PICTURE_QUEUE_SIZE 24 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)}; /* Per-buffer size, in time */ constexpr milliseconds AudioBufferTime{20}; /* Buffer total size, in time (should be divisible by the buffer time) */ constexpr milliseconds AudioBufferTotalTime{800}; constexpr auto AudioBufferCount = AudioBufferTotalTime / AudioBufferTime; enum { FF_MOVIE_DONE_EVENT = SDL_USEREVENT }; enum class SyncMaster { Audio, Video, External, Default = External }; inline microseconds get_avtime() { return microseconds{av_gettime()}; } /* Define unique_ptrs to auto-cleanup associated ffmpeg objects. */ struct AVIOContextDeleter { void operator()(AVIOContext *ptr) { avio_closep(&ptr); } }; using AVIOContextPtr = std::unique_ptr; struct AVFormatCtxDeleter { void operator()(AVFormatContext *ptr) { avformat_close_input(&ptr); } }; using AVFormatCtxPtr = std::unique_ptr; struct AVCodecCtxDeleter { void operator()(AVCodecContext *ptr) { avcodec_free_context(&ptr); } }; using AVCodecCtxPtr = std::unique_ptr; struct AVFrameDeleter { void operator()(AVFrame *ptr) { av_frame_free(&ptr); } }; using AVFramePtr = std::unique_ptr; struct SwrContextDeleter { void operator()(SwrContext *ptr) { swr_free(&ptr); } }; using SwrContextPtr = std::unique_ptr; struct SwsContextDeleter { void operator()(SwsContext *ptr) { sws_freeContext(ptr); } }; using SwsContextPtr = std::unique_ptr; template class PacketQueue { std::mutex mMutex; std::condition_variable mCondVar; std::deque mPackets; size_t mTotalSize{0}; bool mFinished{false}; AVPacket *getPacket(std::unique_lock &lock) { while(mPackets.empty() && !mFinished) mCondVar.wait(lock); return mPackets.empty() ? nullptr : &mPackets.front(); } void pop() { AVPacket *pkt = &mPackets.front(); mTotalSize -= static_cast(pkt->size); av_packet_unref(pkt); mPackets.pop_front(); } public: ~PacketQueue() { for(AVPacket &pkt : mPackets) av_packet_unref(&pkt); mPackets.clear(); mTotalSize = 0; } int sendTo(AVCodecContext *codecctx) { std::unique_lock lock{mMutex}; AVPacket *pkt{getPacket(lock)}; if(!pkt) return avcodec_send_packet(codecctx, nullptr); const int ret{avcodec_send_packet(codecctx, pkt)}; if(ret != AVERROR(EAGAIN)) { if(ret < 0) std::cerr<< "Failed to send packet: "< _{mMutex}; mFinished = true; } mCondVar.notify_one(); } bool put(const AVPacket *pkt) { { std::unique_lock lock{mMutex}; if(mTotalSize >= SizeLimit) return false; mPackets.push_back(AVPacket{}); if(av_packet_ref(&mPackets.back(), pkt) != 0) { mPackets.pop_back(); return true; } mTotalSize += static_cast(mPackets.back().size); } mCondVar.notify_one(); return true; } }; struct MovieState; struct AudioState { MovieState &mMovie; AVStream *mStream{nullptr}; AVCodecCtxPtr mCodecCtx; PacketQueue<2*1024*1024> mPackets; /* Used for clock difference average computation */ seconds_d64 mClockDiffAvg{0}; /* Time of the next sample to be buffered */ nanoseconds mCurrentPts{0}; /* Device clock time that the stream started at. */ nanoseconds mDeviceStartTime{nanoseconds::min()}; /* Decompressed sample frame, and swresample context for conversion */ AVFramePtr mDecodedFrame; SwrContextPtr mSwresCtx; /* Conversion format, for what gets fed to OpenAL */ uint64_t mDstChanLayout{0}; AVSampleFormat mDstSampleFmt{AV_SAMPLE_FMT_NONE}; /* Storage of converted samples */ uint8_t *mSamples{nullptr}; int mSamplesLen{0}; /* In samples */ int mSamplesPos{0}; int mSamplesMax{0}; std::unique_ptr mBufferData; size_t mBufferDataSize{0}; std::atomic mReadPos{0}; std::atomic mWritePos{0}; /* OpenAL format */ ALenum mFormat{AL_NONE}; ALuint mFrameSize{0}; std::mutex mSrcMutex; std::condition_variable mSrcCond; std::atomic_flag mConnected; ALuint mSource{0}; std::array mBuffers{}; ALuint mBufferIdx{0}; AudioState(MovieState &movie) : mMovie(movie) { mConnected.test_and_set(std::memory_order_relaxed); } ~AudioState() { if(mSource) alDeleteSources(1, &mSource); if(mBuffers[0]) alDeleteBuffers(static_cast(mBuffers.size()), mBuffers.data()); av_freep(&mSamples); } #ifdef AL_SOFT_events static void AL_APIENTRY EventCallback(ALenum eventType, ALuint object, ALuint param, ALsizei length, const ALchar *message, void *userParam); #endif #ifdef AL_SOFT_callback_buffer static ALsizei AL_APIENTRY bufferCallbackC(void *userptr, void *data, ALsizei size) { return static_cast(userptr)->bufferCallback(data, size); } ALsizei bufferCallback(void *data, ALsizei size); #endif nanoseconds getClockNoLock(); nanoseconds getClock() { std::lock_guard lock{mSrcMutex}; return getClockNoLock(); } bool startPlayback(); int getSync(); int decodeFrame(); bool readAudio(uint8_t *samples, unsigned int length, int &sample_skip); void readAudio(int sample_skip); int handler(); }; struct VideoState { MovieState &mMovie; AVStream *mStream{nullptr}; AVCodecCtxPtr mCodecCtx; PacketQueue<14*1024*1024> mPackets; /* The pts of the currently displayed frame, and the time (av_gettime) it * was last updated - used to have running video pts */ nanoseconds mDisplayPts{0}; microseconds mDisplayPtsTime{microseconds::min()}; std::mutex mDispPtsMutex; /* Swscale context for format conversion */ SwsContextPtr mSwscaleCtx; struct Picture { AVFramePtr mFrame{}; nanoseconds mPts{nanoseconds::min()}; }; std::array mPictQ; std::atomic mPictQRead{0u}, mPictQWrite{1u}; std::mutex mPictQMutex; std::condition_variable mPictQCond; SDL_Texture *mImage{nullptr}; int mWidth{0}, mHeight{0}; /* Logical image size (actual size may be larger) */ bool mFirstUpdate{true}; std::atomic mEOS{false}; std::atomic mFinalUpdate{false}; VideoState(MovieState &movie) : mMovie(movie) { } ~VideoState() { if(mImage) SDL_DestroyTexture(mImage); mImage = nullptr; } nanoseconds getClock(); void display(SDL_Window *screen, SDL_Renderer *renderer); void updateVideo(SDL_Window *screen, SDL_Renderer *renderer, bool redraw); int handler(); }; struct MovieState { AVIOContextPtr mIOContext; AVFormatCtxPtr mFormatCtx; SyncMaster mAVSyncType{SyncMaster::Default}; microseconds mClockBase{microseconds::min()}; std::atomic mQuit{false}; AudioState mAudio; VideoState mVideo; std::thread mParseThread; std::thread mAudioThread; std::thread mVideoThread; std::string mFilename; MovieState(std::string fname) : mAudio(*this), mVideo(*this), mFilename(std::move(fname)) { } ~MovieState() { mQuit = true; if(mParseThread.joinable()) mParseThread.join(); } static int decode_interrupt_cb(void *ctx); bool prepare(); void setTitle(SDL_Window *window); nanoseconds getClock(); nanoseconds getMasterClock(); nanoseconds getDuration(); int streamComponentOpen(unsigned int stream_index); int parse_handler(); }; nanoseconds AudioState::getClockNoLock() { // The audio clock is the timestamp of the sample currently being heard. if(alcGetInteger64vSOFT) { // If device start time = min, we aren't playing yet. if(mDeviceStartTime == nanoseconds::min()) return nanoseconds::zero(); // Get the current device clock time and latency. auto device = alcGetContextsDevice(alcGetCurrentContext()); 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]}; // 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; } if(mBufferDataSize > 0) { if(mDeviceStartTime == nanoseconds::min()) return nanoseconds::zero(); /* With a callback buffer and no device clock, mDeviceStartTime is * actually the timestamp of the first sample frame played. The audio * clock, then, is that plus the current source offset. */ ALint64SOFT offset[2]; if(alGetSourcei64vSOFT) alGetSourcei64vSOFT(mSource, AL_SAMPLE_OFFSET_LATENCY_SOFT, offset); else { ALint ioffset; alGetSourcei(mSource, AL_SAMPLE_OFFSET, &ioffset); offset[0] = ALint64SOFT{ioffset} << 32; offset[1] = 0; } /* NOTE: The source state must be checked last, in case an underrun * occurs and the source stops between getting the state and retrieving * the offset+latency. */ ALint status; alGetSourcei(mSource, AL_SOURCE_STATE, &status); nanoseconds pts{}; if(status == AL_PLAYING || status == AL_PAUSED) pts = mDeviceStartTime - nanoseconds{offset[1]} + duration_cast(fixed32{offset[0] / mCodecCtx->sample_rate}); else { /* If the source is stopped, the pts of the next sample to be heard * is the pts of the next sample to be buffered, minus the amount * already in the buffer ready to play. */ const size_t woffset{mWritePos.load(std::memory_order_acquire)}; const size_t roffset{mReadPos.load(std::memory_order_relaxed)}; const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) - roffset}; pts = mCurrentPts - nanoseconds{seconds{readable/mFrameSize}}/mCodecCtx->sample_rate; } return pts; } /* The source-based clock is based on 4 components: * 1 - The timestamp of the next sample to buffer (mCurrentPts) * 2 - The length of the source's buffer queue * (AudioBufferTime*AL_BUFFERS_QUEUED) * 3 - The offset OpenAL is currently at in the source (the first value * from AL_SAMPLE_OFFSET_LATENCY_SOFT) * 4 - The latency between OpenAL and the DAC (the second value from * AL_SAMPLE_OFFSET_LATENCY_SOFT) * * Subtracting the length of the source queue from the next sample's * timestamp gives the timestamp of the sample at the start of the source * queue. Adding the source offset to that results in the timestamp for the * 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}; if(mSource) { ALint64SOFT offset[2]; if(alGetSourcei64vSOFT) alGetSourcei64vSOFT(mSource, AL_SAMPLE_OFFSET_LATENCY_SOFT, offset); else { ALint ioffset; alGetSourcei(mSource, AL_SAMPLE_OFFSET, &ioffset); offset[0] = ALint64SOFT{ioffset} << 32; offset[1] = 0; } ALint queued, status; alGetSourcei(mSource, AL_BUFFERS_QUEUED, &queued); alGetSourcei(mSource, AL_SOURCE_STATE, &status); /* If the source is AL_STOPPED, then there was an underrun and all * buffers are processed, so ignore the source queue. The audio thread * will put the source into an AL_INITIAL state and clear the queue * when it starts recovery. */ if(status != AL_STOPPED) { pts -= AudioBufferTime*queued; pts += duration_cast(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]}; } return std::max(pts, nanoseconds::zero()); } bool AudioState::startPlayback() { const size_t woffset{mWritePos.load(std::memory_order_acquire)}; const size_t roffset{mReadPos.load(std::memory_order_relaxed)}; const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) - roffset}; if(mBufferDataSize > 0) { if(readable == 0) return false; if(!alcGetInteger64vSOFT) mDeviceStartTime = mCurrentPts - nanoseconds{seconds{readable/mFrameSize}}/mCodecCtx->sample_rate; } else { ALint queued{}; alGetSourcei(mSource, AL_BUFFERS_QUEUED, &queued); if(queued == 0) return false; } alSourcePlay(mSource); if(alcGetInteger64vSOFT) { /* Subtract the total buffer queue time from the current pts to get the * pts of the start of the queue. */ int64_t srctimes[2]{0,0}; alGetSourcei64vSOFT(mSource, AL_SAMPLE_OFFSET_CLOCK_SOFT, srctimes); auto device_time = nanoseconds{srctimes[1]}; auto src_offset = duration_cast(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 * the device time the source started at. Also subtract startpts to get * the device time the stream would have started at to reach where it * is now. */ if(mBufferDataSize > 0) { nanoseconds startpts{mCurrentPts - nanoseconds{seconds{readable/mFrameSize}}/mCodecCtx->sample_rate}; mDeviceStartTime = device_time - src_offset - startpts; } else { nanoseconds startpts{mCurrentPts - AudioBufferTotalTime}; mDeviceStartTime = device_time - src_offset - startpts; } } return true; } int AudioState::getSync() { if(mMovie.mAVSyncType == SyncMaster::Audio) return 0; auto ref_clock = mMovie.getMasterClock(); auto diff = ref_clock - getClockNoLock(); if(!(diff < AVNoSyncThreshold && diff > -AVNoSyncThreshold)) { /* Difference is TOO big; reset accumulated average */ mClockDiffAvg = seconds_d64::zero(); return 0; } /* Accumulate the diffs */ mClockDiffAvg = mClockDiffAvg*AudioAvgFilterCoeff + diff; auto avg_diff = mClockDiffAvg*(1.0 - AudioAvgFilterCoeff); if(avg_diff < AudioSyncThreshold/2.0 && avg_diff > -AudioSyncThreshold) return 0; /* Constrain the per-update difference to avoid exceedingly large skips */ diff = std::min(diff, AudioSampleCorrectionMax); return static_cast(duration_cast(diff*mCodecCtx->sample_rate).count()); } int AudioState::decodeFrame() { while(!mMovie.mQuit.load(std::memory_order_relaxed)) { int ret; while((ret=avcodec_receive_frame(mCodecCtx.get(), mDecodedFrame.get())) == AVERROR(EAGAIN)) mPackets.sendTo(mCodecCtx.get()); if(ret != 0) { if(ret == AVErrorEOF) break; std::cerr<< "Failed to receive frame: "<nb_samples <= 0) continue; /* If provided, update w/ pts */ if(mDecodedFrame->best_effort_timestamp != AVNoPtsValue) mCurrentPts = duration_cast(seconds_d64{av_q2d(mStream->time_base) * static_cast(mDecodedFrame->best_effort_timestamp)}); if(mDecodedFrame->nb_samples > mSamplesMax) { av_freep(&mSamples); av_samples_alloc(&mSamples, nullptr, mCodecCtx->channels, mDecodedFrame->nb_samples, mDstSampleFmt, 0); mSamplesMax = mDecodedFrame->nb_samples; } /* Return the amount of sample frames converted */ int data_size{swr_convert(mSwresCtx.get(), &mSamples, mDecodedFrame->nb_samples, const_cast(mDecodedFrame->data), mDecodedFrame->nb_samples)}; av_frame_unref(mDecodedFrame.get()); return data_size; } return 0; } /* Duplicates the sample at in to out, count times. The frame size is a * multiple of the template type size. */ template static void sample_dup(uint8_t *out, const uint8_t *in, size_t count, size_t frame_size) { auto *sample = reinterpret_cast(in); auto *dst = reinterpret_cast(out); if(frame_size == sizeof(T)) std::fill_n(dst, count, *sample); else { /* NOTE: frame_size is a multiple of sizeof(T). */ size_t type_mult{frame_size / sizeof(T)}; size_t i{0}; std::generate_n(dst, count*type_mult, [sample,type_mult,&i]() -> T { T ret = sample[i]; i = (i+1)%type_mult; return ret; } ); } } bool AudioState::readAudio(uint8_t *samples, unsigned int length, int &sample_skip) { unsigned int audio_size{0}; /* Read the next chunk of data, refill the buffer, and queue it * on the source */ length /= mFrameSize; while(mSamplesLen > 0 && audio_size < length) { unsigned int rem{length - audio_size}; if(mSamplesPos >= 0) { const auto len = static_cast(mSamplesLen - mSamplesPos); if(rem > len) rem = len; std::copy_n(mSamples + static_cast(mSamplesPos)*mFrameSize, rem*mFrameSize, samples); } else { rem = std::min(rem, static_cast(-mSamplesPos)); /* Add samples by copying the first sample */ if((mFrameSize&7) == 0) sample_dup(samples, mSamples, rem, mFrameSize); else if((mFrameSize&3) == 0) sample_dup(samples, mSamples, rem, mFrameSize); else if((mFrameSize&1) == 0) sample_dup(samples, mSamples, rem, mFrameSize); else sample_dup(samples, mSamples, rem, mFrameSize); } mSamplesPos += rem; mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate; samples += rem*mFrameSize; audio_size += rem; while(mSamplesPos >= mSamplesLen) { int frame_len = decodeFrame(); if(frame_len <= 0) break; mSamplesLen = frame_len; mSamplesPos = std::min(mSamplesLen, sample_skip); sample_skip -= mSamplesPos; // Adjust the device start time and current pts by the amount we're // skipping/duplicating, so that the clock remains correct for the // current stream position. auto skip = nanoseconds{seconds{mSamplesPos}} / mCodecCtx->sample_rate; mDeviceStartTime -= skip; mCurrentPts += skip; continue; } } if(audio_size <= 0) return false; if(audio_size < length) { const unsigned int rem{length - audio_size}; std::fill_n(samples, rem*mFrameSize, (mDstSampleFmt == AV_SAMPLE_FMT_U8) ? 0x80 : 0x00); mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate; audio_size += rem; } return true; } void AudioState::readAudio(int sample_skip) { size_t woffset{mWritePos.load(std::memory_order_acquire)}; while(mSamplesLen > 0) { const size_t roffset{mReadPos.load(std::memory_order_relaxed)}; if(mSamplesPos < 0) { size_t rem{(((roffset > woffset) ? roffset-1 : ((roffset == 0) ? mBufferDataSize-1 : mBufferDataSize)) - woffset) / mFrameSize}; rem = std::min(rem, static_cast(-mSamplesPos)); if(rem == 0) break; auto *splout{&mBufferData[woffset]}; if((mFrameSize&7) == 0) sample_dup(splout, mSamples, rem, mFrameSize); else if((mFrameSize&3) == 0) sample_dup(splout, mSamples, rem, mFrameSize); else if((mFrameSize&1) == 0) sample_dup(splout, mSamples, rem, mFrameSize); else sample_dup(splout, mSamples, rem, mFrameSize); woffset += rem * mFrameSize; if(woffset == mBufferDataSize) woffset = 0; mWritePos.store(woffset, std::memory_order_release); mSamplesPos += static_cast(rem); mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate; continue; } const size_t boffset{static_cast(mSamplesPos) * size_t{mFrameSize}}; const size_t nbytes{static_cast(mSamplesLen)*size_t{mFrameSize} - boffset}; if(roffset > woffset) { const size_t writable{roffset-woffset-1}; if(writable < nbytes) break; memcpy(&mBufferData[woffset], mSamples+boffset, nbytes); woffset += nbytes; } else { const size_t writable{mBufferDataSize+roffset-woffset-1}; if(writable < nbytes) break; const size_t todo1{std::min(nbytes, mBufferDataSize-woffset)}; const size_t todo2{nbytes - todo1}; memcpy(&mBufferData[woffset], mSamples+boffset, todo1); woffset += todo1; if(woffset == mBufferDataSize) { woffset = 0; if(todo2 > 0) { memcpy(&mBufferData[woffset], mSamples+boffset+todo1, todo2); woffset += todo2; } } } mWritePos.store(woffset, std::memory_order_release); mCurrentPts += nanoseconds{seconds{mSamplesLen-mSamplesPos}} / mCodecCtx->sample_rate; do { mSamplesLen = decodeFrame(); if(mSamplesLen <= 0) break; mSamplesPos = std::min(mSamplesLen, sample_skip); sample_skip -= mSamplesPos; auto skip = nanoseconds{seconds{mSamplesPos}} / mCodecCtx->sample_rate; mDeviceStartTime -= skip; mCurrentPts += skip; } while(mSamplesPos >= mSamplesLen); } } #ifdef AL_SOFT_events void AL_APIENTRY AudioState::EventCallback(ALenum eventType, ALuint object, ALuint param, ALsizei length, const ALchar *message, void *userParam) { auto self = static_cast(userParam); if(eventType == AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT) { /* Temporarily lock the source mutex to ensure it's not between * checking the processed count and going to sleep. */ std::unique_lock{self->mSrcMutex}.unlock(); self->mSrcCond.notify_one(); return; } std::cout<< "\n---- AL Event on AudioState "<(length)}<<"\n----"<< std::endl; if(eventType == AL_EVENT_TYPE_DISCONNECTED_SOFT) { { std::lock_guard lock{self->mSrcMutex}; self->mConnected.clear(std::memory_order_release); } self->mSrcCond.notify_one(); } } #endif #ifdef AL_SOFT_callback_buffer ALsizei AudioState::bufferCallback(void *data, ALsizei size) { ALsizei got{0}; size_t roffset{mReadPos.load(std::memory_order_acquire)}; while(got < size) { const size_t woffset{mWritePos.load(std::memory_order_relaxed)}; if(woffset == roffset) break; size_t todo{((woffset < roffset) ? mBufferDataSize : woffset) - roffset}; todo = std::min(todo, static_cast(size-got)); memcpy(data, &mBufferData[roffset], todo); data = static_cast(data) + todo; got += static_cast(todo); roffset += todo; if(roffset == mBufferDataSize) roffset = 0; } mReadPos.store(roffset, std::memory_order_release); return got; } #endif int AudioState::handler() { std::unique_lock srclock{mSrcMutex, std::defer_lock}; milliseconds sleep_time{AudioBufferTime / 3}; ALenum fmt; #ifdef AL_SOFT_events const std::array evt_types{{ AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT, AL_EVENT_TYPE_DISCONNECTED_SOFT}}; if(alEventControlSOFT) { alEventControlSOFT(evt_types.size(), evt_types.data(), AL_TRUE); alEventCallbackSOFT(EventCallback, this); sleep_time = AudioBufferTotalTime; } #endif #ifdef AL_SOFT_bformat_ex const bool has_bfmt_ex{alIsExtensionPresent("AL_SOFT_bformat_ex") != AL_FALSE}; ALenum ambi_layout{AL_FUMA_SOFT}; ALenum ambi_scale{AL_FUMA_SOFT}; #endif /* Find a suitable format for OpenAL. */ mDstChanLayout = 0; mFormat = AL_NONE; if((mCodecCtx->sample_fmt == AV_SAMPLE_FMT_FLT || mCodecCtx->sample_fmt == AV_SAMPLE_FMT_FLTP) && alIsExtensionPresent("AL_EXT_FLOAT32")) { mDstSampleFmt = AV_SAMPLE_FMT_FLT; mFrameSize = 4; if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1 && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_71CHN32")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 8; mFormat = fmt; } if((mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1 || mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_51CHN32")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 6; mFormat = fmt; } if(mCodecCtx->channel_layout == AV_CH_LAYOUT_MONO) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 1; mFormat = AL_FORMAT_MONO_FLOAT32; } /* Assume 3D B-Format (ambisonics) if the channel layout is blank and * there's 4 or more channels. FFmpeg/libavcodec otherwise seems to * have no way to specify if the source is actually B-Format (let alone * if it's 2D or 3D). */ if(mCodecCtx->channel_layout == 0 && mCodecCtx->channels >= 4 && alIsExtensionPresent("AL_EXT_BFORMAT") && (fmt=alGetEnumValue("AL_FORMAT_BFORMAT3D_FLOAT32")) != AL_NONE && fmt != -1) { int order{static_cast(std::sqrt(mCodecCtx->channels)) - 1}; if((order+1)*(order+1) == mCodecCtx->channels || (order+1)*(order+1) + 2 == mCodecCtx->channels) { /* OpenAL only supports first-order with AL_EXT_BFORMAT, which * is 4 channels for 3D buffers. */ mFrameSize *= 4; mFormat = fmt; } } if(!mFormat) { mDstChanLayout = AV_CH_LAYOUT_STEREO; mFrameSize *= 2; mFormat = AL_FORMAT_STEREO_FLOAT32; } } if(mCodecCtx->sample_fmt == AV_SAMPLE_FMT_U8 || mCodecCtx->sample_fmt == AV_SAMPLE_FMT_U8P) { mDstSampleFmt = AV_SAMPLE_FMT_U8; mFrameSize = 1; if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1 && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_71CHN8")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 8; mFormat = fmt; } if((mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1 || mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_51CHN8")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 6; mFormat = fmt; } if(mCodecCtx->channel_layout == AV_CH_LAYOUT_MONO) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 1; mFormat = AL_FORMAT_MONO8; } if(mCodecCtx->channel_layout == 0 && mCodecCtx->channels >= 4 && alIsExtensionPresent("AL_EXT_BFORMAT") && (fmt=alGetEnumValue("AL_FORMAT_BFORMAT3D8")) != AL_NONE && fmt != -1) { int order{static_cast(std::sqrt(mCodecCtx->channels)) - 1}; if((order+1)*(order+1) == mCodecCtx->channels || (order+1)*(order+1) + 2 == mCodecCtx->channels) { mFrameSize *= 4; mFormat = fmt; } } if(!mFormat) { mDstChanLayout = AV_CH_LAYOUT_STEREO; mFrameSize *= 2; mFormat = AL_FORMAT_STEREO8; } } if(!mFormat) { mDstSampleFmt = AV_SAMPLE_FMT_S16; mFrameSize = 2; if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1 && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_71CHN16")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 8; mFormat = fmt; } if((mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1 || mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_51CHN16")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 6; mFormat = fmt; } if(mCodecCtx->channel_layout == AV_CH_LAYOUT_MONO) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 1; mFormat = AL_FORMAT_MONO16; } if(mCodecCtx->channel_layout == 0 && mCodecCtx->channels >= 4 && alIsExtensionPresent("AL_EXT_BFORMAT") && (fmt=alGetEnumValue("AL_FORMAT_BFORMAT3D16")) != AL_NONE && fmt != -1) { int order{static_cast(std::sqrt(mCodecCtx->channels)) - 1}; if((order+1)*(order+1) == mCodecCtx->channels || (order+1)*(order+1) + 2 == mCodecCtx->channels) { mFrameSize *= 4; mFormat = fmt; } } if(!mFormat) { mDstChanLayout = AV_CH_LAYOUT_STEREO; mFrameSize *= 2; mFormat = AL_FORMAT_STEREO16; } } void *samples{nullptr}; ALsizei buffer_len{0}; mSamples = nullptr; mSamplesMax = 0; mSamplesPos = 0; mSamplesLen = 0; mDecodedFrame.reset(av_frame_alloc()); if(!mDecodedFrame) { std::cerr<< "Failed to allocate audio frame" <sample_rate, (1_i64<channels)-1, mCodecCtx->sample_fmt, mCodecCtx->sample_rate, 0, nullptr)); /* Note that ffmpeg/libavcodec has no method to check the ambisonic * channel order and normalization, so we can only assume AmbiX as the * defacto-standard. This is not true for .amb files, which use FuMa. */ std::vector mtx(64*64, 0.0); #ifdef AL_SOFT_bformat_ex ambi_layout = AL_ACN_SOFT; ambi_scale = AL_SN3D_SOFT; if(has_bfmt_ex) { /* An identity matrix that doesn't remix any channels. */ std::cout<< "Found AL_SOFT_bformat_ex" <(mDstChanLayout), mDstSampleFmt, mCodecCtx->sample_rate, mCodecCtx->channel_layout ? static_cast(mCodecCtx->channel_layout) : av_get_default_channel_layout(mCodecCtx->channels), mCodecCtx->sample_fmt, mCodecCtx->sample_rate, 0, nullptr)); if(!mSwresCtx || swr_init(mSwresCtx.get()) != 0) { std::cerr<< "Failed to initialize audio converter" <(mBuffers.size()), mBuffers.data()); alGenSources(1, &mSource); if(DirectOutMode) alSourcei(mSource, AL_DIRECT_CHANNELS_SOFT, DirectOutMode); if(EnableWideStereo) { const float angles[2]{static_cast(M_PI / 3.0), static_cast(-M_PI / 3.0)}; alSourcefv(mSource, AL_STEREO_ANGLES, angles); } #ifdef AL_SOFT_bformat_ex if(has_bfmt_ex) { for(ALuint bufid : mBuffers) { alBufferi(bufid, AL_AMBISONIC_LAYOUT_SOFT, ambi_layout); alBufferi(bufid, AL_AMBISONIC_SCALING_SOFT, ambi_scale); } } #endif if(alGetError() != AL_NO_ERROR) goto finish; #ifdef AL_SOFT_callback_buffer if(alBufferCallbackSOFT) { alBufferCallbackSOFT(mBuffers[0], mFormat, mCodecCtx->sample_rate, bufferCallbackC, this, 0); alSourcei(mSource, AL_BUFFER, static_cast(mBuffers[0])); if(alGetError() != AL_NO_ERROR) { fprintf(stderr, "Failed to set buffer callback\n"); alSourcei(mSource, AL_BUFFER, 0); buffer_len = static_cast(duration_cast(mCodecCtx->sample_rate * AudioBufferTime).count() * mFrameSize); } else { mBufferDataSize = static_cast(duration_cast(mCodecCtx->sample_rate * AudioBufferTotalTime).count()) * mFrameSize; mBufferData.reset(new uint8_t[mBufferDataSize]); mReadPos.store(0, std::memory_order_relaxed); mWritePos.store(0, std::memory_order_relaxed); ALCint refresh{}; alcGetIntegerv(alcGetContextsDevice(alcGetCurrentContext()), ALC_REFRESH, 1, &refresh); sleep_time = milliseconds{seconds{1}} / refresh; } } else #endif buffer_len = static_cast(duration_cast(mCodecCtx->sample_rate * AudioBufferTime).count() * mFrameSize); if(buffer_len > 0) samples = av_malloc(static_cast(buffer_len)); /* Prefill the codec buffer. */ do { const int ret{mPackets.sendTo(mCodecCtx.get())}; if(ret == AVERROR(EAGAIN) || ret == AVErrorEOF) break; } while(1); srclock.lock(); if(alcGetInteger64vSOFT) { int64_t devtime{}; alcGetInteger64vSOFT(alcGetContextsDevice(alcGetCurrentContext()), ALC_DEVICE_CLOCK_SOFT, 1, &devtime); mDeviceStartTime = nanoseconds{devtime} - mCurrentPts; } mSamplesLen = decodeFrame(); if(mSamplesLen > 0) { mSamplesPos = std::min(mSamplesLen, getSync()); auto skip = nanoseconds{seconds{mSamplesPos}} / mCodecCtx->sample_rate; mDeviceStartTime -= skip; mCurrentPts += skip; } while(!mMovie.mQuit.load(std::memory_order_relaxed) && mConnected.test_and_set(std::memory_order_relaxed)) { ALenum state; if(mBufferDataSize > 0) { alGetSourcei(mSource, AL_SOURCE_STATE, &state); readAudio(getSync()); } else { ALint processed, queued; /* First remove any processed buffers. */ alGetSourcei(mSource, AL_BUFFERS_PROCESSED, &processed); while(processed > 0) { ALuint bid; alSourceUnqueueBuffers(mSource, 1, &bid); --processed; } /* Refill the buffer queue. */ int sync_skip{getSync()}; alGetSourcei(mSource, AL_BUFFERS_QUEUED, &queued); while(static_cast(queued) < mBuffers.size()) { /* Read the next chunk of data, filling the buffer, and queue * it on the source. */ const bool got_audio{readAudio(static_cast(samples), static_cast(buffer_len), sync_skip)}; if(!got_audio) break; const ALuint bufid{mBuffers[mBufferIdx]}; mBufferIdx = static_cast((mBufferIdx+1) % mBuffers.size()); alBufferData(bufid, mFormat, samples, buffer_len, mCodecCtx->sample_rate); alSourceQueueBuffers(mSource, 1, &bufid); ++queued; } /* Check that the source is playing. */ alGetSourcei(mSource, AL_SOURCE_STATE, &state); if(state == AL_STOPPED) { /* AL_STOPPED means there was an underrun. Clear the buffer * queue since this likely means we're late, and rewind the * source to get it back into an AL_INITIAL state. */ alSourceRewind(mSource); alSourcei(mSource, AL_BUFFER, 0); if(alcGetInteger64vSOFT) { /* Also update the device start time with the current * device clock, so the decoder knows we're running behind. */ int64_t devtime{}; alcGetInteger64vSOFT(alcGetContextsDevice(alcGetCurrentContext()), ALC_DEVICE_CLOCK_SOFT, 1, &devtime); mDeviceStartTime = nanoseconds{devtime} - mCurrentPts; } continue; } } /* (re)start the source if needed, and wait for a buffer to finish */ if(state != AL_PLAYING && state != AL_PAUSED) { if(!startPlayback()) break; } if(alGetError() != AL_NO_ERROR) return false; mSrcCond.wait_for(srclock, sleep_time); } alSourceRewind(mSource); alSourcei(mSource, AL_BUFFER, 0); srclock.unlock(); finish: av_freep(&samples); #ifdef AL_SOFT_events if(alEventControlSOFT) { alEventControlSOFT(evt_types.size(), evt_types.data(), AL_FALSE); alEventCallbackSOFT(nullptr, nullptr); } #endif return 0; } nanoseconds VideoState::getClock() { /* NOTE: This returns incorrect times while not playing. */ std::lock_guard _{mDispPtsMutex}; if(mDisplayPtsTime == microseconds::min()) return nanoseconds::zero(); auto delta = get_avtime() - mDisplayPtsTime; return mDisplayPts + delta; } /* Called by VideoState::updateVideo to display the next video frame. */ void VideoState::display(SDL_Window *screen, SDL_Renderer *renderer) { if(!mImage) return; double aspect_ratio; int win_w, win_h; int w, h, x, y; if(mCodecCtx->sample_aspect_ratio.num == 0) aspect_ratio = 0.0; else { aspect_ratio = av_q2d(mCodecCtx->sample_aspect_ratio) * mCodecCtx->width / mCodecCtx->height; } if(aspect_ratio <= 0.0) aspect_ratio = static_cast(mCodecCtx->width) / mCodecCtx->height; SDL_GetWindowSize(screen, &win_w, &win_h); h = win_h; w = (static_cast(std::rint(h * aspect_ratio)) + 3) & ~3; if(w > win_w) { w = win_w; h = (static_cast(std::rint(w / aspect_ratio)) + 3) & ~3; } x = (win_w - w) / 2; y = (win_h - h) / 2; SDL_Rect src_rect{ 0, 0, mWidth, mHeight }; SDL_Rect dst_rect{ x, y, w, h }; SDL_RenderCopy(renderer, mImage, &src_rect, &dst_rect); SDL_RenderPresent(renderer); } /* Called regularly on the main thread where the SDL_Renderer was created. It * handles updating the textures of decoded frames and displaying the latest * frame. */ void VideoState::updateVideo(SDL_Window *screen, SDL_Renderer *renderer, bool redraw) { size_t read_idx{mPictQRead.load(std::memory_order_relaxed)}; Picture *vp{&mPictQ[read_idx]}; auto clocktime = mMovie.getMasterClock(); bool updated{false}; while(1) { size_t next_idx{(read_idx+1)%mPictQ.size()}; if(next_idx == mPictQWrite.load(std::memory_order_acquire)) break; Picture *nextvp{&mPictQ[next_idx]}; if(clocktime < nextvp->mPts) break; vp = nextvp; updated = true; read_idx = next_idx; } if(mMovie.mQuit.load(std::memory_order_relaxed)) { if(mEOS) mFinalUpdate = true; mPictQRead.store(read_idx, std::memory_order_release); std::unique_lock{mPictQMutex}.unlock(); mPictQCond.notify_one(); return; } if(updated) { mPictQRead.store(read_idx, std::memory_order_release); std::unique_lock{mPictQMutex}.unlock(); mPictQCond.notify_one(); /* allocate or resize the buffer! */ bool fmt_updated{false}; if(!mImage || mWidth != mCodecCtx->width || mHeight != mCodecCtx->height) { fmt_updated = true; if(mImage) SDL_DestroyTexture(mImage); mImage = SDL_CreateTexture(renderer, SDL_PIXELFORMAT_IYUV, SDL_TEXTUREACCESS_STREAMING, mCodecCtx->coded_width, mCodecCtx->coded_height); if(!mImage) std::cerr<< "Failed to create YV12 texture!" <width; mHeight = mCodecCtx->height; if(mFirstUpdate && mWidth > 0 && mHeight > 0) { /* For the first update, set the window size to the video size. */ mFirstUpdate = false; int w{mWidth}; int h{mHeight}; if(mCodecCtx->sample_aspect_ratio.den != 0) { double aspect_ratio = av_q2d(mCodecCtx->sample_aspect_ratio); if(aspect_ratio >= 1.0) w = static_cast(w*aspect_ratio + 0.5); else if(aspect_ratio > 0.0) h = static_cast(h/aspect_ratio + 0.5); } SDL_SetWindowSize(screen, w, h); } } if(mImage) { AVFrame *frame{vp->mFrame.get()}; void *pixels{nullptr}; int pitch{0}; if(mCodecCtx->pix_fmt == AV_PIX_FMT_YUV420P) SDL_UpdateYUVTexture(mImage, nullptr, frame->data[0], frame->linesize[0], frame->data[1], frame->linesize[1], frame->data[2], frame->linesize[2] ); else if(SDL_LockTexture(mImage, nullptr, &pixels, &pitch) != 0) std::cerr<< "Failed to lock texture" <coded_width}; int coded_h{mCodecCtx->coded_height}; int w{mCodecCtx->width}; int h{mCodecCtx->height}; if(!mSwscaleCtx || fmt_updated) { mSwscaleCtx.reset(sws_getContext( w, h, mCodecCtx->pix_fmt, w, h, AV_PIX_FMT_YUV420P, 0, nullptr, nullptr, nullptr )); } /* point pict at the queue */ uint8_t *pict_data[3]; pict_data[0] = static_cast(pixels); pict_data[1] = pict_data[0] + coded_w*coded_h; pict_data[2] = pict_data[1] + coded_w*coded_h/4; int pict_linesize[3]; pict_linesize[0] = pitch; pict_linesize[1] = pitch / 2; pict_linesize[2] = pitch / 2; sws_scale(mSwscaleCtx.get(), reinterpret_cast(frame->data), frame->linesize, 0, h, pict_data, pict_linesize); SDL_UnlockTexture(mImage); } } redraw = true; } if(redraw) { /* Show the picture! */ display(screen, renderer); } if(updated) { auto disp_time = get_avtime(); std::lock_guard _{mDispPtsMutex}; mDisplayPts = vp->mPts; mDisplayPtsTime = disp_time; } if(mEOS.load(std::memory_order_acquire)) { if((read_idx+1)%mPictQ.size() == mPictQWrite.load(std::memory_order_acquire)) { mFinalUpdate = true; std::unique_lock{mPictQMutex}.unlock(); mPictQCond.notify_one(); } } } int VideoState::handler() { std::for_each(mPictQ.begin(), mPictQ.end(), [](Picture &pict) -> void { pict.mFrame = AVFramePtr{av_frame_alloc()}; }); /* Prefill the codec buffer. */ do { const int ret{mPackets.sendTo(mCodecCtx.get())}; if(ret == AVERROR(EAGAIN) || ret == AVErrorEOF) break; } while(1); { std::lock_guard _{mDispPtsMutex}; mDisplayPtsTime = get_avtime(); } auto current_pts = nanoseconds::zero(); while(!mMovie.mQuit.load(std::memory_order_relaxed)) { size_t write_idx{mPictQWrite.load(std::memory_order_relaxed)}; Picture *vp{&mPictQ[write_idx]}; /* Retrieve video frame. */ AVFrame *decoded_frame{vp->mFrame.get()}; int ret; while((ret=avcodec_receive_frame(mCodecCtx.get(), decoded_frame)) == AVERROR(EAGAIN)) mPackets.sendTo(mCodecCtx.get()); if(ret != 0) { if(ret == AVErrorEOF) break; std::cerr<< "Failed to receive frame: "<best_effort_timestamp != AVNoPtsValue) current_pts = duration_cast(seconds_d64{av_q2d(mStream->time_base) * static_cast(decoded_frame->best_effort_timestamp)}); vp->mPts = current_pts; /* Update the video clock to the next expected PTS. */ auto frame_delay = av_q2d(mCodecCtx->time_base); frame_delay += decoded_frame->repeat_pict * (frame_delay * 0.5); current_pts += duration_cast(seconds_d64{frame_delay}); /* Put the frame in the queue to be loaded into a texture and displayed * by the rendering thread. */ write_idx = (write_idx+1)%mPictQ.size(); mPictQWrite.store(write_idx, std::memory_order_release); /* Send a packet now so it's hopefully ready by the time it's needed. */ mPackets.sendTo(mCodecCtx.get()); if(write_idx == mPictQRead.load(std::memory_order_acquire)) { /* Wait until we have space for a new pic */ std::unique_lock lock{mPictQMutex}; while(write_idx == mPictQRead.load(std::memory_order_acquire) && !mMovie.mQuit.load(std::memory_order_relaxed)) mPictQCond.wait(lock); } } mEOS = true; std::unique_lock lock{mPictQMutex}; while(!mFinalUpdate) mPictQCond.wait(lock); return 0; } int MovieState::decode_interrupt_cb(void *ctx) { return static_cast(ctx)->mQuit.load(std::memory_order_relaxed); } bool MovieState::prepare() { AVIOContext *avioctx{nullptr}; AVIOInterruptCB intcb{decode_interrupt_cb, this}; if(avio_open2(&avioctx, mFilename.c_str(), AVIO_FLAG_READ, &intcb, nullptr)) { std::cerr<< "Failed to open "<pb = mIOContext.get(); fmtctx->interrupt_callback = intcb; if(avformat_open_input(&fmtctx, mFilename.c_str(), nullptr, nullptr) != 0) { std::cerr<< "Failed to open "<>(mFormatCtx->duration); } int MovieState::streamComponentOpen(unsigned int stream_index) { if(stream_index >= mFormatCtx->nb_streams) return -1; /* Get a pointer to the codec context for the stream, and open the * associated codec. */ AVCodecCtxPtr avctx{avcodec_alloc_context3(nullptr)}; if(!avctx) return -1; if(avcodec_parameters_to_context(avctx.get(), mFormatCtx->streams[stream_index]->codecpar)) return -1; AVCodec *codec{avcodec_find_decoder(avctx->codec_id)}; if(!codec || avcodec_open2(avctx.get(), codec, nullptr) < 0) { std::cerr<< "Unsupported codec: "<codec_id) << " (0x"<codec_id<codec_type) { case AVMEDIA_TYPE_AUDIO: mAudio.mStream = mFormatCtx->streams[stream_index]; mAudio.mCodecCtx = std::move(avctx); break; case AVMEDIA_TYPE_VIDEO: mVideo.mStream = mFormatCtx->streams[stream_index]; mVideo.mCodecCtx = std::move(avctx); break; default: return -1; } return static_cast(stream_index); } int MovieState::parse_handler() { auto &audio_queue = mAudio.mPackets; auto &video_queue = mVideo.mPackets; int video_index{-1}; int audio_index{-1}; /* Find the first video and audio streams */ for(unsigned int i{0u};i < mFormatCtx->nb_streams;i++) { auto codecpar = mFormatCtx->streams[i]->codecpar; if(codecpar->codec_type == AVMEDIA_TYPE_VIDEO && !DisableVideo && video_index < 0) video_index = streamComponentOpen(i); else if(codecpar->codec_type == AVMEDIA_TYPE_AUDIO && audio_index < 0) audio_index = streamComponentOpen(i); } if(video_index < 0 && audio_index < 0) { std::cerr<< mFilename<<": could not open codecs" <= 0) mAudioThread = std::thread{std::mem_fn(&AudioState::handler), &mAudio}; if(video_index >= 0) mVideoThread = std::thread{std::mem_fn(&VideoState::handler), &mVideo}; /* Main packet reading/dispatching loop */ while(!mQuit.load(std::memory_order_relaxed)) { AVPacket packet; if(av_read_frame(mFormatCtx.get(), &packet) < 0) break; /* Copy the packet into the queue it's meant for. */ if(packet.stream_index == video_index) { while(!mQuit.load(std::memory_order_acquire) && !video_queue.put(&packet)) std::this_thread::sleep_for(milliseconds{100}); } else if(packet.stream_index == audio_index) { while(!mQuit.load(std::memory_order_acquire) && !audio_queue.put(&packet)) std::this_thread::sleep_for(milliseconds{100}); } av_packet_unref(&packet); } /* Finish the queues so the receivers know nothing more is coming. */ if(mVideo.mCodecCtx) video_queue.setFinished(); if(mAudio.mCodecCtx) audio_queue.setFinished(); /* all done - wait for it */ if(mVideoThread.joinable()) mVideoThread.join(); if(mAudioThread.joinable()) mAudioThread.join(); mVideo.mEOS = true; std::unique_lock lock{mVideo.mPictQMutex}; while(!mVideo.mFinalUpdate) mVideo.mPictQCond.wait(lock); lock.unlock(); SDL_Event evt{}; evt.user.type = FF_MOVIE_DONE_EVENT; SDL_PushEvent(&evt); return 0; } // Helper class+method to print the time with human-readable formatting. struct PrettyTime { seconds mTime; }; std::ostream &operator<<(std::ostream &os, const PrettyTime &rhs) { using hours = std::chrono::hours; using minutes = std::chrono::minutes; seconds t{rhs.mTime}; if(t.count() < 0) { os << '-'; t *= -1; } // Only handle up to hour formatting if(t >= hours{1}) os << duration_cast(t).count() << 'h' << std::setfill('0') << std::setw(2) << (duration_cast(t).count() % 60) << 'm'; else os << duration_cast(t).count() << 'm' << std::setfill('0'); os << std::setw(2) << (duration_cast(t).count() % 60) << 's' << std::setw(0) << std::setfill(' '); return os; } } // namespace int main(int argc, char *argv[]) { std::unique_ptr movState; if(argc < 2) { std::cerr<< "Usage: "<] [-direct] " <= AV_VERSION_INT(58, 9, 100)) av_register_all(); #endif /* Initialize networking protocols */ avformat_network_init(); if(SDL_Init(SDL_INIT_VIDEO | SDL_INIT_EVENTS)) { std::cerr<< "Could not initialize SDL - <<"<( alcGetProcAddress(device, "alcGetInteger64vSOFT") ); } } if(alIsExtensionPresent("AL_SOFT_source_latency")) { std::cout<< "Found AL_SOFT_source_latency" <( alGetProcAddress("alGetSourcei64vSOFT") ); } #ifdef AL_SOFT_events if(alIsExtensionPresent("AL_SOFT_events")) { std::cout<< "Found AL_SOFT_events" <( alGetProcAddress("alEventControlSOFT")); alEventCallbackSOFT = reinterpret_cast( alGetProcAddress("alEventCallbackSOFT")); } #endif #ifdef AL_SOFT_callback_buffer if(alIsExtensionPresent("AL_SOFTX_callback_buffer")) { std::cout<< "Found AL_SOFT_callback_buffer" <( alGetProcAddress("alBufferCallbackSOFT")); } #endif int fileidx{0}; for(;fileidx < argc;++fileidx) { if(strcmp(argv[fileidx], "-direct") == 0) { if(alIsExtensionPresent("AL_SOFT_direct_channels_remix")) { std::cout<< "Found AL_SOFT_direct_channels_remix" <{new MovieState{argv[fileidx++]}}; if(!movState->prepare()) movState = nullptr; } if(!movState) { std::cerr<< "Could not start a video" <setTitle(screen); /* Default to going to the next movie at the end of one. */ enum class EomAction { Next, Quit } eom_action{EomAction::Next}; seconds last_time{seconds::min()}; while(1) { SDL_Event event{}; int have_evt{SDL_WaitEventTimeout(&event, 10)}; auto cur_time = std::chrono::duration_cast(movState->getMasterClock()); if(cur_time != last_time) { auto end_time = std::chrono::duration_cast(movState->getDuration()); std::cout<< " \r "<mQuit = true; eom_action = EomAction::Quit; break; case SDLK_n: movState->mQuit = true; eom_action = EomAction::Next; break; default: break; } break; case SDL_WINDOWEVENT: switch(event.window.event) { case SDL_WINDOWEVENT_RESIZED: SDL_SetRenderDrawColor(renderer, 0, 0, 0, 255); SDL_RenderFillRect(renderer, nullptr); force_redraw = true; break; case SDL_WINDOWEVENT_EXPOSED: force_redraw = true; break; default: break; } break; case SDL_QUIT: movState->mQuit = true; eom_action = EomAction::Quit; break; case FF_MOVIE_DONE_EVENT: std::cout<<'\n'; last_time = seconds::min(); if(eom_action != EomAction::Quit) { movState = nullptr; while(fileidx < argc && !movState) { movState = std::unique_ptr{new MovieState{argv[fileidx++]}}; if(!movState->prepare()) movState = nullptr; } if(movState) { movState->setTitle(screen); break; } } /* Nothing more to play. Shut everything down and quit. */ movState = nullptr; CloseAL(); SDL_DestroyRenderer(renderer); renderer = nullptr; SDL_DestroyWindow(screen); screen = nullptr; SDL_Quit(); exit(0); default: break; } } while(SDL_PollEvent(&event)); movState->mVideo.updateVideo(screen, renderer, force_redraw); } std::cerr<< "SDL_WaitEvent error - "<