antkeeper
/
superbuild
1
0
Fork 0
Code Issues 0 Pull Requests 0 Releases 0 Activity
🛠️🐜 Antkeeper superbuild with dependencies included https://antkeeper.com
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
69 Commits
1 Branch
60 MiB
 
 
 
 
 
 
Tree: 7d35d66e7d
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from '7d35d66e7d'
${ noResults }
superbuild/modules/openal-soft/examples/alffplay.cpp

2155 lines
69 KiB
Raw Blame History

/*
* An example showing how to play a stream sync'd to video, using ffmpeg.
*
* Requires C++14.
*/
#include <condition_variable>
#include <functional>
#include <algorithm>
#include <iostream>
#include <utility>
#include <iomanip>
#include <cstdint>
#include <cstring>
#include <cstdlib>
#include <atomic>
#include <cerrno>
#include <chrono>
#include <cstdio>
#include <future>
#include <memory>
#include <string>
#include <thread>
#include <vector>
#include <array>
#include <cmath>
#include <deque>
#include <mutex>
#include <ratio>
#ifdef __GNUC__
_Pragma("GCC diagnostic push")
_Pragma("GCC diagnostic ignored \"-Wconversion\"")
_Pragma("GCC diagnostic ignored \"-Wold-style-cast\"")
#endif
extern "C" {
#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;
}
#include "SDL.h"
#ifdef __GNUC__
_Pragma("GCC diagnostic pop")
#endif
#include "AL/alc.h"
#include "AL/al.h"
#include "AL/alext.h"
#include "common/alhelpers.h"
namespace {
inline constexpr int64_t operator "" _i64(unsigned long long int n) noexcept { return static_cast<int64_t>(n); }
#ifndef M_PI
#define M_PI (3.14159265358979323846)
#endif
using fixed32 = std::chrono::duration<int64_t,std::ratio<1,(1_i64<<32)>>;
using nanoseconds = std::chrono::nanoseconds;
using microseconds = std::chrono::microseconds;
using milliseconds = std::chrono::milliseconds;
using seconds = std::chrono::seconds;
using seconds_d64 = std::chrono::duration<double>;
using std::chrono::duration_cast;
const std::string AppName{"alffplay"};
ALenum DirectOutMode{AL_FALSE};
bool EnableWideStereo{false};
bool EnableUhj{false};
bool EnableSuperStereo{false};
bool DisableVideo{false};
LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT;
LPALCGETINTEGER64VSOFT alcGetInteger64vSOFT;
LPALEVENTCONTROLSOFT alEventControlSOFT;
LPALEVENTCALLBACKSOFT alEventCallbackSOFT;
LPALBUFFERCALLBACKSOFT alBufferCallbackSOFT;
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 = Audio
};
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<AVIOContext,AVIOContextDeleter>;
struct AVFormatCtxDeleter {
void operator()(AVFormatContext *ptr) { avformat_close_input(&ptr); }
};
using AVFormatCtxPtr = std::unique_ptr<AVFormatContext,AVFormatCtxDeleter>;
struct AVCodecCtxDeleter {
void operator()(AVCodecContext *ptr) { avcodec_free_context(&ptr); }
};
using AVCodecCtxPtr = std::unique_ptr<AVCodecContext,AVCodecCtxDeleter>;
struct AVPacketDeleter {
void operator()(AVPacket *pkt) { av_packet_free(&pkt); }
};
using AVPacketPtr = std::unique_ptr<AVPacket,AVPacketDeleter>;
struct AVFrameDeleter {
void operator()(AVFrame *ptr) { av_frame_free(&ptr); }
};
using AVFramePtr = std::unique_ptr<AVFrame,AVFrameDeleter>;
struct SwrContextDeleter {
void operator()(SwrContext *ptr) { swr_free(&ptr); }
};
using SwrContextPtr = std::unique_ptr<SwrContext,SwrContextDeleter>;
struct SwsContextDeleter {
void operator()(SwsContext *ptr) { sws_freeContext(ptr); }
};
using SwsContextPtr = std::unique_ptr<SwsContext,SwsContextDeleter>;
template<size_t SizeLimit>
class DataQueue {
std::mutex mPacketMutex, mFrameMutex;
std::condition_variable mPacketCond;
std::condition_variable mInFrameCond, mOutFrameCond;
std::deque<AVPacketPtr> mPackets;
size_t mTotalSize{0};
bool mFinished{false};
AVPacketPtr getPacket()
{
std::unique_lock<std::mutex> plock{mPacketMutex};
while(mPackets.empty() && !mFinished)
mPacketCond.wait(plock);
if(mPackets.empty())
return nullptr;
auto ret = std::move(mPackets.front());
mPackets.pop_front();
mTotalSize -= static_cast<unsigned int>(ret->size);
return ret;
}
public:
int sendPacket(AVCodecContext *codecctx)
{
AVPacketPtr packet{getPacket()};
int ret{};
{
std::unique_lock<std::mutex> flock{mFrameMutex};
while((ret=avcodec_send_packet(codecctx, packet.get())) == AVERROR(EAGAIN))
mInFrameCond.wait_for(flock, milliseconds{50});
}
mOutFrameCond.notify_one();
if(!packet)
{
if(!ret) return AVErrorEOF;
std::cerr<< "Failed to send flush packet: "<<ret <<std::endl;
return ret;
}
if(ret < 0)
std::cerr<< "Failed to send packet: "<<ret <<std::endl;
return ret;
}
int receiveFrame(AVCodecContext *codecctx, AVFrame *frame)
{
int ret{};
{
std::unique_lock<std::mutex> flock{mFrameMutex};
while((ret=avcodec_receive_frame(codecctx, frame)) == AVERROR(EAGAIN))
mOutFrameCond.wait_for(flock, milliseconds{50});
}
mInFrameCond.notify_one();
return ret;
}
void setFinished()
{
{
std::lock_guard<std::mutex> _{mPacketMutex};
mFinished = true;
}
mPacketCond.notify_one();
}
void flush()
{
{
std::lock_guard<std::mutex> _{mPacketMutex};
mFinished = true;
mPackets.clear();
mTotalSize = 0;
}
mPacketCond.notify_one();
}
bool put(const AVPacket *pkt)
{
{
std::unique_lock<std::mutex> lock{mPacketMutex};
if(mTotalSize >= SizeLimit || mFinished)
return false;
mPackets.push_back(AVPacketPtr{av_packet_alloc()});
if(av_packet_ref(mPackets.back().get(), pkt) != 0)
{
mPackets.pop_back();
return true;
}
mTotalSize += static_cast<unsigned int>(mPackets.back()->size);
}
mPacketCond.notify_one();
return true;
}
};
struct MovieState;
struct AudioState {
MovieState &mMovie;
AVStream *mStream{nullptr};
AVCodecCtxPtr mCodecCtx;
DataQueue<2*1024*1024> mQueue;
/* 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<uint8_t[]> mBufferData;
size_t mBufferDataSize{0};
std::atomic<size_t> mReadPos{0};
std::atomic<size_t> 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<ALuint,AudioBufferCount> 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<ALsizei>(mBuffers.size()), mBuffers.data());
av_freep(&mSamples);
}
static void AL_APIENTRY eventCallbackC(ALenum eventType, ALuint object, ALuint param,
ALsizei length, const ALchar *message, void *userParam)
{ static_cast<AudioState*>(userParam)->eventCallback(eventType, object, param, length, message); }
void eventCallback(ALenum eventType, ALuint object, ALuint param, ALsizei length,
const ALchar *message);
static ALsizei AL_APIENTRY bufferCallbackC(void *userptr, void *data, ALsizei size)
{ return static_cast<AudioState*>(userptr)->bufferCallback(data, size); }
ALsizei bufferCallback(void *data, ALsizei size);
nanoseconds getClockNoLock();
nanoseconds getClock()
{
std::lock_guard<std::mutex> lock{mSrcMutex};
return getClockNoLock();
}
bool startPlayback();
int getSync();
int decodeFrame();
bool readAudio(uint8_t *samples, unsigned int length, int &sample_skip);
bool readAudio(int sample_skip);
int handler();
};
struct VideoState {
MovieState &mMovie;
AVStream *mStream{nullptr};
AVCodecCtxPtr mCodecCtx;
DataQueue<14*1024*1024> mQueue;
/* 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<Picture,VIDEO_PICTURE_QUEUE_SIZE> mPictQ;
std::atomic<size_t> mPictQRead{0u}, mPictQWrite{1u};
std::mutex mPictQMutex;
std::condition_variable mPictQCond;
SDL_Texture *mImage{nullptr};
int mWidth{0}, mHeight{0}; /* Full texture size */
bool mFirstUpdate{true};
std::atomic<bool> mEOS{false};
std::atomic<bool> 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, AVFrame *frame);
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<bool> mQuit{false};
AudioState mAudio;
VideoState mVideo;
std::mutex mStartupMutex;
std::condition_variable mStartupCond;
bool mStartupDone{false};
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()
{
stop();
if(mParseThread.joinable())
mParseThread.join();
}
static int decode_interrupt_cb(void *ctx);
bool prepare();
void setTitle(SDL_Window *window);
void stop();
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<nanoseconds>(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<nanoseconds>(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<nanoseconds>(fixed32{srctimes[0]}) /
mCodecCtx->sample_rate;
/* The mixer may have ticked and incremented the device time and sample
* offset, so subtract the source offset from the device time to get
* 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<nanoseconds>(diff, AudioSampleCorrectionMax);
return static_cast<int>(duration_cast<seconds>(diff*mCodecCtx->sample_rate).count());
}
int AudioState::decodeFrame()
{
do {
while(int ret{mQueue.receiveFrame(mCodecCtx.get(), mDecodedFrame.get())})
{
if(ret == AVErrorEOF) return 0;
std::cerr<< "Failed to receive frame: "<<ret <<std::endl;
}
} while(mDecodedFrame->nb_samples <= 0);
/* If provided, update w/ pts */
if(mDecodedFrame->best_effort_timestamp != AVNoPtsValue)
mCurrentPts = duration_cast<nanoseconds>(seconds_d64{av_q2d(mStream->time_base) *
static_cast<double>(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<const uint8_t**>(mDecodedFrame->data), mDecodedFrame->nb_samples)};
av_frame_unref(mDecodedFrame.get());
return data_size;
}
/* Duplicates the sample at in to out, count times. The frame size is a
* multiple of the template type size.
*/
template<typename T>
static void sample_dup(uint8_t *out, const uint8_t *in, size_t count, size_t frame_size)
{
auto *sample = reinterpret_cast<const T*>(in);
auto *dst = reinterpret_cast<T*>(out);
/* NOTE: frame_size is a multiple of sizeof(T). */
size_t type_mult{frame_size / sizeof(T)};
if(type_mult == 1)
std::fill_n(dst, count, *sample);
else for(size_t i{0};i < count;++i)
{
for(size_t j{0};j < type_mult;++j)
dst[i*type_mult + j] = sample[j];
}
}
static void sample_dup(uint8_t *out, const uint8_t *in, size_t count, size_t frame_size)
{
if((frame_size&7) == 0)
sample_dup<uint64_t>(out, in, count, frame_size);
else if((frame_size&3) == 0)
sample_dup<uint32_t>(out, in, count, frame_size);
else if((frame_size&1) == 0)
sample_dup<uint16_t>(out, in, count, frame_size);
else
sample_dup<uint8_t>(out, in, count, frame_size);
}
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<unsigned int>(mSamplesLen - mSamplesPos);
if(rem > len) rem = len;
std::copy_n(mSamples + static_cast<unsigned int>(mSamplesPos)*mFrameSize,
rem*mFrameSize, samples);
}
else
{
rem = std::min(rem, static_cast<unsigned int>(-mSamplesPos));
/* Add samples by copying the first sample */
sample_dup(samples, mSamples, rem, mFrameSize);
}
mSamplesPos += rem;
mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate;
samples += rem*mFrameSize;
audio_size += rem;
while(mSamplesPos >= mSamplesLen)
{
mSamplesLen = decodeFrame();
mSamplesPos = std::min(mSamplesLen, sample_skip);
if(mSamplesLen <= 0) break;
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;
}
}
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;
}
return true;
}
bool AudioState::readAudio(int sample_skip)
{
size_t woffset{mWritePos.load(std::memory_order_acquire)};
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
while(mSamplesLen > 0)
{
const size_t nsamples{((roffset > woffset) ? roffset-woffset-1
: (roffset == 0) ? (mBufferDataSize-woffset-1)
: (mBufferDataSize-woffset)) / mFrameSize};
if(!nsamples) break;
if(mSamplesPos < 0)
{
const size_t rem{std::min<size_t>(nsamples, static_cast<ALuint>(-mSamplesPos))};
sample_dup(&mBufferData[woffset], mSamples, rem, mFrameSize);
woffset += rem * mFrameSize;
if(woffset == mBufferDataSize) woffset = 0;
mWritePos.store(woffset, std::memory_order_release);
mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate;
mSamplesPos += static_cast<int>(rem);
continue;
}
const size_t rem{std::min<size_t>(nsamples, static_cast<ALuint>(mSamplesLen-mSamplesPos))};
const size_t boffset{static_cast<ALuint>(mSamplesPos) * size_t{mFrameSize}};
const size_t nbytes{rem * mFrameSize};
memcpy(&mBufferData[woffset], mSamples + boffset, nbytes);
woffset += nbytes;
if(woffset == mBufferDataSize) woffset = 0;
mWritePos.store(woffset, std::memory_order_release);
mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate;
mSamplesPos += static_cast<int>(rem);
while(mSamplesPos >= mSamplesLen)
{
mSamplesLen = decodeFrame();
mSamplesPos = std::min(mSamplesLen, sample_skip);
if(mSamplesLen <= 0) return false;
sample_skip -= mSamplesPos;
auto skip = nanoseconds{seconds{mSamplesPos}} / mCodecCtx->sample_rate;
mDeviceStartTime -= skip;
mCurrentPts += skip;
}
}
return true;
}
void AL_APIENTRY AudioState::eventCallback(ALenum eventType, ALuint object, ALuint param,
ALsizei length, const ALchar *message)
{
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<std::mutex>{mSrcMutex}.unlock();
mSrcCond.notify_one();
return;
}
std::cout<< "\n---- AL Event on AudioState "<<this<<" ----\nEvent: ";
switch(eventType)
{
case AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT: std::cout<< "Buffer completed"; break;
case AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT: std::cout<< "Source state changed"; break;
case AL_EVENT_TYPE_DISCONNECTED_SOFT: std::cout<< "Disconnected"; break;
default:
std::cout<< "0x"<<std::hex<<std::setw(4)<<std::setfill('0')<<eventType<<std::dec<<
std::setw(0)<<std::setfill(' '); break;
}
std::cout<< "\n"
"Object ID: "<<object<<"\n"
"Parameter: "<<param<<"\n"
"Message: "<<std::string{message, static_cast<ALuint>(length)}<<"\n----"<<
std::endl;
if(eventType == AL_EVENT_TYPE_DISCONNECTED_SOFT)
{
{
std::lock_guard<std::mutex> lock{mSrcMutex};
mConnected.clear(std::memory_order_release);
}
mSrcCond.notify_one();
}
}
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<size_t>(todo, static_cast<ALuint>(size-got));
memcpy(data, &mBufferData[roffset], todo);
data = static_cast<ALbyte*>(data) + todo;
got += static_cast<ALsizei>(todo);
roffset += todo;
if(roffset == mBufferDataSize)
roffset = 0;
}
mReadPos.store(roffset, std::memory_order_release);
return got;
}
int AudioState::handler()
{
std::unique_lock<std::mutex> srclock{mSrcMutex, std::defer_lock};
milliseconds sleep_time{AudioBufferTime / 3};
struct EventControlManager {
const std::array<ALenum,3> evt_types{{
AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT,
AL_EVENT_TYPE_DISCONNECTED_SOFT}};
EventControlManager(milliseconds &sleep_time)
{
if(alEventControlSOFT)
{
alEventControlSOFT(static_cast<ALsizei>(evt_types.size()), evt_types.data(),
AL_TRUE);
alEventCallbackSOFT(&AudioState::eventCallbackC, this);
sleep_time = AudioBufferTotalTime;
}
}
~EventControlManager()
{
if(alEventControlSOFT)
{
alEventControlSOFT(static_cast<ALsizei>(evt_types.size()), evt_types.data(),
AL_FALSE);
alEventCallbackSOFT(nullptr, nullptr);
}
}
};
EventControlManager event_controller{sleep_time};
const bool has_bfmt_ex{alIsExtensionPresent("AL_SOFT_bformat_ex") != AL_FALSE};
ALenum ambi_layout{AL_FUMA_SOFT};
ALenum ambi_scale{AL_FUMA_SOFT};
std::unique_ptr<uint8_t[]> samples;
ALsizei buffer_len{0};
/* 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
|| mCodecCtx->sample_fmt == AV_SAMPLE_FMT_DBL
|| mCodecCtx->sample_fmt == AV_SAMPLE_FMT_DBLP
|| mCodecCtx->sample_fmt == AV_SAMPLE_FMT_S32
|| mCodecCtx->sample_fmt == AV_SAMPLE_FMT_S32P
|| mCodecCtx->sample_fmt == AV_SAMPLE_FMT_S64
|| mCodecCtx->sample_fmt == AV_SAMPLE_FMT_S64P)
&& alIsExtensionPresent("AL_EXT_FLOAT32"))
{
mDstSampleFmt = AV_SAMPLE_FMT_FLT;
mFrameSize = 4;
if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
{
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 8;
mFormat = alGetEnumValue("AL_FORMAT_71CHN32");
}
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1
|| mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 6;
mFormat = alGetEnumValue("AL_FORMAT_51CHN32");
}
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_QUAD)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 4;
mFormat = alGetEnumValue("AL_FORMAT_QUAD32");
}
}
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"))
{
/* Calculate what should be the ambisonic order from the number of
* channels, and confirm that's the number of channels. Opus allows
* an optional non-diegetic stereo stream with the B-Format stream,
* which we can ignore, so check for that too.
*/
auto order = static_cast<int>(std::sqrt(mCodecCtx->channels)) - 1;
int channels{(order+1) * (order+1)};
if(channels == mCodecCtx->channels || channels+2 == mCodecCtx->channels)
{
/* OpenAL only supports first-order with AL_EXT_BFORMAT, which
* is 4 channels for 3D buffers.
*/
mFrameSize *= 4;
mFormat = alGetEnumValue("AL_FORMAT_BFORMAT3D_FLOAT32");
}
}
if(!mFormat || mFormat == -1)
{
mDstChanLayout = AV_CH_LAYOUT_STEREO;
mFrameSize *= 2;
mFormat = EnableUhj ? AL_FORMAT_UHJ2CHN_FLOAT32_SOFT : 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(alIsExtensionPresent("AL_EXT_MCFORMATS"))
{
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 8;
mFormat = alGetEnumValue("AL_FORMAT_71CHN8");
}
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1
|| mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 6;
mFormat = alGetEnumValue("AL_FORMAT_51CHN8");
}
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_QUAD)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 4;
mFormat = alGetEnumValue("AL_FORMAT_QUAD8");
}
}
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"))
{
auto order = static_cast<int>(std::sqrt(mCodecCtx->channels)) - 1;
int channels{(order+1) * (order+1)};
if(channels == mCodecCtx->channels || channels+2 == mCodecCtx->channels)
{
mFrameSize *= 4;
mFormat = alGetEnumValue("AL_FORMAT_BFORMAT3D_8");
}
}
if(!mFormat || mFormat == -1)
{
mDstChanLayout = AV_CH_LAYOUT_STEREO;
mFrameSize *= 2;
mFormat = EnableUhj ? AL_FORMAT_UHJ2CHN8_SOFT : AL_FORMAT_STEREO8;
}
}
if(!mFormat || mFormat == -1)
{
mDstSampleFmt = AV_SAMPLE_FMT_S16;
mFrameSize = 2;
if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
{
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 8;
mFormat = alGetEnumValue("AL_FORMAT_71CHN16");
}
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1
|| mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 6;
mFormat = alGetEnumValue("AL_FORMAT_51CHN16");
}
if(mCodecCtx->channel_layout == AV_CH_LAYOUT_QUAD)
{
mDstChanLayout = mCodecCtx->channel_layout;
mFrameSize *= 4;
mFormat = alGetEnumValue("AL_FORMAT_QUAD16");
}
}
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"))
{
auto order = static_cast<int>(std::sqrt(mCodecCtx->channels)) - 1;
int channels{(order+1) * (order+1)};
if(channels == mCodecCtx->channels || channels+2 == mCodecCtx->channels)
{
mFrameSize *= 4;
mFormat = alGetEnumValue("AL_FORMAT_BFORMAT3D_16");
}
}
if(!mFormat || mFormat == -1)
{
mDstChanLayout = AV_CH_LAYOUT_STEREO;
mFrameSize *= 2;
mFormat = EnableUhj ? AL_FORMAT_UHJ2CHN16_SOFT : AL_FORMAT_STEREO16;
}
}
mSamples = nullptr;
mSamplesMax = 0;
mSamplesPos = 0;
mSamplesLen = 0;
mDecodedFrame.reset(av_frame_alloc());
if(!mDecodedFrame)
{
std::cerr<< "Failed to allocate audio frame" <<std::endl;
return 0;
}
if(!mDstChanLayout)
{
/* OpenAL only supports first-order ambisonics with AL_EXT_BFORMAT, so
* we have to drop any extra channels.
*/
mSwresCtx.reset(swr_alloc_set_opts(nullptr,
(1_i64<<4)-1, mDstSampleFmt, mCodecCtx->sample_rate,
(1_i64<<mCodecCtx->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<double> mtx(64*64, 0.0);
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" <<std::endl;
mtx[0 + 0*64] = 1.0;
mtx[1 + 1*64] = 1.0;
mtx[2 + 2*64] = 1.0;
mtx[3 + 3*64] = 1.0;
}
else
{
std::cout<< "Found AL_EXT_BFORMAT" <<std::endl;
/* Without AL_SOFT_bformat_ex, OpenAL only supports FuMa channel
* ordering and normalization, so a custom matrix is needed to
* scale and reorder the source from AmbiX.
*/
mtx[0 + 0*64] = std::sqrt(0.5);
mtx[3 + 1*64] = 1.0;
mtx[1 + 2*64] = 1.0;
mtx[2 + 3*64] = 1.0;
}
swr_set_matrix(mSwresCtx.get(), mtx.data(), 64);
}
else
mSwresCtx.reset(swr_alloc_set_opts(nullptr,
static_cast<int64_t>(mDstChanLayout), mDstSampleFmt, mCodecCtx->sample_rate,
mCodecCtx->channel_layout ? static_cast<int64_t>(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" <<std::endl;
return 0;
}
alGenBuffers(static_cast<ALsizei>(mBuffers.size()), mBuffers.data());
alGenSources(1, &mSource);
if(DirectOutMode)
alSourcei(mSource, AL_DIRECT_CHANNELS_SOFT, DirectOutMode);
if(EnableWideStereo)
{
const float angles[2]{static_cast<float>(M_PI / 3.0), static_cast<float>(-M_PI / 3.0)};
alSourcefv(mSource, AL_STEREO_ANGLES, angles);
}
if(has_bfmt_ex)
{
for(ALuint bufid : mBuffers)
{
alBufferi(bufid, AL_AMBISONIC_LAYOUT_SOFT, ambi_layout);
alBufferi(bufid, AL_AMBISONIC_SCALING_SOFT, ambi_scale);
}
}
#ifdef AL_SOFT_UHJ
if(EnableSuperStereo)
alSourcei(mSource, AL_STEREO_MODE_SOFT, AL_SUPER_STEREO_SOFT);
#endif
if(alGetError() != AL_NO_ERROR)
return 0;
bool callback_ok{false};
if(alBufferCallbackSOFT)
{
alBufferCallbackSOFT(mBuffers[0], mFormat, mCodecCtx->sample_rate, bufferCallbackC, this);
alSourcei(mSource, AL_BUFFER, static_cast<ALint>(mBuffers[0]));
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Failed to set buffer callback\n");
alSourcei(mSource, AL_BUFFER, 0);
}
else
{
mBufferDataSize = static_cast<size_t>(duration_cast<seconds>(mCodecCtx->sample_rate *
AudioBufferTotalTime).count()) * mFrameSize;
mBufferData = std::make_unique<uint8_t[]>(mBufferDataSize);
std::fill_n(mBufferData.get(), mBufferDataSize, uint8_t{});
mReadPos.store(0, std::memory_order_relaxed);
mWritePos.store(mBufferDataSize/mFrameSize/2*mFrameSize, std::memory_order_relaxed);
ALCint refresh{};
alcGetIntegerv(alcGetContextsDevice(alcGetCurrentContext()), ALC_REFRESH, 1, &refresh);
sleep_time = milliseconds{seconds{1}} / refresh;
callback_ok = true;
}
}
if(!callback_ok)
buffer_len = static_cast<int>(duration_cast<seconds>(mCodecCtx->sample_rate *
AudioBufferTime).count() * mFrameSize);
if(buffer_len > 0)
samples = std::make_unique<uint8_t[]>(static_cast<ALuint>(buffer_len));
/* Prefill the codec buffer. */
auto packet_sender = [this]()
{
while(1)
{
const int ret{mQueue.sendPacket(mCodecCtx.get())};
if(ret == AVErrorEOF) break;
}
};
auto sender = std::async(std::launch::async, packet_sender);
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(1)
{
if(mMovie.mQuit.load(std::memory_order_relaxed))
{
/* If mQuit is set, drain frames until we can't get more audio,
* indicating we've reached the flush packet and the packet sender
* will also quit.
*/
do {
mSamplesLen = decodeFrame();
mSamplesPos = mSamplesLen;
} while(mSamplesLen > 0);
goto finish;
}
ALenum state;
if(mBufferDataSize > 0)
{
alGetSourcei(mSource, AL_SOURCE_STATE, &state);
/* If mQuit is not set, don't quit even if there's no more audio,
* so what's buffered has a chance to play to the real end.
*/
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<ALuint>(queued) < mBuffers.size())
{
/* Read the next chunk of data, filling the buffer, and queue
* it on the source.
*/
if(!readAudio(samples.get(), static_cast<ALuint>(buffer_len), sync_skip))
break;
const ALuint bufid{mBuffers[mBufferIdx]};
mBufferIdx = static_cast<ALuint>((mBufferIdx+1) % mBuffers.size());
alBufferData(bufid, mFormat, samples.get(), 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(ALenum err{alGetError()})
std::cerr<< "Got AL error: 0x"<<std::hex<<err<<std::dec
<< " ("<<alGetString(err)<<")" <<std::endl;
mSrcCond.wait_for(srclock, sleep_time);
}
finish:
alSourceRewind(mSource);
alSourcei(mSource, AL_BUFFER, 0);
srclock.unlock();
return 0;
}
nanoseconds VideoState::getClock()
{
/* NOTE: This returns incorrect times while not playing. */
std::lock_guard<std::mutex> _{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, AVFrame *frame)
{
if(!mImage)
return;
double aspect_ratio;
int win_w, win_h;
int w, h, x, y;
int frame_width{frame->width - static_cast<int>(frame->crop_left + frame->crop_right)};
int frame_height{frame->height - static_cast<int>(frame->crop_top + frame->crop_bottom)};
if(frame->sample_aspect_ratio.num == 0)
aspect_ratio = 0.0;
else
{
aspect_ratio = av_q2d(frame->sample_aspect_ratio) * frame_width /
frame_height;
}
if(aspect_ratio <= 0.0)
aspect_ratio = static_cast<double>(frame_width) / frame_height;
SDL_GetWindowSize(screen, &win_w, &win_h);
h = win_h;
w = (static_cast<int>(std::rint(h * aspect_ratio)) + 3) & ~3;
if(w > win_w)
{
w = win_w;
h = (static_cast<int>(std::rint(w / aspect_ratio)) + 3) & ~3;
}
x = (win_w - w) / 2;
y = (win_h - h) / 2;
SDL_Rect src_rect{ static_cast<int>(frame->crop_left), static_cast<int>(frame->crop_top),
frame_width, frame_height };
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 && !mMovie.mQuit.load(std::memory_order_relaxed))
{
/* For the first update, ensure the first frame gets shown. */
if(!mFirstUpdate || updated)
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<std::mutex>{mPictQMutex}.unlock();
mPictQCond.notify_one();
return;
}
AVFrame *frame{vp->mFrame.get()};
if(updated)
{
mPictQRead.store(read_idx, std::memory_order_release);
std::unique_lock<std::mutex>{mPictQMutex}.unlock();
mPictQCond.notify_one();
/* allocate or resize the buffer! */
bool fmt_updated{false};
if(!mImage || mWidth != frame->width || mHeight != frame->height)
{
fmt_updated = true;
if(mImage)
SDL_DestroyTexture(mImage);
mImage = SDL_CreateTexture(renderer, SDL_PIXELFORMAT_IYUV, SDL_TEXTUREACCESS_STREAMING,
frame->width, frame->height);
if(!mImage)
std::cerr<< "Failed to create YV12 texture!" <<std::endl;
mWidth = frame->width;
mHeight = frame->height;
}
int frame_width{frame->width - static_cast<int>(frame->crop_left + frame->crop_right)};
int frame_height{frame->height - static_cast<int>(frame->crop_top + frame->crop_bottom)};
if(mFirstUpdate && frame_width > 0 && frame_height > 0)
{
/* For the first update, set the window size to the video size. */
mFirstUpdate = false;
if(frame->sample_aspect_ratio.den != 0)
{
double aspect_ratio = av_q2d(frame->sample_aspect_ratio);
if(aspect_ratio >= 1.0)
frame_width = static_cast<int>(frame_width*aspect_ratio + 0.5);
else if(aspect_ratio > 0.0)
frame_height = static_cast<int>(frame_height/aspect_ratio + 0.5);
}
SDL_SetWindowSize(screen, frame_width, frame_height);
}
if(mImage)
{
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" <<std::endl;
else
{
// Convert the image into YUV format that SDL uses
int w{frame->width};
int h{frame->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<uint8_t*>(pixels);
pict_data[1] = pict_data[0] + w*h;
pict_data[2] = pict_data[1] + w*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<uint8_t**>(frame->data), frame->linesize,
0, h, pict_data, pict_linesize);
SDL_UnlockTexture(mImage);
}
redraw = true;
}
}
if(redraw)
{
/* Show the picture! */
display(screen, renderer, frame);
}
if(updated)
{
auto disp_time = get_avtime();
std::lock_guard<std::mutex> _{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<std::mutex>{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. */
auto packet_sender = [this]()
{
while(1)
{
const int ret{mQueue.sendPacket(mCodecCtx.get())};
if(ret == AVErrorEOF) break;
}
};
auto sender = std::async(std::launch::async, packet_sender);
{
std::lock_guard<std::mutex> _{mDispPtsMutex};
mDisplayPtsTime = get_avtime();
}
auto current_pts = nanoseconds::zero();
while(1)
{
size_t write_idx{mPictQWrite.load(std::memory_order_relaxed)};
Picture *vp{&mPictQ[write_idx]};
/* Retrieve video frame. */
AVFrame *decoded_frame{vp->mFrame.get()};
while(int ret{mQueue.receiveFrame(mCodecCtx.get(), decoded_frame)})
{
if(ret == AVErrorEOF) goto finish;
std::cerr<< "Failed to receive frame: "<<ret <<std::endl;
}
/* Get the PTS for this frame. */
if(decoded_frame->best_effort_timestamp != AVNoPtsValue)
current_pts = duration_cast<nanoseconds>(seconds_d64{av_q2d(mStream->time_base) *
static_cast<double>(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<nanoseconds>(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);
if(write_idx == mPictQRead.load(std::memory_order_acquire))
{
/* Wait until we have space for a new pic */
std::unique_lock<std::mutex> lock{mPictQMutex};
while(write_idx == mPictQRead.load(std::memory_order_acquire))
mPictQCond.wait(lock);
}
}
finish:
mEOS = true;
std::unique_lock<std::mutex> lock{mPictQMutex};
while(!mFinalUpdate) mPictQCond.wait(lock);
return 0;
}
int MovieState::decode_interrupt_cb(void *ctx)
{
return static_cast<MovieState*>(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 "<<mFilename <<std::endl;
return false;
}
mIOContext.reset(avioctx);
/* Open movie file. If avformat_open_input fails it will automatically free
* this context, so don't set it onto a smart pointer yet.
*/
AVFormatContext *fmtctx{avformat_alloc_context()};
fmtctx->pb = mIOContext.get();
fmtctx->interrupt_callback = intcb;
if(avformat_open_input(&fmtctx, mFilename.c_str(), nullptr, nullptr) != 0)
{
std::cerr<< "Failed to open "<<mFilename <<std::endl;
return false;
}
mFormatCtx.reset(fmtctx);
/* Retrieve stream information */
if(avformat_find_stream_info(mFormatCtx.get(), nullptr) < 0)
{
std::cerr<< mFilename<<": failed to find stream info" <<std::endl;
return false;
}
/* Dump information about file onto standard error */
av_dump_format(mFormatCtx.get(), 0, mFilename.c_str(), 0);
mParseThread = std::thread{std::mem_fn(&MovieState::parse_handler), this};
std::unique_lock<std::mutex> slock{mStartupMutex};
while(!mStartupDone) mStartupCond.wait(slock);
return true;
}
void MovieState::setTitle(SDL_Window *window)
{
auto pos1 = mFilename.rfind('/');
auto pos2 = mFilename.rfind('\\');
auto fpos = ((pos1 == std::string::npos) ? pos2 :
(pos2 == std::string::npos) ? pos1 :
std::max(pos1, pos2)) + 1;
SDL_SetWindowTitle(window, (mFilename.substr(fpos)+" - "+AppName).c_str());
}
nanoseconds MovieState::getClock()
{
if(mClockBase == microseconds::min())
return nanoseconds::zero();
return get_avtime() - mClockBase;
}
nanoseconds MovieState::getMasterClock()
{
if(mAVSyncType == SyncMaster::Video && mVideo.mStream)
return mVideo.getClock();
if(mAVSyncType == SyncMaster::Audio && mAudio.mStream)
return mAudio.getClock();
return getClock();
}
nanoseconds MovieState::getDuration()
{ return std::chrono::duration<int64_t,std::ratio<1,AV_TIME_BASE>>(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;
const AVCodec *codec{avcodec_find_decoder(avctx->codec_id)};
if(!codec || avcodec_open2(avctx.get(), codec, nullptr) < 0)
{
std::cerr<< "Unsupported codec: "<<avcodec_get_name(avctx->codec_id)
<< " (0x"<<std::hex<<avctx->codec_id<<std::dec<<")" <<std::endl;
return -1;
}
/* Initialize and start the media type handler */
switch(avctx->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<int>(stream_index);
}
int MovieState::parse_handler()
{
auto &audio_queue = mAudio.mQueue;
auto &video_queue = mVideo.mQueue;
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);
}
{
std::unique_lock<std::mutex> slock{mStartupMutex};
mStartupDone = true;
}
mStartupCond.notify_all();
if(video_index < 0 && audio_index < 0)
{
std::cerr<< mFilename<<": could not open codecs" <<std::endl;
mQuit = true;
}
/* Set the base time 750ms ahead of the current av time. */
mClockBase = get_avtime() + milliseconds{750};
if(audio_index >= 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 */
AVPacketPtr packet{av_packet_alloc()};
while(!mQuit.load(std::memory_order_relaxed))
{
if(av_read_frame(mFormatCtx.get(), packet.get()) < 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.get()))
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.get()))
std::this_thread::sleep_for(milliseconds{100});
}
av_packet_unref(packet.get());
}
/* Finish the queues so the receivers know nothing more is coming. */
video_queue.setFinished();
audio_queue.setFinished();
/* all done - wait for it */
if(mVideoThread.joinable())
mVideoThread.join();
if(mAudioThread.joinable())
mAudioThread.join();
mVideo.mEOS = true;
std::unique_lock<std::mutex> 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;
}
void MovieState::stop()
{
mQuit = true;
mAudio.mQueue.flush();
mVideo.mQueue.flush();
}
// 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<hours>(t).count() << 'h' << std::setfill('0') << std::setw(2)
<< (duration_cast<minutes>(t).count() % 60) << 'm';
else
os << duration_cast<minutes>(t).count() << 'm' << std::setfill('0');
os << std::setw(2) << (duration_cast<seconds>(t).count() % 60) << 's' << std::setw(0)
<< std::setfill(' ');
return os;
}
} // namespace
int main(int argc, char *argv[])
{
std::unique_ptr<MovieState> movState;
if(argc < 2)
{
std::cerr<< "Usage: "<<argv[0]<<" [-device <device name>] [-direct] <files...>" <<std::endl;
return 1;
}
/* Register all formats and codecs */
#if !(LIBAVFORMAT_VERSION_INT >= 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 - <<"<<SDL_GetError() <<std::endl;
return 1;
}
/* Make a window to put our video */
SDL_Window *screen{SDL_CreateWindow(AppName.c_str(), 0, 0, 640, 480, SDL_WINDOW_RESIZABLE)};
if(!screen)
{
std::cerr<< "SDL: could not set video mode - exiting" <<std::endl;
return 1;
}
/* Make a renderer to handle the texture image surface and rendering. */
Uint32 render_flags{SDL_RENDERER_ACCELERATED | SDL_RENDERER_PRESENTVSYNC};
SDL_Renderer *renderer{SDL_CreateRenderer(screen, -1, render_flags)};
if(renderer)
{
SDL_RendererInfo rinf{};
bool ok{false};
/* Make sure the renderer supports IYUV textures. If not, fallback to a
* software renderer. */
if(SDL_GetRendererInfo(renderer, &rinf) == 0)
{
for(Uint32 i{0u};!ok && i < rinf.num_texture_formats;i++)
ok = (rinf.texture_formats[i] == SDL_PIXELFORMAT_IYUV);
}
if(!ok)
{
std::cerr<< "IYUV pixelformat textures not supported on renderer "<<rinf.name <<std::endl;
SDL_DestroyRenderer(renderer);
renderer = nullptr;
}
}
if(!renderer)
{
render_flags = SDL_RENDERER_SOFTWARE | SDL_RENDERER_PRESENTVSYNC;
renderer = SDL_CreateRenderer(screen, -1, render_flags);
}
if(!renderer)
{
std::cerr<< "SDL: could not create renderer - exiting" <<std::endl;
return 1;
}
SDL_SetRenderDrawColor(renderer, 0, 0, 0, 255);
SDL_RenderFillRect(renderer, nullptr);
SDL_RenderPresent(renderer);
/* Open an audio device */
++argv; --argc;
if(InitAL(&argv, &argc))
{
std::cerr<< "Failed to set up audio device" <<std::endl;
return 1;
}
{
auto device = alcGetContextsDevice(alcGetCurrentContext());
if(alcIsExtensionPresent(device, "ALC_SOFT_device_clock"))
{
std::cout<< "Found ALC_SOFT_device_clock" <<std::endl;
alcGetInteger64vSOFT = reinterpret_cast<LPALCGETINTEGER64VSOFT>(
alcGetProcAddress(device, "alcGetInteger64vSOFT")
);
}
}
if(alIsExtensionPresent("AL_SOFT_source_latency"))
{
std::cout<< "Found AL_SOFT_source_latency" <<std::endl;
alGetSourcei64vSOFT = reinterpret_cast<LPALGETSOURCEI64VSOFT>(
alGetProcAddress("alGetSourcei64vSOFT")
);
}
if(alIsExtensionPresent("AL_SOFT_events"))
{
std::cout<< "Found AL_SOFT_events" <<std::endl;
alEventControlSOFT = reinterpret_cast<LPALEVENTCONTROLSOFT>(
alGetProcAddress("alEventControlSOFT"));
alEventCallbackSOFT = reinterpret_cast<LPALEVENTCALLBACKSOFT>(
alGetProcAddress("alEventCallbackSOFT"));
}
if(alIsExtensionPresent("AL_SOFT_callback_buffer"))
{
std::cout<< "Found AL_SOFT_callback_buffer" <<std::endl;
alBufferCallbackSOFT = reinterpret_cast<LPALBUFFERCALLBACKSOFT>(
alGetProcAddress("alBufferCallbackSOFT"));
}
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" <<std::endl;
DirectOutMode = AL_REMIX_UNMATCHED_SOFT;
}
else if(alIsExtensionPresent("AL_SOFT_direct_channels"))
{
std::cout<< "Found AL_SOFT_direct_channels" <<std::endl;
DirectOutMode = AL_DROP_UNMATCHED_SOFT;
}
else
std::cerr<< "AL_SOFT_direct_channels not supported for direct output" <<std::endl;
}
else if(strcmp(argv[fileidx], "-wide") == 0)
{
if(!alIsExtensionPresent("AL_EXT_STEREO_ANGLES"))
std::cerr<< "AL_EXT_STEREO_ANGLES not supported for wide stereo" <<std::endl;
else
{
std::cout<< "Found AL_EXT_STEREO_ANGLES" <<std::endl;
EnableWideStereo = true;
}
}
else if(strcmp(argv[fileidx], "-uhj") == 0)
{
if(!alIsExtensionPresent("AL_SOFT_UHJ"))
std::cerr<< "AL_SOFT_UHJ not supported for UHJ decoding" <<std::endl;
else
{
std::cout<< "Found AL_SOFT_UHJ" <<std::endl;
EnableUhj = true;
}
}
else if(strcmp(argv[fileidx], "-superstereo") == 0)
{
if(!alIsExtensionPresent("AL_SOFT_UHJ"))
std::cerr<< "AL_SOFT_UHJ not supported for Super Stereo decoding" <<std::endl;
else
{
std::cout<< "Found AL_SOFT_UHJ (Super Stereo)" <<std::endl;
EnableSuperStereo = true;
}
}
else if(strcmp(argv[fileidx], "-novideo") == 0)
DisableVideo = true;
else
break;
}
while(fileidx < argc && !movState)
{
movState = std::unique_ptr<MovieState>{new MovieState{argv[fileidx++]}};
if(!movState->prepare()) movState = nullptr;
}
if(!movState)
{
std::cerr<< "Could not start a video" <<std::endl;
return 1;
}
movState->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_WaitEventTimeout is broken, just force a 10ms sleep. */
std::this_thread::sleep_for(milliseconds{10});
auto cur_time = std::chrono::duration_cast<seconds>(movState->getMasterClock());
if(cur_time != last_time)
{
auto end_time = std::chrono::duration_cast<seconds>(movState->getDuration());
std::cout<< " \r "<<PrettyTime{cur_time}<<" / "<<PrettyTime{end_time} <<std::flush;
last_time = cur_time;
}
bool force_redraw{false};
SDL_Event event{};
while(SDL_PollEvent(&event) != 0)
{
switch(event.type)
{
case SDL_KEYDOWN:
switch(event.key.keysym.sym)
{
case SDLK_ESCAPE:
movState->stop();
eom_action = EomAction::Quit;
break;
case SDLK_n:
movState->stop();
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->stop();
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<MovieState>{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;
}
}
movState->mVideo.updateVideo(screen, renderer, force_redraw);
}
std::cerr<< "SDL_WaitEvent error - "<<SDL_GetError() <<std::endl;
return 1;
}