diff options
author | Chris Robinson <[email protected]> | 2021-12-24 14:48:58 -0800 |
---|---|---|
committer | Chris Robinson <[email protected]> | 2021-12-24 14:48:58 -0800 |
commit | 0886d8efa7704a46f5fbeffecbf7d2985c92af68 (patch) | |
tree | c7dd2f0a2e893eafb0f7a458d24b21f3ceb1ea42 /examples | |
parent | f1aa10ff0b710d6647f7ce728de5a8d0e5b8a45f (diff) |
Use the audio clock as the default in alffplay
Diffstat (limited to 'examples')
-rw-r--r-- | examples/alffplay.cpp | 108 |
1 files changed, 59 insertions, 49 deletions
diff --git a/examples/alffplay.cpp b/examples/alffplay.cpp index 4c505c57..ebb8ac76 100644 --- a/examples/alffplay.cpp +++ b/examples/alffplay.cpp @@ -164,7 +164,7 @@ enum class SyncMaster { Video, External, - Default = External + Default = Audio }; @@ -349,8 +349,11 @@ struct AudioState { } #ifdef AL_SOFT_events - static void AL_APIENTRY EventCallback(ALenum eventType, ALuint object, ALuint param, - ALsizei length, const ALchar *message, void *userParam); + 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); #endif #ifdef AL_SOFT_callback_buffer static ALsizei AL_APIENTRY bufferCallbackC(void *userptr, void *data, ALsizei size) @@ -887,22 +890,20 @@ void AudioState::readAudio(int sample_skip) #ifdef AL_SOFT_events -void AL_APIENTRY AudioState::EventCallback(ALenum eventType, ALuint object, ALuint param, - ALsizei length, const ALchar *message, void *userParam) +void AL_APIENTRY AudioState::eventCallback(ALenum eventType, ALuint object, ALuint param, + ALsizei length, const ALchar *message) { - auto self = static_cast<AudioState*>(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<std::mutex>{self->mSrcMutex}.unlock(); - self->mSrcCond.notify_one(); + std::unique_lock<std::mutex>{mSrcMutex}.unlock(); + mSrcCond.notify_one(); return; } - std::cout<< "\n---- AL Event on AudioState "<<self<<" ----\nEvent: "; + std::cout<< "\n---- AL Event on AudioState "<<this<<" ----\nEvent: "; switch(eventType) { case AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT: std::cout<< "Buffer completed"; break; @@ -921,10 +922,10 @@ void AL_APIENTRY AudioState::EventCallback(ALenum eventType, ALuint object, ALui if(eventType == AL_EVENT_TYPE_DISCONNECTED_SOFT) { { - std::lock_guard<std::mutex> lock{self->mSrcMutex}; - self->mConnected.clear(std::memory_order_release); + std::lock_guard<std::mutex> lock{mSrcMutex}; + mConnected.clear(std::memory_order_release); } - self->mSrcCond.notify_one(); + mSrcCond.notify_one(); } } #endif @@ -963,15 +964,32 @@ int AudioState::handler() milliseconds sleep_time{AudioBufferTime / 3}; #ifdef AL_SOFT_events - 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}}; - if(alEventControlSOFT) - { - alEventControlSOFT(evt_types.size(), evt_types.data(), AL_TRUE); - alEventCallbackSOFT(EventCallback, this); - sleep_time = AudioBufferTotalTime; - } + 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}; #endif #ifdef AL_SOFT_bformat_ex const bool has_bfmt_ex{alIsExtensionPresent("AL_SOFT_bformat_ex") != AL_FALSE}; @@ -991,6 +1009,9 @@ int AudioState::handler() } #endif + std::unique_ptr<uint8_t[]> samples; + ALsizei buffer_len{0}; + /* Find a suitable format for OpenAL. */ mDstChanLayout = 0; mFormat = AL_NONE; @@ -1164,8 +1185,6 @@ int AudioState::handler() mFormat = FormatStereo16; } } - void *samples{nullptr}; - ALsizei buffer_len{0}; mSamples = nullptr; mSamplesMax = 0; @@ -1176,7 +1195,7 @@ int AudioState::handler() if(!mDecodedFrame) { std::cerr<< "Failed to allocate audio frame" <<std::endl; - goto finish; + return 0; } if(!mDstChanLayout) @@ -1231,7 +1250,7 @@ int AudioState::handler() if(!mSwresCtx || swr_init(mSwresCtx.get()) != 0) { std::cerr<< "Failed to initialize audio converter" <<std::endl; - goto finish; + return 0; } alGenBuffers(static_cast<ALsizei>(mBuffers.size()), mBuffers.data()); @@ -1260,9 +1279,10 @@ int AudioState::handler() #endif if(alGetError() != AL_NO_ERROR) - goto finish; + return 0; #ifdef AL_SOFT_callback_buffer + bool callback_ok{false}; if(alBufferCallbackSOFT) { alBufferCallbackSOFT(mBuffers[0], mFormat, mCodecCtx->sample_rate, bufferCallbackC, this, @@ -1272,28 +1292,29 @@ int AudioState::handler() { fprintf(stderr, "Failed to set buffer callback\n"); alSourcei(mSource, AL_BUFFER, 0); - buffer_len = static_cast<int>(duration_cast<seconds>(mCodecCtx->sample_rate * - AudioBufferTime).count() * mFrameSize); } else { mBufferDataSize = static_cast<size_t>(duration_cast<seconds>(mCodecCtx->sample_rate * AudioBufferTotalTime).count()) * mFrameSize; - mBufferData.reset(new uint8_t[mBufferDataSize]); + mBufferData = std::make_unique<uint8_t[]>(mBufferDataSize); + std::fill_n(mBufferData.get(), mBufferDataSize, uint8_t{}); + mReadPos.store(0, std::memory_order_relaxed); - mWritePos.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; } } - else + if(!callback_ok) #endif buffer_len = static_cast<int>(duration_cast<seconds>(mCodecCtx->sample_rate * AudioBufferTime).count() * mFrameSize); if(buffer_len > 0) - samples = av_malloc(static_cast<ALuint>(buffer_len)); + samples = std::make_unique<uint8_t[]>(static_cast<ALuint>(buffer_len)); /* Prefill the codec buffer. */ do { @@ -1351,14 +1372,14 @@ int AudioState::handler() /* Read the next chunk of data, filling the buffer, and queue * it on the source. */ - const bool got_audio{readAudio(static_cast<uint8_t*>(samples), - static_cast<ALuint>(buffer_len), sync_skip)}; + const bool got_audio{readAudio(samples.get(), static_cast<ALuint>(buffer_len), + sync_skip)}; if(!got_audio) break; const ALuint bufid{mBuffers[mBufferIdx]}; mBufferIdx = static_cast<ALuint>((mBufferIdx+1) % mBuffers.size()); - alBufferData(bufid, mFormat, samples, buffer_len, mCodecCtx->sample_rate); + alBufferData(bufid, mFormat, samples.get(), buffer_len, mCodecCtx->sample_rate); alSourceQueueBuffers(mSource, 1, &bufid); ++queued; } @@ -1403,17 +1424,6 @@ int AudioState::handler() 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; } @@ -1753,9 +1763,9 @@ nanoseconds MovieState::getClock() nanoseconds MovieState::getMasterClock() { - if(mAVSyncType == SyncMaster::Video) + if(mAVSyncType == SyncMaster::Video && mVideo.mStream) return mVideo.getClock(); - if(mAVSyncType == SyncMaster::Audio) + if(mAVSyncType == SyncMaster::Audio && mAudio.mStream) return mAudio.getClock(); return getClock(); } |