diff options
Diffstat (limited to 'examples/alffplay.cpp')
| -rw-r--r-- | examples/alffplay.cpp | 56 |
1 files changed, 27 insertions, 29 deletions
diff --git a/examples/alffplay.cpp b/examples/alffplay.cpp index a8f692a7..d7031774 100644 --- a/examples/alffplay.cpp +++ b/examples/alffplay.cpp @@ -348,7 +348,7 @@ struct AudioState { int getSync(); int decodeFrame(); - bool readAudio(uint8_t *samples, unsigned int length); + bool readAudio(uint8_t *samples, unsigned int length, int *sample_skip); int handler(); }; @@ -641,9 +641,8 @@ static void sample_dup(uint8_t *out, const uint8_t *in, unsigned int count, size } -bool AudioState::readAudio(uint8_t *samples, unsigned int length) +bool AudioState::readAudio(uint8_t *samples, unsigned int length, int *sample_skip) { - int sample_skip{getSync()}; unsigned int audio_size{0}; /* Read the next chunk of data, refill the buffer, and queue it @@ -657,8 +656,8 @@ bool AudioState::readAudio(uint8_t *samples, unsigned int length) if(frame_len <= 0) break; mSamplesLen = frame_len; - mSamplesPos = std::min(mSamplesLen, sample_skip); - sample_skip -= mSamplesPos; + 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 @@ -714,8 +713,7 @@ bool AudioState::readAudio(uint8_t *samples, unsigned int length) #ifdef AL_SOFT_events void AL_APIENTRY AudioState::EventCallback(ALenum eventType, ALuint object, ALuint param, - ALsizei length, const ALchar *message, - void *userParam) + ALsizei length, const ALchar *message, void *userParam) { auto self = static_cast<AudioState*>(userParam); @@ -732,14 +730,15 @@ void AL_APIENTRY AudioState::EventCallback(ALenum eventType, ALuint object, ALui std::cout<< "\n---- AL Event on AudioState "<<self<<" ----\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_ERROR_SOFT: std::cout<< "API error"; break; - case AL_EVENT_TYPE_PERFORMANCE_SOFT: std::cout<< "Performance"; break; - case AL_EVENT_TYPE_DEPRECATED_SOFT: std::cout<< "Deprecated"; 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; + 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_ERROR_SOFT: std::cout<< "API error"; break; + case AL_EVENT_TYPE_PERFORMANCE_SOFT: std::cout<< "Performance"; break; + case AL_EVENT_TYPE_DEPRECATED_SOFT: std::cout<< "Deprecated"; 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" @@ -768,8 +767,7 @@ int AudioState::handler() const std::array<ALenum,6> evt_types{{ AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT, AL_EVENT_TYPE_ERROR_SOFT, AL_EVENT_TYPE_PERFORMANCE_SOFT, AL_EVENT_TYPE_DEPRECATED_SOFT, - AL_EVENT_TYPE_DISCONNECTED_SOFT - }}; + AL_EVENT_TYPE_DISCONNECTED_SOFT}}; if(alEventControlSOFT) { alEventControlSOFT(evt_types.size(), evt_types.data(), AL_TRUE); @@ -963,33 +961,33 @@ int AudioState::handler() * 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); #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::vector<double> mtx(64*64, 0.0); + 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; - swr_set_matrix(mSwresCtx.get(), mtx.data(), 64); } else #endif { + 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. */ - std::vector<double> mtx(64*64, 0.0); 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); } + swr_set_matrix(mSwresCtx.get(), mtx.data(), 64); } else mSwresCtx.reset(swr_alloc_set_opts(nullptr, @@ -1010,9 +1008,9 @@ int AudioState::handler() if(DirectOutMode) alSourcei(mSource, AL_DIRECT_CHANNELS_SOFT, DirectOutMode); - if (EnableWideStereo) { - ALfloat angles[2] = {static_cast<ALfloat>(M_PI / 3.0), - static_cast<ALfloat>(-M_PI / 3.0)}; + 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); } @@ -1063,8 +1061,9 @@ int AudioState::handler() while(alGetError() == AL_NO_ERROR && !mMovie.mQuit.load(std::memory_order_relaxed) && mConnected.test_and_set(std::memory_order_relaxed)) { + ALint processed, queued, state; + /* First remove any processed buffers. */ - ALint processed; alGetSourcei(mSource, AL_BUFFERS_PROCESSED, &processed); while(processed > 0) { @@ -1075,7 +1074,7 @@ int AudioState::handler() } /* Refill the buffer queue. */ - ALint queued; + int sync_skip{getSync()}; alGetSourcei(mSource, AL_BUFFERS_QUEUED, &queued); while(static_cast<ALuint>(queued) < mBuffers.size()) { @@ -1088,7 +1087,7 @@ int AudioState::handler() { auto ptr = static_cast<uint8_t*>(alMapBufferSOFT(bufid, 0, buffer_len, AL_MAP_WRITE_BIT_SOFT)); - bool got_audio{readAudio(ptr, static_cast<unsigned int>(buffer_len))}; + bool got_audio{readAudio(ptr, static_cast<unsigned int>(buffer_len), &sync_skip)}; alUnmapBufferSOFT(bufid); if(!got_audio) break; } @@ -1096,7 +1095,7 @@ int AudioState::handler() #endif { auto ptr = static_cast<uint8_t*>(samples); - if(!readAudio(ptr, static_cast<unsigned int>(buffer_len))) + if(!readAudio(ptr, static_cast<unsigned int>(buffer_len), &sync_skip)) break; alBufferData(bufid, mFormat, samples, buffer_len, mCodecCtx->sample_rate); } @@ -1109,7 +1108,6 @@ int AudioState::handler() break; /* Check that the source is playing. */ - ALint state; alGetSourcei(mSource, AL_SOURCE_STATE, &state); if(state == AL_STOPPED) { |