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Get the audio sync once per update in alffplay

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This commit is contained in:
Chris Robinson
2020-01-13 17:31:35 -08:00
parent a56a6116c0
commit aca1d02024
1 changed files with 27 additions and 29 deletions
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examples/alffplay.cpp
+27 -29
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@@ -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)
{