use callback model instead of blocking read/write.
fixes delay on linux. currently not implementing the overflow bit, not sure it's needed since we manage our own buffer now.
This commit is contained in:
parent
1d6bd3248c
commit
c528978608
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@ -40,19 +40,64 @@ void CEXIMic::StreamInit()
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// Setup the wonderful c-interfaced lib...
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if ((pa_error = Pa_Initialize()) != paNoError)
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StreamLog("Pa_Initialize");
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stream_wpos = stream_rpos = 0;
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memset(stream_buffer, 0, stream_size * sample_size);
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}
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void CEXIMic::StreamTerminate()
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{
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Pa_AbortStream(pa_stream);
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if ((pa_error = Pa_Terminate()) != paNoError)
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StreamLog("Pa_Terminate");
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}
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static int Pa_Callback(const void *inputBuffer, void *outputBuffer,
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unsigned long framesPerBuffer,
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const PaStreamCallbackTimeInfo *timeInfo,
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PaStreamCallbackFlags statusFlags,
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void *userData)
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{
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(void)outputBuffer;
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(void)timeInfo;
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CEXIMic *mic = (CEXIMic *)userData;
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std::lock_guard<std::mutex> lk(mic->ring_lock);
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if (mic->stream_wpos + mic->buff_size_samples >= mic->stream_size)
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mic->stream_wpos = 0;
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s16 *buff_in = (s16 *)inputBuffer;
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s16 *buff_out = &mic->stream_buffer[mic->stream_wpos];
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if (buff_in == NULL)
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{
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for (int i = 0; i < mic->buff_size_samples; i++)
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{
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buff_out[i] = 0;
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}
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}
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else
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{
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for (int i = 0; i < mic->buff_size_samples; i++)
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{
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buff_out[i] = buff_in[i];
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}
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}
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mic->stream_wpos += mic->buff_size_samples;
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mic->stream_wpos %= mic->stream_size;
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return paContinue;
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}
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void CEXIMic::StreamStart()
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{
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// Open stream with current parameters
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pa_error = Pa_OpenDefaultStream(&pa_stream, 1, 0, paInt16,
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sample_rate, buff_size_samples, NULL, NULL);
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sample_rate, buff_size_samples, Pa_Callback, this);
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StreamLog("Pa_OpenDefaultStream");
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pa_error = Pa_StartStream(pa_stream);
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StreamLog("Pa_StartStream");
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@ -67,23 +112,26 @@ void CEXIMic::StreamStop()
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void CEXIMic::StreamReadOne()
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{
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// Returns num samples or error
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pa_error = Pa_GetStreamReadAvailable(pa_stream);
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if (pa_error >= buff_size_samples)
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{
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pa_error = Pa_ReadStream(pa_stream, ring_buffer, buff_size_samples);
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std::lock_guard<std::mutex> lk(ring_lock);
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if (pa_error == paInputOverflowed)
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{
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status.buff_ovrflw = 1;
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// Input overflowed - is re-setting the stream the only to recover?
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StreamStop();
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StreamStart();
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}
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else if (pa_error != paNoError)
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{
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StreamLog("Pa_ReadStream");
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}
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int samples_avail = (stream_wpos > stream_rpos) ?
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stream_wpos - stream_rpos :
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stream_size - stream_rpos + stream_wpos;
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if (samples_avail >= buff_size_samples)
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{
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s16 *last_buffer = &stream_buffer[stream_rpos];
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memcpy(ring_buffer, last_buffer, buff_size);
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stream_rpos += buff_size_samples;
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stream_rpos %= stream_size;
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// TODO: if overflow bit matters, find a nice way
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//if (samples_avail >= buff_size_samples * 2)
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//{
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// status.buff_ovrflw = 1;
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// stream_rpos = stream_wpos = 0;
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//}
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}
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}
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@ -91,13 +139,11 @@ void CEXIMic::StreamReadOne()
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// This works by opening and starting a portaudio input stream when the is_active
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// bit is set. The interrupt is scheduled in the future based on sample rate and
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// buffer size settings. When the console handles the interrupt, it will send
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// cmdGetBuffer, which is when we actually read data from a buffer portaudio fills
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// in the background (ie on demand instead of realtime). Because of this we need
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// to clear portaudio's buffer if emulation speed drops below realtime, or else
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// a bad audio lag develops. It's actually kind of convenient because it gives
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// us a way to detect if buff_ovrflw should be set.
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// cmdGetBuffer, which is when we actually read data from a buffer filled
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// in the background by Pa_Callback.
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u8 const CEXIMic::exi_id[] = { 0, 0x0a, 0, 0, 0 };
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int const CEXIMic::stream_size = sizeof(stream_buffer) / sizeof(*stream_buffer);
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CEXIMic::CEXIMic(int index)
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: slot(index)
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@ -211,13 +257,13 @@ void CEXIMic::TransferByte(u8 &byte)
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break;
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case cmdGetBuffer:
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if (ring_pos == 0)
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{
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// Can set buff_ovrflw
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if (ring_pos == 0)
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StreamReadOne();
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}
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byte = ring_buffer[ring_pos ^ 1];
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ring_pos = (ring_pos + 1) % buff_size;
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}
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break;
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default:
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@ -20,6 +20,8 @@
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#if HAVE_PORTAUDIO
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#include "StdMutex.h"
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class CEXIMic : public IEXIDevice
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{
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public:
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@ -68,11 +70,6 @@ private:
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};
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UStatus status;
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// status bits converted to nice numbers
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int sample_rate;
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int buff_size;
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int buff_size_samples;
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// 64 is the max size, can be 16 or 32 as well
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int ring_pos;
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u8 ring_buffer[64 * sample_size];
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@ -93,6 +90,21 @@ private:
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void StreamStop();
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void StreamReadOne();
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public:
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std::mutex ring_lock;
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// status bits converted to nice numbers
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int sample_rate;
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int buff_size;
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int buff_size_samples;
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// Arbitrarily small ringbuffer used by audio input backend in order to
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// keep delay tolerable
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s16 stream_buffer[64 * sample_size * 500];
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static int const stream_size;
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int stream_wpos;
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int stream_rpos;
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protected:
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virtual void TransferByte(u8 &byte);
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};
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