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Fixed a longstanding bug that caused stereo channels to be flipped incorrectly.

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This was due to the initial audio buffer write position being aligned to the middle of a 4 byte (2 byte left, 2 byte right) audio sample.
In multi-threaded mode, some sort of race condition caused this alignment to be fixed until audio playback was temporarily paused (via pausing, loading a state, etc.)
Audio playback should now be fixed and work consistently in all cases.
This commit is contained in:
SpinDizzy 2020-09-14 06:59:38 +00:00
parent 38c3ff61f9
commit 72b2c0eb28
1 changed files with 22 additions and 8 deletions
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30
Src/OSD/SDL/Audio.cpp
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@ -23,6 +23,11 @@
* Audio.cpp * Audio.cpp
* *
* SDL audio playback. Implements the OSD audio interface. * SDL audio playback. Implements the OSD audio interface.
*
* Buffer sizes and read/write positions must be sample-aligned. A sample is
* defined to encompass both channels so for, e.g., 16-bit audio as used here,
* a sample is 4 bytes. Static assertions are employed to ensure that the
* initial set up of the buffer is correct.
*/ */
#include "Supermodel.h" #include "Supermodel.h"
@ -43,10 +48,10 @@
#define MAX_LATENCY 100 #define MAX_LATENCY 100
static bool enabled = true; // True if sound output is enabled static bool enabled = true; // True if sound output is enabled
static unsigned latency = 20; // Audio latency to use (ie size of audio buffer) as percentage of max buffer size static constexpr unsigned latency = 20; // Audio latency to use (ie size of audio buffer) as percentage of max buffer size
static bool underRunLoop = true; // True if should loop back to beginning of buffer on under-run, otherwise sound is just skipped static constexpr bool underRunLoop = true; // True if should loop back to beginning of buffer on under-run, otherwise sound is just skipped
static unsigned playSamples = 512; // Size (in samples) of callback play buffer static constexpr unsigned playSamples = 512; // Size (in samples) of callback play buffer
static UINT32 audioBufferSize = 0; // Size (in bytes) of audio buffer static UINT32 audioBufferSize = 0; // Size (in bytes) of audio buffer
static INT8 *audioBuffer = NULL; // Audio buffer static INT8 *audioBuffer = NULL; // Audio buffer
@ -195,16 +200,16 @@ static void MixChannels(unsigned numSamples, INT16 *leftBuffer, INT16 *rightBuff
{ {
for (unsigned i = 0; i < numSamples; i++) for (unsigned i = 0; i < numSamples; i++)
{ {
*p++ = leftBuffer[i];
*p++ = rightBuffer[i]; *p++ = rightBuffer[i];
*p++ = leftBuffer[i];
} }
} }
else // stereo as God intended! else // correct stereo
{ {
for (unsigned i = 0; i < numSamples; i++) for (unsigned i = 0; i < numSamples; i++)
{ {
*p++ = rightBuffer[i];
*p++ = leftBuffer[i]; *p++ = leftBuffer[i];
*p++ = rightBuffer[i];
} }
} }
#endif // NUM_CHANNELS #endif // NUM_CHANNELS
@ -241,7 +246,10 @@ bool OpenAudio()
return ErrorLog("Unable to open 44.1KHz 2-channel audio with SDL: %s\n", SDL_GetError()); return ErrorLog("Unable to open 44.1KHz 2-channel audio with SDL: %s\n", SDL_GetError());
// Create audio buffer // Create audio buffer
audioBufferSize = SAMPLE_RATE * BYTES_PER_SAMPLE * latency / MAX_LATENCY; const constexpr uint32_t bufferSize = SAMPLE_RATE * BYTES_PER_SAMPLE * latency / MAX_LATENCY;
static_assert(bufferSize % BYTES_PER_SAMPLE == 0); // must be an integer multiple of the sample size
audioBufferSize = bufferSize;
int minBufferSize = 3 * BYTES_PER_FRAME; int minBufferSize = 3 * BYTES_PER_FRAME;
audioBufferSize = std::max<int>(minBufferSize, audioBufferSize); audioBufferSize = std::max<int>(minBufferSize, audioBufferSize);
audioBuffer = new(std::nothrow) INT8[audioBufferSize]; audioBuffer = new(std::nothrow) INT8[audioBufferSize];
@ -254,7 +262,13 @@ bool OpenAudio()
// Set initial play position to be beginning of buffer and initial write position to be half-way into buffer // Set initial play position to be beginning of buffer and initial write position to be half-way into buffer
playPos = 0; playPos = 0;
writePos = std::min<int>(audioBufferSize - BYTES_PER_FRAME, (BYTES_PER_FRAME + audioBufferSize) / 2); const constexpr uint32_t endOfBuffer = bufferSize - BYTES_PER_FRAME;
const constexpr uint32_t midpointAfterFirstFrameUnaligned = BYTES_PER_FRAME + (bufferSize - BYTES_PER_FRAME) / 2;
const constexpr uint32_t extraPaddingNeeded = (BYTES_PER_SAMPLE - midpointAfterFirstFrameUnaligned % BYTES_PER_SAMPLE) % BYTES_PER_SAMPLE;
const constexpr uint32_t midpointAfterFirstFrame = midpointAfterFirstFrameUnaligned + extraPaddingNeeded;
static_assert(endOfBuffer % BYTES_PER_SAMPLE == 0); // make sure we are aligned to a sample boundary otherwise underrun/overrun adjustment will end up shifting playback by one channel causing stereo to flip
static_assert(midpointAfterFirstFrame % BYTES_PER_SAMPLE == 0);
writePos = std::min<int>(endOfBuffer, midpointAfterFirstFrame);
writeWrapped = false; writeWrapped = false;
// Reset counters // Reset counters