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Fixed sine wave distortion over time.

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Audio distortion after a while caused by loss of precision in dividing a large floating point number resolved by keeping `current_sine_sample` (formelly named `total_samples_generated`) between 0 and freq - 1.
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Nicolas Firmo do Patrocinio Barra 2025-01-17 22:08:07 -03:00 • committed by Sam Lantinga
parent 010f27dc70
commit b95989d14a
2 changed files with 14 additions and 8 deletions
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11
examples/audio/01-simple-playback/simple-playback.c
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@ -14,7 +14,7 @@
static SDL_Window *window = NULL; static SDL_Window *window = NULL;
static SDL_Renderer *renderer = NULL; static SDL_Renderer *renderer = NULL;
static SDL_AudioStream *stream = NULL; static SDL_AudioStream *stream = NULL;
static int total_samples_generated = 0; static int current_sine_sample = 0;
/* This function runs once at startup. */ /* This function runs once at startup. */
SDL_AppResult SDL_AppInit(void **appstate, int argc, char *argv[]) SDL_AppResult SDL_AppInit(void **appstate, int argc, char *argv[])
@ -76,11 +76,14 @@ SDL_AppResult SDL_AppIterate(void *appstate)
/* generate a 440Hz pure tone */ /* generate a 440Hz pure tone */
for (i = 0; i < SDL_arraysize(samples); i++) { for (i = 0; i < SDL_arraysize(samples); i++) {
const int freq = 440; const int freq = 440;
const int phase = (total_samples_generated * freq) % 8000; const int phase = current_sine_sample * freq / 8000.0f;
samples[i] = (float)SDL_sin(phase * 2 * SDL_PI_D / 8000.0); samples[i] = SDL_sinf(phase * 2 * SDL_PI_F);
total_samples_generated++; current_sine_sample++;
} }
/* wrapping around to avoid floating-point errors */
current_sine_sample %= 8000;
/* feed the new data to the stream. It will queue at the end, and trickle out as the hardware needs more data. */ /* feed the new data to the stream. It will queue at the end, and trickle out as the hardware needs more data. */
SDL_PutAudioStreamData(stream, samples, sizeof (samples)); SDL_PutAudioStreamData(stream, samples, sizeof (samples));
} }

11
examples/audio/02-simple-playback-callback/simple-playback-callback.c
View file

@ -17,7 +17,7 @@
static SDL_Window *window = NULL; static SDL_Window *window = NULL;
static SDL_Renderer *renderer = NULL; static SDL_Renderer *renderer = NULL;
static SDL_AudioStream *stream = NULL; static SDL_AudioStream *stream = NULL;
static int total_samples_generated = 0; static int current_sine_sample = 0;
/* this function will be called (usually in a background thread) when the audio stream is consuming data. */ /* this function will be called (usually in a background thread) when the audio stream is consuming data. */
static void SDLCALL FeedTheAudioStreamMore(void *userdata, SDL_AudioStream *astream, int additional_amount, int total_amount) static void SDLCALL FeedTheAudioStreamMore(void *userdata, SDL_AudioStream *astream, int additional_amount, int total_amount)
@ -36,11 +36,14 @@ static void SDLCALL FeedTheAudioStreamMore(void *userdata, SDL_AudioStream *astr
/* generate a 440Hz pure tone */ /* generate a 440Hz pure tone */
for (i = 0; i < total; i++) { for (i = 0; i < total; i++) {
const int freq = 440; const int freq = 440;
const int phase = (total_samples_generated * freq) % 8000; const int phase = current_sine_sample * freq / 8000.0f;
samples[i] = (float)SDL_sin(phase * 2 * SDL_PI_D / 8000.0); samples[i] = SDL_sinf(phase * 2 * SDL_PI_F);
total_samples_generated++; current_sine_sample++;
} }
/* wrapping around to avoid floating-point errors */
current_sine_sample %= 8000;
/* feed the new data to the stream. It will queue at the end, and trickle out as the hardware needs more data. */ /* feed the new data to the stream. It will queue at the end, and trickle out as the hardware needs more data. */
SDL_PutAudioStreamData(astream, samples, total * sizeof (float)); SDL_PutAudioStreamData(astream, samples, total * sizeof (float));
additional_amount -= total; /* subtract what we've just fed the stream. */ additional_amount -= total; /* subtract what we've just fed the stream. */