From b95989d14a195df3d4eb93d6e9130c5f0fee64e3 Mon Sep 17 00:00:00 2001
From: Nicolas Firmo do Patrocinio Barra <[EMAIL REDACTED]>
Date: Fri, 17 Jan 2025 22:08:07 -0300
Subject: [PATCH] Fixed sine wave distortion over time.
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.
---
examples/audio/01-simple-playback/simple-playback.c | 11 +++++++----
.../simple-playback-callback.c | 11 +++++++----
2 files changed, 14 insertions(+), 8 deletions(-)
diff --git a/examples/audio/01-simple-playback/simple-playback.c b/examples/audio/01-simple-playback/simple-playback.c
index 0661de1d18c6a..8fb4fe49017fb 100644
--- a/examples/audio/01-simple-playback/simple-playback.c
+++ b/examples/audio/01-simple-playback/simple-playback.c
@@ -14,7 +14,7 @@
static SDL_Window *window = NULL;
static SDL_Renderer *renderer = NULL;
static SDL_AudioStream *stream = NULL;
-static int total_samples_generated = 0;
+static int current_sine_sample = 0;
/* This function runs once at startup. */
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 */
for (i = 0; i < SDL_arraysize(samples); i++) {
const int freq = 440;
- const int phase = (total_samples_generated * freq) % 8000;
- samples[i] = (float)SDL_sin(phase * 2 * SDL_PI_D / 8000.0);
- total_samples_generated++;
+ const int phase = current_sine_sample * freq / 8000.0f;
+ samples[i] = SDL_sinf(phase * 2 * SDL_PI_F);
+ 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. */
SDL_PutAudioStreamData(stream, samples, sizeof (samples));
}
diff --git a/examples/audio/02-simple-playback-callback/simple-playback-callback.c b/examples/audio/02-simple-playback-callback/simple-playback-callback.c
index fe79ffecea088..27d570951cd9d 100644
--- a/examples/audio/02-simple-playback-callback/simple-playback-callback.c
+++ b/examples/audio/02-simple-playback-callback/simple-playback-callback.c
@@ -17,7 +17,7 @@
static SDL_Window *window = NULL;
static SDL_Renderer *renderer = 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. */
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 */
for (i = 0; i < total; i++) {
const int freq = 440;
- const int phase = (total_samples_generated * freq) % 8000;
- samples[i] = (float)SDL_sin(phase * 2 * SDL_PI_D / 8000.0);
- total_samples_generated++;
+ const int phase = current_sine_sample * freq / 8000.0f;
+ samples[i] = SDL_sinf(phase * 2 * SDL_PI_F);
+ 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. */
SDL_PutAudioStreamData(astream, samples, total * sizeof (float));
additional_amount -= total; /* subtract what we've just fed the stream. */