fix: audio resampler_out buffer overflow. static queue element

This commit is contained in:
awalol
2026-04-30 14:44:25 +08:00
parent d1d54eaa41
commit 5b04cbda80
+5 -5
View File
@@ -75,7 +75,7 @@ void audio_loop() {
audio_buf[audio_buf_pos++] = raw[i * INPUT_CHANNELS] / 32768.0f * (volume[0] - 1.0f); audio_buf[audio_buf_pos++] = raw[i * INPUT_CHANNELS] / 32768.0f * (volume[0] - 1.0f);
audio_buf[audio_buf_pos++] = raw[i * INPUT_CHANNELS + 1] / 32768.0f * (volume[0] - 1.0f); audio_buf[audio_buf_pos++] = raw[i * INPUT_CHANNELS + 1] / 32768.0f * (volume[0] - 1.0f);
if (audio_buf_pos == 512 * 2) { if (audio_buf_pos == 512 * 2) {
audio_raw_element element{}; static audio_raw_element element{};
memcpy(element.data,audio_buf,512 * 2 * 4); memcpy(element.data,audio_buf,512 * 2 * 4);
if (queue_is_full(&audio_fifo)){ if (queue_is_full(&audio_fifo)){
queue_try_remove(&audio_fifo,NULL); queue_try_remove(&audio_fifo,NULL);
@@ -91,7 +91,7 @@ void audio_loop() {
} }
// 3. 48kHz -> 3kHz 重采样 // 3. 48kHz -> 3kHz 重采样
WDL_ResampleSample out_buf[SAMPLE_SIZE]; // 64 floats = 32帧 × 2ch static WDL_ResampleSample out_buf[SAMPLE_SIZE]; // 64 floats = 32帧 × 2ch
int out_frames = resampler.ResampleOut(out_buf, nframes, SAMPLE_SIZE / OUTPUT_CHANNELS, OUTPUT_CHANNELS); int out_frames = resampler.ResampleOut(out_buf, nframes, SAMPLE_SIZE / OUTPUT_CHANNELS, OUTPUT_CHANNELS);
static int8_t haptic_buf[SAMPLE_SIZE]; static int8_t haptic_buf[SAMPLE_SIZE];
@@ -169,7 +169,7 @@ void core1_entry() {
resampler_audio.SetFeedMode(true); resampler_audio.SetFeedMode(true);
while (true) { while (true) {
audio_raw_element audio_element{}; static audio_raw_element audio_element{};
queue_remove_blocking(&audio_fifo,&audio_element); queue_remove_blocking(&audio_fifo,&audio_element);
// 将 512 frames 重采样成 480 frames 以解决噪音问题。感谢 @Junhoo // 将 512 frames 重采样成 480 frames 以解决噪音问题。感谢 @Junhoo
WDL_ResampleSample *in_buf; WDL_ResampleSample *in_buf;
@@ -177,9 +177,9 @@ void core1_entry() {
for (int i = 0; i < nframes * 2;i++) { for (int i = 0; i < nframes * 2;i++) {
in_buf[i] = audio_element.data[i]; in_buf[i] = audio_element.data[i];
} }
WDL_ResampleSample out_buf[200]; static WDL_ResampleSample out_buf[480 * 2];
resampler_audio.ResampleOut(out_buf,nframes,480,2); resampler_audio.ResampleOut(out_buf,nframes,480,2);
opus_element opus_element{}; static opus_element opus_element{};
(void)opus_encode_float(encoder,out_buf,480,opus_element.data,200); (void)opus_encode_float(encoder,out_buf,480,opus_element.data,200);
if (queue_is_full(&opus_fifo)) { if (queue_is_full(&opus_fifo)) {
queue_try_remove(&opus_fifo,NULL); queue_try_remove(&opus_fifo,NULL);