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Fixed integer overflow in audio buffer calculations, added workaround to speed up audio if too many samples get queued somehow (prevents audio lag bug)

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Mr-Wiseguy 2024-05-04 00:28:42 -04:00
parent 55aa02c4b3
commit 3fdb2dc001
1 changed files with 21 additions and 6 deletions
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29
src/main/main.cpp
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@ -160,6 +160,8 @@ constexpr uint32_t duplicated_input_frames = 4;
// The number of output frames to skip for playback (to avoid playing duplicate inputs twice).
static uint32_t discarded_output_frames;
constexpr uint32_t bytes_per_frame = input_channels * sizeof(float);
void queue_samples(int16_t* audio_data, size_t sample_count) {
// Buffer for holding the output of swapping the audio channels. This is reused across
// calls to reduce runtime allocations.
@ -203,18 +205,31 @@ void queue_samples(int16_t* audio_data, size_t sample_count) {
throw std::runtime_error("Error using SDL audio converter");
}
// Queue the swapped audio data.
// Offset the data start by only half the discarded frame count as the other half of the discarded frames are at the end of the buffer.
SDL_QueueAudio(audio_device, swap_buffer.data() + output_channels * discarded_output_frames / 2,
audio_convert.len_cvt - output_channels * discarded_output_frames * sizeof(swap_buffer[0]));
uint64_t cur_queued_microseconds = uint64_t(SDL_GetQueuedAudioSize(audio_device)) / bytes_per_frame * 1000000 / sample_rate;
uint32_t num_bytes_to_queue = audio_convert.len_cvt - output_channels * discarded_output_frames * sizeof(swap_buffer[0]);
float* samples_to_queue = swap_buffer.data() + output_channels * discarded_output_frames / 2;
// Prevent audio latency from building up by skipping samples in incoming audio when too many samples are already queued.
// Skip samples based on how many microseconds of samples are queued already.
uint32_t skip_factor = cur_queued_microseconds / 100000;
if (skip_factor != 0) {
uint32_t skip_ratio = 1 << skip_factor;
num_bytes_to_queue /= skip_ratio;
for (size_t i = 0; i < num_bytes_to_queue / (output_channels * sizeof(swap_buffer[0])); i++) {
samples_to_queue[2 * i + 0] = samples_to_queue[2 * skip_ratio * i + 0];
samples_to_queue[2 * i + 1] = samples_to_queue[2 * skip_ratio * i + 1];
}
}
constexpr uint32_t bytes_per_frame = input_channels * sizeof(float);
// Queue the swapped audio data.
// Offset the data start by only half the discarded frame count as the other half of the discarded frames are at the end of the buffer.
SDL_QueueAudio(audio_device, samples_to_queue, num_bytes_to_queue);
}
size_t get_frames_remaining() {
constexpr float buffer_offset_frames = 1.0f;
// Get the number of remaining buffered audio bytes.
uint32_t buffered_byte_count = SDL_GetQueuedAudioSize(audio_device);
uint64_t buffered_byte_count = SDL_GetQueuedAudioSize(audio_device);
// Scale the byte count based on the ratio of sample rates and channel counts.
buffered_byte_count = buffered_byte_count * 2 * sample_rate / output_sample_rate / output_channels;
@ -231,7 +246,7 @@ size_t get_frames_remaining() {
buffered_byte_count = 0;
}
// Convert from byte count to sample count.
return buffered_byte_count / bytes_per_frame;
return static_cast<uint32_t>(buffered_byte_count / bytes_per_frame);
}
void update_audio_converter() {