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Revert "Testing audio latency compensation"

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This reverts commit 2692a7d22c.
This commit is contained in:
dec05eba
2024-04-14 00:53:15 +02:00
parent 2692a7d22c
commit b554628099
3 changed files with 8 additions and 22 deletions
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2
include/sound.hpp
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@@ -55,7 +55,7 @@ void sound_device_close(SoundDevice *device);
Returns the next chunk of audio into @buffer.
Returns the number of frames read, or a negative value on failure.
*/
int sound_device_read_next_chunk(SoundDevice *device, void **buffer, double timeout_sec, double *latency_sec);
int sound_device_read_next_chunk(SoundDevice *device, void **buffer, double timeout_sec);
std::vector<AudioInput> get_pulseaudio_inputs();

9
src/main.cpp
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@@ -2417,15 +2417,12 @@ int main(int argc, char **argv) {
while(running) {
void *sound_buffer;
int sound_buffer_size = -1;
double latency_seconds = 0.0;
const double time_before_read = clock_get_monotonic_seconds();
if(audio_device.sound_device.handle)
sound_buffer_size = sound_device_read_next_chunk(&audio_device.sound_device, &sound_buffer, 1.0, &latency_seconds);
sound_buffer_size = sound_device_read_next_chunk(&audio_device.sound_device, &sound_buffer, timeout_sec);
const bool got_audio_data = sound_buffer_size >= 0;
const double time_after_read = clock_get_monotonic_seconds();
latency_seconds = time_after_read - time_before_read; // TODO: Remove this
const double this_audio_frame_time = (time_after_read - paused_time_offset) - latency_seconds;
const double this_audio_frame_time = clock_get_monotonic_seconds() - paused_time_offset;
if(paused) {
if(got_audio_data)

19
src/sound.cpp
View File

@@ -41,7 +41,6 @@ struct pa_handle {
size_t output_index, output_length;
int operation_success;
double latency_seconds;
};
static void pa_sound_device_free(pa_handle *s) {
@@ -80,9 +79,8 @@ static pa_handle* pa_sound_device_new(const char *server,
p->read_data = NULL;
p->read_length = 0;
p->read_index = 0;
p->latency_seconds = 0.0;
const int buffer_size = attr->fragsize;
const int buffer_size = attr->maxlength;
void *buffer = malloc(buffer_size);
if(!buffer) {
fprintf(stderr, "failed to allocate buffer for audio\n");
@@ -163,9 +161,6 @@ static int pa_sound_device_read(pa_handle *p, double timeout_seconds) {
bool success = false;
int r = 0;
int *rerror = &r;
pa_usec_t latency = 0;
int negative = 0;
CHECK_DEAD_GOTO(p, rerror, fail);
while (p->output_index < p->output_length) {
@@ -198,10 +193,6 @@ static int pa_sound_device_read(pa_handle *p, double timeout_seconds) {
CHECK_DEAD_GOTO(p, rerror, fail);
continue;
}
if(pa_stream_get_latency(p->stream, &latency, &negative) >= 0) {
p->latency_seconds = (negative ? -(int64_t)latency : (int64_t)latency) * 0.000001;
}
}
const size_t space_free_in_output_buffer = p->output_length - p->output_index;
@@ -264,8 +255,8 @@ int sound_device_get_by_name(SoundDevice *device, const char *device_name, const
buffer_attr.tlength = -1;
buffer_attr.prebuf = -1;
buffer_attr.minreq = -1;
buffer_attr.maxlength = -1;
buffer_attr.fragsize = period_frame_size * audio_format_to_get_bytes_per_sample(audio_format) * num_channels; // 2/4 bytes/sample, @num_channels channels
buffer_attr.maxlength = period_frame_size * audio_format_to_get_bytes_per_sample(audio_format) * num_channels; // 2/4 bytes/sample, @num_channels channels
buffer_attr.fragsize = buffer_attr.maxlength;
int error = 0;
pa_handle *handle = pa_sound_device_new(nullptr, description, device_name, description, &ss, &buffer_attr, &error);
@@ -285,15 +276,13 @@ void sound_device_close(SoundDevice *device) {
device->handle = NULL;
}
int sound_device_read_next_chunk(SoundDevice *device, void **buffer, double timeout_sec, double *latency_sec) {
int sound_device_read_next_chunk(SoundDevice *device, void **buffer, double timeout_sec) {
pa_handle *pa = (pa_handle*)device->handle;
if(pa_sound_device_read(pa, timeout_sec) < 0) {
//fprintf(stderr, "pa_simple_read() failed: %s\n", pa_strerror(error));
*latency_sec = 0.0;
return -1;
}
*buffer = pa->output_data;
*latency_sec = pa->latency_seconds;
return device->frames;
}