Revert "Testing audio latency compensation"

This reverts commit 2692a7d22c0b691f7f27f969c91e7af05c8ede7d.

3 files changed, 8 insertions, 22 deletions

diff --git a/include/sound.hpp b/include/sound.hpp
index c0d0465..731e787 100644
--- a/include/sound.hpp
+++ b/include/sound.hpp
@@ -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();
 
diff --git a/src/main.cpp b/src/main.cpp
index eb90e91..5aacd8b 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -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)
diff --git a/src/sound.cpp b/src/sound.cpp
index 8be32b2..485548b 100644
--- a/src/sound.cpp
+++ b/src/sound.cpp
@@ -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;
 }