Make the UI show free disk space, and a rough estimation of for how much longer we...

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index 9e281cbdf72583470a1289b8030cdac7d042effa..1a4b2cf141ab1562e9eae438a3bc458ded7af476 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -37,6 +37,7 @@
 #include "context.h"
 #include "decklink_capture.h"
 #include "defs.h"
+#include "disk_space_estimator.h"
 #include "flags.h"
 #include "video_encoder.h"
 #include "pbo_frame_allocator.h"
@@ -165,7 +166,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
        display_chain->set_dither_bits(0);  // Don't bother.
        display_chain->finalize();
 
-       video_encoder.reset(new VideoEncoder(resource_pool.get(), h264_encoder_surface, global_flags.va_display, WIDTH, HEIGHT, &httpd));
+       video_encoder.reset(new VideoEncoder(resource_pool.get(), h264_encoder_surface, global_flags.va_display, WIDTH, HEIGHT, &httpd, global_disk_space_estimator));
 
        // Start listening for clients only once VideoEncoder has written its header, if any.
        httpd.start(9095);
@@ -855,13 +856,27 @@ void Mixer::schedule_audio_resampling_tasks(unsigned dropped_frames, int num_sam
        // Resample the audio as needed, including from previously dropped frames.
        assert(num_cards > 0);
        for (unsigned frame_num = 0; frame_num < dropped_frames + 1; ++frame_num) {
+               const bool dropped_frame = (frame_num != dropped_frames);
                {
                        // Signal to the audio thread to process this frame.
+                       // Note that if the frame is a dropped frame, we signal that
+                       // we don't want to use this frame as base for adjusting
+                       // the resampler rate. The reason for this is that the timing
+                       // of these frames is often way too late; they typically don't
+                       // “arrive” before we synthesize them. Thus, we could end up
+                       // in a situation where we have inserted e.g. five audio frames
+                       // into the queue before we then start pulling five of them
+                       // back out. This makes ResamplingQueue overestimate the delay,
+                       // causing undue resampler changes. (We _do_ use the last,
+                       // non-dropped frame; perhaps we should just discard that as well,
+                       // since dropped frames are expected to be rare, and it might be
+                       // better to just wait until we have a slightly more normal situation).
                        unique_lock<mutex> lock(audio_mutex);
-                       audio_task_queue.push(AudioTask{pts_int, num_samples_per_frame});
+                       bool adjust_rate = !dropped_frame;
+                       audio_task_queue.push(AudioTask{pts_int, num_samples_per_frame, adjust_rate});
                        audio_task_queue_changed.notify_one();
                }
-               if (frame_num != dropped_frames) {
+               if (dropped_frame) {
                        // For dropped frames, increase the pts. Note that if the format changed
                        // in the meantime, we have no way of detecting that; we just have to
                        // assume the frame length is always the same.
@@ -961,11 +976,11 @@ void Mixer::audio_thread_func()
                        audio_task_queue.pop();
                }
 
-               process_audio_one_frame(task.pts_int, task.num_samples);
+               process_audio_one_frame(task.pts_int, task.num_samples, task.adjust_rate);
        }
 }
 
-void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
+void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples, bool adjust_rate)
 {
        vector<float> samples_card;
        vector<float> samples_out;
@@ -978,7 +993,13 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
                samples_card.resize(num_samples * 2);
                {
                        unique_lock<mutex> lock(cards[card_index].audio_mutex);
-                       cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], num_samples);
+                       ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy =
+                               adjust_rate ? ResamplingQueue::ADJUST_RATE : ResamplingQueue::DO_NOT_ADJUST_RATE;
+                       cards[card_index].resampling_queue->get_output_samples(
+                               double(frame_pts_int) / TIMEBASE,
+                               &samples_card[0],
+                               num_samples,
+                               rate_adjustment_policy);
                }
                if (card_index == selected_audio_card) {
                        samples_out = move(samples_card);