Make a symbolic constant for master_card_index, instead of writing 0 everywhere.

diff --git a/mixer.cpp b/mixer.cpp
index 8ceee21a28e1dabd2bfd9ff5b3ebc784b3ad2b9d..104a6c03a22278b9b9ddfde7e8eb7edc4f9ad96e 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -581,12 +581,14 @@ void Mixer::thread_func()
                bool has_new_frame[MAX_CARDS] = { false };
                int num_samples[MAX_CARDS] = { 0 };
 
+               // The first card is the master timer, so wait for it to have a new frame.
+               // TODO: Make configurable, and with a timeout.
+               unsigned master_card_index = 0;
+
                {
                        unique_lock<mutex> lock(bmusb_mutex);
 
-                       // The first card is the master timer, so wait for it to have a new frame.
-                       // TODO: Make configurable, and with a timeout.
-                       cards[0].new_frames_changed.wait(lock, [this]{ return !cards[0].new_frames.empty(); });
+                       cards[master_card_index].new_frames_changed.wait(lock, [this, master_card_index]{ return !cards[master_card_index].new_frames.empty(); });
 
                        for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
                                CaptureCard *card = &cards[card_index];
@@ -607,19 +609,19 @@ void Mixer::thread_func()
 
                // Resample the audio as needed, including from previously dropped frames.
                assert(num_cards > 0);
-               for (unsigned frame_num = 0; frame_num < new_frames[0].dropped_frames + 1; ++frame_num) {
+               for (unsigned frame_num = 0; frame_num < new_frames[master_card_index].dropped_frames + 1; ++frame_num) {
                        {
                                // Signal to the audio thread to process this frame.
                                unique_lock<mutex> lock(audio_mutex);
-                               audio_task_queue.push(AudioTask{pts_int, num_samples[0]});
+                               audio_task_queue.push(AudioTask{pts_int, num_samples[master_card_index]});
                                audio_task_queue_changed.notify_one();
                        }
-                       if (frame_num != new_frames[0].dropped_frames) {
+                       if (frame_num != new_frames[master_card_index].dropped_frames) {
                                // 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.
                                ++stats_dropped_frames;
-                               pts_int += new_frames[0].length;
+                               pts_int += new_frames[master_card_index].length;
                        }
                }
 
@@ -636,8 +638,8 @@ void Mixer::thread_func()
                                             correlation.get_correlation());
                }
 
-               for (unsigned card_index = 1; card_index < num_cards; ++card_index) {
-                       if (!has_new_frame[card_index]) {
+               for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+                       if (card_index == master_card_index || !has_new_frame[card_index]) {
                                continue;
                        }
                        if (new_frames[card_index].frame->len == 0) {
@@ -651,9 +653,9 @@ void Mixer::thread_func()
 
                // If the first card is reporting a corrupted or otherwise dropped frame,
                // just increase the pts (skipping over this frame) and don't try to compute anything new.
-               if (new_frames[0].frame->len == 0) {
+               if (new_frames[master_card_index].frame->len == 0) {
                        ++stats_dropped_frames;
-                       pts_int += new_frames[0].length;
+                       pts_int += new_frames[master_card_index].length;
                        continue;
                }
 
@@ -710,7 +712,7 @@ void Mixer::thread_func()
                const int64_t av_delay = TIMEBASE / 10;  // Corresponds to the fixed delay in resampling_queue.h. TODO: Make less hard-coded.
                h264_encoder->end_frame(fence, pts_int + av_delay, theme_main_chain.input_frames);
                ++frame;
-               pts_int += new_frames[0].length;
+               pts_int += new_frames[master_card_index].length;
 
                // The live frame just shows the RGBA texture we just rendered.
                // It owns rgba_tex now.