Small helpful comment.

[nageru] / mixer.h

diff --git a/mixer.h b/mixer.h
index 6725a7f46e242fe44bdba8a7c0e583b422b42102..6d0cbce1593b6da92c4f426668d948c982683921 100644 (file)
--- a/mixer.h
+++ b/mixer.h
@@ -10,6 +10,7 @@
 
 #include <movit/effect_chain.h>
 #include <movit/flat_input.h>
+#include <zita-resampler/resampler.h>
 #include <atomic>
 #include <condition_variable>
 #include <cstddef>
@@ -21,17 +22,19 @@
 #include <vector>
 
 #include "bmusb/bmusb.h"
+#include "alsa_output.h"
 #include "ebu_r128_proc.h"
 #include "h264encode.h"
 #include "httpd.h"
 #include "pbo_frame_allocator.h"
 #include "ref_counted_frame.h"
 #include "ref_counted_gl_sync.h"
-#include "resampler.h"
+#include "resampling_queue.h"
 #include "theme.h"
 #include "timebase.h"
 #include "stereocompressor.h"
 #include "filter.h"
+#include "input_state.h"
 
 class H264Encoder;
 class QSurface;
@@ -136,6 +139,36 @@ public:
                locut_cutoff_hz = cutoff_hz;
        }
 
+       float get_limiter_threshold_dbfs()
+       {
+               return limiter_threshold_dbfs;
+       }
+
+       float get_compressor_threshold_dbfs()
+       {
+               return compressor_threshold_dbfs;
+       }
+
+       void set_limiter_threshold_dbfs(float threshold_dbfs)
+       {
+               limiter_threshold_dbfs = threshold_dbfs;
+       }
+
+       void set_compressor_threshold_dbfs(float threshold_dbfs)
+       {
+               compressor_threshold_dbfs = threshold_dbfs;
+       }
+
+       void set_limiter_enabled(bool enabled)
+       {
+               limiter_enabled = enabled;
+       }
+
+       void set_compressor_enabled(bool enabled)
+       {
+               compressor_enabled = enabled;
+       }
+
        void reset_meters();
 
 private:
@@ -144,7 +177,8 @@ private:
                FrameAllocator::Frame audio_frame, size_t audio_offset, uint16_t audio_format);
        void place_rectangle(movit::Effect *resample_effect, movit::Effect *padding_effect, float x0, float y0, float x1, float y1);
        void thread_func();
-       void process_audio_one_frame();
+       void audio_thread_func();
+       void process_audio_one_frame(int64_t frame_pts_int, int num_samples);
        void subsample_chroma(GLuint src_tex, GLuint dst_dst);
        void release_display_frame(DisplayFrame *frame);
        double pts() { return double(pts_int) / TIMEBASE; }
@@ -176,17 +210,25 @@ private:
                bool new_data_ready = false;  // Whether new_frame contains anything.
                bool should_quit = false;
                RefCountedFrame new_frame;
+               int64_t new_frame_length;  // In TIMEBASE units.
+               bool new_frame_interlaced;
+               unsigned new_frame_field;  // Which field (0 or 1) of the frame to use. Always 0 for progressive.
                GLsync new_data_ready_fence;  // Whether new_frame is ready for rendering.
                std::condition_variable new_data_ready_changed;  // Set whenever new_data_ready is changed.
                unsigned dropped_frames = 0;  // Before new_frame.
 
+               // Accumulated errors in number of 1/TIMEBASE samples. If OUTPUT_FREQUENCY divided by
+               // frame rate is integer, will always stay zero.
+               unsigned fractional_samples = 0;
+
                std::mutex audio_mutex;
-               std::unique_ptr<Resampler> resampler;  // Under audio_mutex.
+               std::unique_ptr<ResamplingQueue> resampling_queue;  // Under audio_mutex.
                int last_timecode = -1;  // Unwrapped.
+               int64_t next_local_pts = 0;  // Beginning of next frame, in TIMEBASE units.
        };
        CaptureCard cards[MAX_CARDS];  // protected by <bmusb_mutex>
 
-       RefCountedFrame bmusb_current_rendering_frame[MAX_CARDS];
+       InputState input_state;
 
        class OutputChannel {
        public:
@@ -208,12 +250,14 @@ private:
        OutputChannel output_channel[NUM_OUTPUTS];
 
        std::thread mixer_thread;
-       bool should_quit = false;
+       std::thread audio_thread;
+       std::atomic<bool> should_quit{false};
 
        audio_level_callback_t audio_level_callback = nullptr;
-       Ebu_r128_proc r128;
+       std::mutex r128_mutex;
+       Ebu_r128_proc r128;  // Under r128_mutex.
 
-       // TODO: Implement oversampled peak detection.
+       Resampler peak_resampler;
        std::atomic<float> peak{0.0f};
 
        StereoFilter locut;  // Default cutoff 150 Hz, 24 dB/oct.
@@ -223,8 +267,24 @@ private:
        StereoCompressor level_compressor;
        float last_gain_staging_db = 0.0f;
 
+       static constexpr float ref_level_dbfs = -14.0f;
+
        StereoCompressor limiter;
+       std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f};   // 4 dB.
+       std::atomic<bool> limiter_enabled{true};
        StereoCompressor compressor;
+       std::atomic<float> compressor_threshold_dbfs{ref_level_dbfs - 12.0f};  // -12 dB.
+       std::atomic<bool> compressor_enabled{true};
+
+       std::unique_ptr<ALSAOutput> alsa;
+
+       struct AudioTask {
+               int64_t pts_int;
+               int num_samples;
+       };
+       std::mutex audio_mutex;
+       std::condition_variable audio_task_queue_changed;
+       std::queue<AudioTask> audio_task_queue;  // Under audio_mutex.
 };
 
 extern Mixer *global_mixer;