Fix another uninitialized sound data issue.

[nageru] / mixer.h

diff --git a/mixer.h b/mixer.h
index 64a74e2f13d855d0030a19645cdadf95956b9558..b9acfc4c607e2a9b880ef8dea5bc7867da9fa286 100644 (file)
--- a/mixer.h
+++ b/mixer.h
@@ -22,6 +22,7 @@
 #include <vector>
 
 #include "bmusb/bmusb.h"
+#include "alsa_output.h"
 #include "ebu_r128_proc.h"
 #include "h264encode.h"
 #include "httpd.h"
@@ -169,13 +170,24 @@ public:
 
        void reset_meters();
 
+       struct BufferedFrame {
+               RefCountedFrame frame;
+               unsigned field_number;
+       };
+
+       BufferedFrame get_buffered_frame(int card, int history_pos)
+       {
+               return buffered_frames[card][history_pos];
+       }
+
 private:
        void bm_frame(unsigned card_index, uint16_t timecode,
                FrameAllocator::Frame video_frame, size_t video_offset, uint16_t video_format,
                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; }
@@ -207,17 +219,30 @@ 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<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];
+       // For each card, the last three frames (or fields), with 0 being the
+       // most recent one. Note that we only need the actual history if we have
+       // interlaced output (for deinterlacing), so if we detect progressive input,
+       // we immediately clear out all history and all entries will point to the same
+       // frame.
+       BufferedFrame buffered_frames[MAX_CARDS][FRAME_HISTORY_LENGTH];
 
        class OutputChannel {
        public:
@@ -239,7 +264,8 @@ 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;
@@ -257,11 +283,21 @@ private:
        static constexpr float ref_level_dbfs = -14.0f;
 
        StereoCompressor limiter;
-       std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 0.0f};   // 0 dB.
+       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;