X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.h;h=f50eb58db5560f620b2fcdb1cc9097f4b6c5d1cb;hb=65d716be70e6295628dfa5bb0a72f3429b9696ba;hp=89a80dd52fabaccd8e81f1743c491f41b4d329ff;hpb=194872611b21bb070f4dfcf5beda6c36d25459ea;p=nageru diff --git a/mixer.h b/mixer.h index 89a80dd..f50eb58 100644 --- a/mixer.h +++ b/mixer.h @@ -34,6 +34,7 @@ #include "timebase.h" #include "stereocompressor.h" #include "filter.h" +#include "input_state.h" class H264Encoder; class QSurface; @@ -102,7 +103,7 @@ public: typedef std::function audio_level_callback_t; + float gain_staging_db)> audio_level_callback_t; void set_audio_level_callback(audio_level_callback_t callback) { audio_level_callback = callback; @@ -168,6 +169,24 @@ public: compressor_enabled = enabled; } + void set_gain_staging_db(float gain_db) + { + std::unique_lock lock(level_compressor_mutex); + level_compressor_enabled = false; + gain_staging_db = gain_db; + } + + void set_gain_staging_auto(bool enabled) + { + std::unique_lock lock(level_compressor_mutex); + level_compressor_enabled = enabled; + } + + void schedule_cut() + { + should_cut = true; + } + void reset_meters(); private: @@ -176,7 +195,8 @@ 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; } @@ -208,17 +228,25 @@ 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 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 - RefCountedFrame bmusb_current_rendering_frame[MAX_CARDS]; + InputState input_state; class OutputChannel { public: @@ -240,10 +268,13 @@ private: OutputChannel output_channel[NUM_OUTPUTS]; std::thread mixer_thread; - bool should_quit = false; + std::thread audio_thread; + std::atomic should_quit{false}; + std::atomic should_cut{false}; audio_level_callback_t audio_level_callback = nullptr; - Ebu_r128_proc r128; + std::mutex compressor_mutex; + Ebu_r128_proc r128; // Under compressor_mutex. Resampler peak_resampler; std::atomic peak{0.0f}; @@ -252,8 +283,10 @@ private: std::atomic locut_cutoff_hz; // First compressor; takes us up to about -12 dBFS. - StereoCompressor level_compressor; - float last_gain_staging_db = 0.0f; + std::mutex level_compressor_mutex; + StereoCompressor level_compressor; // Under compressor_mutex. Used to set/override gain_staging_db if . + float gain_staging_db = 0.0f; // Under compressor_mutex. + bool level_compressor_enabled = true; // Under compressor_mutex. static constexpr float ref_level_dbfs = -14.0f; @@ -265,6 +298,14 @@ private: std::atomic compressor_enabled{true}; std::unique_ptr alsa; + + struct AudioTask { + int64_t pts_int; + int num_samples; + }; + std::mutex audio_mutex; + std::condition_variable audio_task_queue_changed; + std::queue audio_task_queue; // Under audio_mutex. }; extern Mixer *global_mixer;