#include "timebase.h"
#include "stereocompressor.h"
#include "filter.h"
+#include "input_state.h"
class H264Encoder;
class QSurface;
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; }
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];
+ InputState input_state;
class OutputChannel {
public:
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.
Resampler peak_resampler;
std::atomic<float> peak{0.0f};
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;