#include <epoxy/gl.h>
#undef Success
+#include <stdbool.h>
+#include <stdint.h>
+
#include <movit/effect_chain.h>
#include <movit/flat_input.h>
+#include <zita-resampler/resampler.h>
+#include <atomic>
+#include <condition_variable>
+#include <cstddef>
#include <functional>
+#include <memory>
+#include <mutex>
+#include <string>
+#include <thread>
+#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 "resampling_queue.h"
#include "theme.h"
-#include "resampler.h"
#include "timebase.h"
+#include "stereocompressor.h"
+#include "filter.h"
+#include "input_state.h"
+#include "correlation_measurer.h"
-#define NUM_CARDS 2
+class H264Encoder;
+class QSurface;
+namespace movit {
+class Effect;
+class EffectChain;
+class FlatInput;
+class ResourcePool;
+} // namespace movit
namespace movit {
class YCbCrInput;
class Mixer {
public:
// The surface format is used for offscreen destinations for OpenGL contexts we need.
- Mixer(const QSurfaceFormat &format);
+ Mixer(const QSurfaceFormat &format, unsigned num_cards);
~Mixer();
void start();
void quit();
enum Output {
OUTPUT_LIVE = 0,
OUTPUT_PREVIEW,
- OUTPUT_INPUT0,
- OUTPUT_INPUT1,
- OUTPUT_INPUT2,
- OUTPUT_INPUT3,
- NUM_OUTPUTS
+ OUTPUT_INPUT0, // 1, 2, 3, up to 15 follow numerically.
+ NUM_OUTPUTS = 18
};
struct DisplayFrame {
output_channel[output].set_frame_ready_callback(callback);
}
+ typedef std::function<void(float level_lufs, float peak_db,
+ float global_level_lufs, float range_low_lufs, float range_high_lufs,
+ float gain_staging_db, float final_makeup_gain_db,
+ float correlation)> audio_level_callback_t;
+ void set_audio_level_callback(audio_level_callback_t callback)
+ {
+ audio_level_callback = callback;
+ }
+
std::vector<std::string> get_transition_names()
{
return theme->get_transition_names(pts());
}
+ unsigned get_num_channels() const
+ {
+ return theme->get_num_channels();
+ }
+
+ std::string get_channel_name(unsigned channel) const
+ {
+ return theme->get_channel_name(channel);
+ }
+
+ int get_channel_signal(unsigned channel) const
+ {
+ return theme->get_channel_signal(channel);
+ }
+
+ int map_signal(unsigned channel)
+ {
+ return theme->map_signal(channel);
+ }
+
+ unsigned get_audio_source() const
+ {
+ return audio_source_channel;
+ }
+
+ void set_audio_source(unsigned channel)
+ {
+ audio_source_channel = channel;
+ }
+
+ void set_signal_mapping(int signal, int card)
+ {
+ return theme->set_signal_mapping(signal, card);
+ }
+
+ bool get_supports_set_wb(unsigned channel) const
+ {
+ return theme->get_supports_set_wb(channel);
+ }
+
+ void set_wb(unsigned channel, double r, double g, double b) const
+ {
+ theme->set_wb(channel, r, g, b);
+ }
+
+ void set_locut_cutoff(float cutoff_hz)
+ {
+ locut_cutoff_hz = cutoff_hz;
+ }
+
+ void set_locut_enabled(bool enabled)
+ {
+ locut_enabled = enabled;
+ }
+
+ 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 set_gain_staging_db(float gain_db)
+ {
+ std::unique_lock<std::mutex> lock(compressor_mutex);
+ level_compressor_enabled = false;
+ gain_staging_db = gain_db;
+ }
+
+ void set_gain_staging_auto(bool enabled)
+ {
+ std::unique_lock<std::mutex> lock(compressor_mutex);
+ level_compressor_enabled = enabled;
+ }
+
+ void set_final_makeup_gain_db(float gain_db)
+ {
+ std::unique_lock<std::mutex> lock(compressor_mutex);
+ final_makeup_gain_auto = false;
+ final_makeup_gain = pow(10.0f, gain_db / 20.0f);
+ }
+
+ void set_final_makeup_gain_auto(bool enabled)
+ {
+ std::unique_lock<std::mutex> lock(compressor_mutex);
+ final_makeup_gain_auto = enabled;
+ }
+
+ void schedule_cut()
+ {
+ should_cut = true;
+ }
+
+ void reset_meters();
+
+ unsigned get_num_cards() const { return num_cards; }
+
+ std::string get_card_description(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].usb->get_description();
+ }
+
private:
- void bm_frame(int card_index, uint16_t timecode,
+ 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 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; }
+ HTTPD httpd;
+ unsigned num_cards;
+
QSurface *mixer_surface, *h264_encoder_surface;
std::unique_ptr<movit::ResourcePool> resource_pool;
std::unique_ptr<Theme> theme;
+ std::atomic<unsigned> audio_source_channel{0};
std::unique_ptr<movit::EffectChain> display_chain;
GLuint cbcr_program_num; // Owned by <resource_pool>.
std::unique_ptr<H264Encoder> h264_encoder;
QSurface *surface;
QOpenGLContext *context;
- bool new_data_ready = false; // Whether new_frame and new_frame_audio contains anything.
+ 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::vector<float> new_frame_audio;
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[NUM_CARDS]; // protected by <bmusb_mutex>
+ CaptureCard cards[MAX_CARDS]; // protected by <bmusb_mutex>
- RefCountedFrame bmusb_current_rendering_frame[NUM_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};
+ std::atomic<bool> should_cut{false};
+
+ audio_level_callback_t audio_level_callback = nullptr;
+ std::mutex compressor_mutex;
+ Ebu_r128_proc r128; // Under compressor_mutex.
+ CorrelationMeasurer correlation; // Under compressor_mutex.
+
+ Resampler peak_resampler;
+ std::atomic<float> peak{0.0f};
+
+ StereoFilter locut; // Default cutoff 120 Hz, 24 dB/oct.
+ std::atomic<float> locut_cutoff_hz;
+ std::atomic<bool> locut_enabled{true};
+
+ // First compressor; takes us up to about -12 dBFS.
+ StereoCompressor level_compressor; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
+ 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; // Chosen so that we end up around 0 LU in practice.
+ static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
+
+ 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};
+
+ double final_makeup_gain = 1.0; // Under compressor_mutex. Read/write by the user. Note: Not in dB, we want the numeric precision so that we can change it slowly.
+ bool final_makeup_gain_auto = true; // Under compressor_mutex.
+
+ 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;
projects / nageru / blobdiff