#include <movit/effect_chain.h>
#include <movit/flat_input.h>
+#include <zita-resampler/resampler.h>
#include <atomic>
#include <condition_variable>
#include <cstddef>
#include <vector>
#include "bmusb/bmusb.h"
+#include "alsa_output.h"
#include "ebu_r128_proc.h"
#include "h264encode.h"
#include "httpd.h"
#include "timebase.h"
#include "stereocompressor.h"
#include "filter.h"
+#include "input_state.h"
class H264Encoder;
class QSurface;
typedef std::function<void(float level_lufs, float peak_db,
float global_level_lufs, float range_low_lufs, float range_high_lufs,
- float auto_gain_staging_db)> audio_level_callback_t;
+ float gain_staging_db, float final_makeup_gain_db)> audio_level_callback_t;
void set_audio_level_callback(audio_level_callback_t callback)
{
audio_level_callback = callback;
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 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();
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; }
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};
+ std::atomic<bool> 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.
- // TODO: Implement oversampled peak detection.
+ Resampler peak_resampler;
std::atomic<float> peak{0.0f};
StereoFilter locut; // Default cutoff 150 Hz, 24 dB/oct.
std::atomic<float> locut_cutoff_hz;
// First compressor; takes us up to about -12 dBFS.
- StereoCompressor level_compressor;
- float last_gain_staging_db = 0.0f;
+ 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