#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 <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 "resampler.h"
+#include "resampling_queue.h"
#include "theme.h"
#include "timebase.h"
+#include "stereocompressor.h"
+#include "filter.h"
+#include "input_state.h"
+#include "correlation_measurer.h"
class H264Encoder;
class QSurface;
class QOpenGLContext;
class QSurfaceFormat;
+// For any card that's not the master (where we pick out the frames as they
+// come, as fast as we can process), there's going to be a queue. The question
+// is when we should drop frames from that queue (apart from the obvious
+// dropping if the 16-frame queue should become full), especially given that
+// the frame rate could be lower or higher than the master (either subtly or
+// dramatically). We have two (conflicting) demands:
+//
+// 1. We want to avoid starving the queue.
+// 2. We don't want to add more delay than is needed.
+//
+// Our general strategy is to drop as many frames as we can (helping for #2)
+// that we think is safe for #1 given jitter. To this end, we set a lower floor N,
+// where we assume that if we have N frames in the queue, we're always safe from
+// starvation. (Typically, N will be 0 or 1. It starts off at 0.) If we have
+// more than N frames in the queue after reading out the one we need, we head-drop
+// them to reduce the queue.
+//
+// N is reduced as follows: If the queue has had at least one spare frame for
+// at least 50 (master) frames (ie., it's been too conservative for a second),
+// we reduce N by 1 and reset the timers. TODO: Only do this if N ever actually
+// touched the limit.
+//
+// Whenever the queue is starved (we needed a frame but there was none),
+// and we've been at N since the last starvation, N was obviously too low,
+// so we increment it. We will never set N above 5, though.
+class QueueLengthPolicy {
+public:
+ QueueLengthPolicy() {}
+ void reset(unsigned card_index) {
+ this->card_index = card_index;
+ safe_queue_length = 0;
+ frames_with_at_least_one = 0;
+ been_at_safe_point_since_last_starvation = false;
+ }
+
+ void update_policy(int queue_length); // Give in -1 for starvation.
+ unsigned get_safe_queue_length() const { return safe_queue_length; }
+
+private:
+ unsigned card_index; // For debugging only.
+ unsigned safe_queue_length = 0; // Called N in the comments.
+ unsigned frames_with_at_least_one = 0;
+ bool been_at_safe_point_since_last_starvation = false;
+};
+
class Mixer {
public:
// The surface format is used for offscreen destinations for OpenGL contexts we need.
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, float, float, float, float)> audio_level_callback_t;
+ 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;
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);
+ }
+
+ std::string get_channel_color(unsigned channel) const
+ {
+ return theme->get_channel_color(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;
+ }
+
+ unsigned get_master_clock() const
+ {
+ return master_clock_channel;
+ }
+
+ void set_master_clock(unsigned channel)
+ {
+ master_clock_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].capture->get_description();
+ }
+
+ std::map<uint32_t, VideoMode> get_available_video_modes(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].capture->get_available_video_modes();
+ }
+
+ uint32_t get_current_video_mode(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].capture->get_current_video_mode();
+ }
+
+ void set_video_mode(unsigned card_index, uint32_t mode) {
+ assert(card_index < num_cards);
+ cards[card_index].capture->set_video_mode(mode);
+ }
+
+ void start_mode_scanning(unsigned card_index);
+
+ std::map<uint32_t, std::string> get_available_video_inputs(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].capture->get_available_video_inputs();
+ }
+
+ uint32_t get_current_video_input(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].capture->get_current_video_input();
+ }
+
+ void set_video_input(unsigned card_index, uint32_t input) {
+ assert(card_index < num_cards);
+ cards[card_index].capture->set_video_input(input);
+ }
+
+ std::map<uint32_t, std::string> get_available_audio_inputs(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].capture->get_available_audio_inputs();
+ }
+
+ uint32_t get_current_audio_input(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].capture->get_current_audio_input();
+ }
+
+ void set_audio_input(unsigned card_index, uint32_t input) {
+ assert(card_index < num_cards);
+ cards[card_index].capture->set_audio_input(input);
+ }
+
private:
+ void configure_card(unsigned card_index, const QSurfaceFormat &format, CaptureInterface *capture);
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);
+ FrameAllocator::Frame video_frame, size_t video_offset, VideoFormat video_format,
+ FrameAllocator::Frame audio_frame, size_t audio_offset, AudioFormat 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 schedule_audio_resampling_tasks(unsigned dropped_frames, int num_samples_per_frame, int length_per_frame);
+ void render_one_frame();
+ void send_audio_level_callback();
+ 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; }
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::atomic<unsigned> master_clock_channel{0};
std::unique_ptr<movit::EffectChain> display_chain;
GLuint cbcr_program_num; // Owned by <resource_pool>.
+ GLuint cbcr_vbo; // Holds position and texcoord data.
+ GLuint cbcr_position_attribute_index, cbcr_texcoord_attribute_index;
std::unique_ptr<H264Encoder> h264_encoder;
// Effects part of <display_chain>. Owned by <display_chain>.
std::mutex bmusb_mutex;
struct CaptureCard {
- BMUSBCapture *usb;
+ CaptureInterface *capture;
std::unique_ptr<PBOFrameAllocator> frame_allocator;
// Stuff for the OpenGL context (for texture uploading).
QSurface *surface;
QOpenGLContext *context;
- bool new_data_ready = false; // Whether new_frame and new_frame_audio contains anything.
+ struct NewFrame {
+ RefCountedFrame frame;
+ int64_t length; // In TIMEBASE units.
+ bool interlaced;
+ unsigned field; // Which field (0 or 1) of the frame to use. Always 0 for progressive.
+ RefCountedGLsync ready_fence; // Whether frame is ready for rendering.
+ unsigned dropped_frames = 0; // Number of dropped frames before this one.
+ };
+ std::queue<NewFrame> new_frames;
bool should_quit = false;
- RefCountedFrame new_frame;
- 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.
+ std::condition_variable new_frames_changed; // Set whenever new_frames (or should_quit) is changed.
+
+ QueueLengthPolicy queue_length_policy; // Refers to the "new_frames" queue.
+
+ // 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[MAX_CARDS]; // protected by <bmusb_mutex>
+ void get_one_frame_from_each_card(unsigned master_card_index, CaptureCard::NewFrame new_frames[MAX_CARDS], bool has_new_frame[MAX_CARDS], int num_samples[MAX_CARDS]);
- 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.
+ CorrelationMeasurer correlation; // Under compressor_mutex.
+
+ Resampler peak_resampler;
+ std::atomic<float> peak{0.0f};
- // TODO: Implement oversampled peak detection.
- 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.
+
+ // For mode scanning.
+ bool is_mode_scanning[MAX_CARDS]{ false };
+ std::vector<uint32_t> mode_scanlist[MAX_CARDS];
+ unsigned mode_scanlist_index[MAX_CARDS]{ 0 };
+ timespec last_mode_scan_change[MAX_CARDS];
};
extern Mixer *global_mixer;