// The actual video mixer, running in its own separate background thread.
+#include <assert.h>
#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 <chrono>
#include <condition_variable>
#include <cstddef>
#include <functional>
+#include <map>
#include <memory>
#include <mutex>
+#include <queue>
#include <string>
#include <thread>
#include <vector>
+#include <movit/image_format.h>
+
+#include "audio_mixer.h"
#include "bmusb/bmusb.h"
-#include "alsa_output.h"
-#include "ebu_r128_proc.h"
-#include "h264encode.h"
+#include "defs.h"
#include "httpd.h"
+#include "input_state.h"
+#include "libusb.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 "timebase.h"
-#include "stereocompressor.h"
-#include "filter.h"
-#include "input_state.h"
-#include "correlation_measurer.h"
+#include "video_encoder.h"
+#include "ycbcr_interpretation.h"
-class H264Encoder;
+class ALSAOutput;
+class ChromaSubsampler;
+class DeckLinkOutput;
class QSurface;
+class QSurfaceFormat;
+class TimecodeRenderer;
+class v210Converter;
+
namespace movit {
class Effect;
class EffectChain;
-class FlatInput;
class ResourcePool;
+class YCbCrInput;
} // namespace movit
-namespace movit {
-class YCbCrInput;
-}
-class QOpenGLContext;
-class QSurfaceFormat;
+// A class to estimate the future jitter. Used in QueueLengthPolicy (see below).
+//
+// There are many ways to estimate jitter; I've tested a few ones (and also
+// some algorithms that don't explicitly model jitter) with different
+// parameters on some real-life data in experiments/queue_drop_policy.cpp.
+// This is one based on simple order statistics where I've added some margin in
+// the number of starvation events; I believe that about one every hour would
+// probably be acceptable, but this one typically goes lower than that, at the
+// cost of 2–3 ms extra latency. (If the queue is hard-limited to one frame, it's
+// possible to get ~10 ms further down, but this would mean framedrops every
+// second or so.) The general strategy is: Take the 99.9-percentile jitter over
+// last 5000 frames, multiply by two, and that's our worst-case jitter
+// estimate. The fact that we're not using the max value means that we could
+// actually even throw away very late frames immediately, which means we only
+// get one user-visible event instead of seeing something both when the frame
+// arrives late (duplicate frame) and then again when we drop.
+class JitterHistory {
+private:
+ static constexpr size_t history_length = 5000;
+ static constexpr double percentile = 0.999;
+ static constexpr double multiplier = 2.0;
+
+public:
+ void register_metrics(const std::vector<std::pair<std::string, std::string>> &labels);
+ void unregister_metrics(const std::vector<std::pair<std::string, std::string>> &labels);
+
+ void clear() {
+ history.clear();
+ orders.clear();
+ }
+ void frame_arrived(std::chrono::steady_clock::time_point now, int64_t frame_duration, size_t dropped_frames);
+ std::chrono::steady_clock::time_point get_expected_next_frame() const { return expected_timestamp; }
+ double estimate_max_jitter() const;
+
+private:
+ // A simple O(k) based algorithm for getting the k-th largest or
+ // smallest element from our window; we simply keep the multiset
+ // ordered (insertions and deletions are O(n) as always) and then
+ // iterate from one of the sides. If we had larger values of k,
+ // we could go for a more complicated setup with two sets or heaps
+ // (one increasing and one decreasing) that we keep balanced around
+ // the point, or it is possible to reimplement std::set with
+ // counts in each node. However, since k=5, we don't need this.
+ std::multiset<double> orders;
+ std::deque<std::multiset<double>::iterator> history;
+
+ std::chrono::steady_clock::time_point expected_timestamp = std::chrono::steady_clock::time_point::min();
+
+ // Metrics. There are no direct summaries for jitter, since we already have latency summaries.
+ std::atomic<int64_t> metric_input_underestimated_jitter_frames{0};
+ std::atomic<double> metric_input_estimated_max_jitter_seconds{0.0 / 0.0};
+};
+
+// 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 measure the
+// deviation from the expected arrival time for all cards, and use that for
+// continuous jitter estimation.
+//
+// We then drop everything from the queue that we're sure we won't need to
+// serve the output in the time before the next frame arrives. Typically,
+// this means the queue will contain 0 or 1 frames, although more is also
+// possible if the jitter is very high.
+class QueueLengthPolicy {
+public:
+ QueueLengthPolicy() {}
+ void reset(unsigned card_index) {
+ this->card_index = card_index;
+ }
+
+ void register_metrics(const std::vector<std::pair<std::string, std::string>> &labels);
+ void unregister_metrics(const std::vector<std::pair<std::string, std::string>> &labels);
+
+ // Call after picking out a frame, so 0 means starvation.
+ void update_policy(std::chrono::steady_clock::time_point now,
+ std::chrono::steady_clock::time_point expected_next_frame,
+ int64_t input_frame_duration,
+ int64_t master_frame_duration,
+ double max_input_card_jitter_seconds,
+ double max_master_card_jitter_seconds);
+ unsigned get_safe_queue_length() const { return safe_queue_length; }
+
+private:
+ unsigned card_index; // For debugging and metrics only.
+ unsigned safe_queue_length = 0; // Can never go below zero.
+
+ // Metrics.
+ std::atomic<int64_t> metric_input_queue_safe_length_frames{1};
+};
class Mixer {
public:
return output_channel[output].get_display_frame(frame);
}
+ // NOTE: Callbacks will be called with a mutex held, so you should probably
+ // not do real work in them.
typedef std::function<void()> new_frame_ready_callback_t;
- void set_frame_ready_callback(Output output, new_frame_ready_callback_t callback)
+ void add_frame_ready_callback(Output output, void *key, new_frame_ready_callback_t callback)
+ {
+ output_channel[output].add_frame_ready_callback(key, callback);
+ }
+
+ void remove_frame_ready_callback(Output output, void *key)
+ {
+ output_channel[output].remove_frame_ready_callback(key);
+ }
+
+ // TODO: Should this really be per-channel? Shouldn't it just be called for e.g. the live output?
+ typedef std::function<void(const std::vector<std::string> &)> transition_names_updated_callback_t;
+ void set_transition_names_updated_callback(Output output, transition_names_updated_callback_t callback)
+ {
+ output_channel[output].set_transition_names_updated_callback(callback);
+ }
+
+ typedef std::function<void(const std::string &)> name_updated_callback_t;
+ void set_name_updated_callback(Output output, name_updated_callback_t callback)
{
- output_channel[output].set_frame_ready_callback(callback);
+ output_channel[output].set_name_updated_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)
+ typedef std::function<void(const std::string &)> color_updated_callback_t;
+ void set_color_updated_callback(Output output, color_updated_callback_t callback)
{
- audio_level_callback = callback;
+ output_channel[output].set_color_updated_callback(callback);
}
std::vector<std::string> get_transition_names()
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);
return theme->map_signal(channel);
}
- unsigned get_audio_source() const
+ unsigned get_master_clock() const
{
- return audio_source_channel;
+ return master_clock_channel;
}
- void set_audio_source(unsigned channel)
+ void set_master_clock(unsigned channel)
{
- audio_source_channel = channel;
+ master_clock_channel = channel;
}
void set_signal_mapping(int signal, int card)
return theme->set_signal_mapping(signal, card);
}
+ YCbCrInterpretation get_input_ycbcr_interpretation(unsigned card_index) const;
+ void set_input_ycbcr_interpretation(unsigned card_index, const YCbCrInterpretation &interpretation);
+
bool get_supports_set_wb(unsigned channel) const
{
return theme->get_supports_set_wb(channel);
theme->set_wb(channel, r, g, b);
}
- void set_locut_cutoff(float cutoff_hz)
+ // Note: You can also get this through the global variable global_audio_mixer.
+ AudioMixer *get_audio_mixer() { return audio_mixer.get(); }
+ const AudioMixer *get_audio_mixer() const { return audio_mixer.get(); }
+
+ void schedule_cut()
{
- locut_cutoff_hz = cutoff_hz;
+ should_cut = true;
}
- void set_locut_enabled(bool enabled)
- {
- locut_enabled = enabled;
+ 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();
}
- float get_limiter_threshold_dbfs()
- {
- return limiter_threshold_dbfs;
+ // The difference between this and the previous function is that if a card
+ // is used as the current output, get_card_description() will return the
+ // fake card that's replacing it for input, whereas this function will return
+ // the card's actual name.
+ std::string get_output_card_description(unsigned card_index) const {
+ assert(card_can_be_used_as_output(card_index));
+ assert(card_index < num_cards);
+ if (cards[card_index].parked_capture) {
+ return cards[card_index].parked_capture->get_description();
+ } else {
+ return cards[card_index].capture->get_description();
+ }
}
- float get_compressor_threshold_dbfs()
- {
- return compressor_threshold_dbfs;
+ bool card_can_be_used_as_output(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].output != nullptr;
}
- void set_limiter_threshold_dbfs(float threshold_dbfs)
- {
- limiter_threshold_dbfs = threshold_dbfs;
+ bool card_is_ffmpeg(unsigned card_index) const {
+ assert(card_index < num_cards + num_video_inputs);
+ return cards[card_index].type == CardType::FFMPEG_INPUT;
}
- void set_compressor_threshold_dbfs(float threshold_dbfs)
- {
- compressor_threshold_dbfs = threshold_dbfs;
+ std::map<uint32_t, bmusb::VideoMode> get_available_video_modes(unsigned card_index) const {
+ assert(card_index < num_cards);
+ return cards[card_index].capture->get_available_video_modes();
}
- void set_limiter_enabled(bool enabled)
- {
- limiter_enabled = enabled;
+ 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_compressor_enabled(bool enabled)
- {
- compressor_enabled = enabled;
+ void set_video_mode(unsigned card_index, uint32_t mode) {
+ assert(card_index < num_cards);
+ cards[card_index].capture->set_video_mode(mode);
}
- 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 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();
}
- void set_gain_staging_auto(bool enabled)
- {
- std::unique_lock<std::mutex> lock(compressor_mutex);
- level_compressor_enabled = enabled;
+ 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_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_video_input(unsigned card_index, uint32_t input) {
+ assert(card_index < num_cards);
+ cards[card_index].capture->set_video_input(input);
}
- void set_final_makeup_gain_auto(bool enabled)
- {
- std::unique_lock<std::mutex> lock(compressor_mutex);
- final_makeup_gain_auto = enabled;
+ 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();
}
- void schedule_cut()
- {
- should_cut = true;
+ 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 reset_meters();
+ void set_audio_input(unsigned card_index, uint32_t input) {
+ assert(card_index < num_cards);
+ cards[card_index].capture->set_audio_input(input);
+ }
- unsigned get_num_cards() const { return num_cards; }
+ std::string get_ffmpeg_filename(unsigned card_index) const;
- std::string get_card_description(unsigned card_index) const {
- assert(card_index < num_cards);
- return cards[card_index].capture->get_description();
+ void set_ffmpeg_filename(unsigned card_index, const std::string &filename);
+
+ void change_x264_bitrate(unsigned rate_kbit) {
+ video_encoder->change_x264_bitrate(rate_kbit);
}
- 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();
+ int get_output_card_index() const { // -1 = no output, just stream.
+ return desired_output_card_index;
}
- 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_output_card(int card_index) { // -1 = no output, just stream.
+ desired_output_card_index = card_index;
}
- void set_video_mode(unsigned card_index, uint32_t mode) {
- assert(card_index < num_cards);
- cards[card_index].capture->set_video_mode(mode);
+ std::map<uint32_t, bmusb::VideoMode> get_available_output_video_modes() const;
+
+ uint32_t get_output_video_mode() const {
+ return desired_output_video_mode;
}
- void start_mode_scanning(unsigned card_index);
+ void set_output_video_mode(uint32_t mode) {
+ desired_output_video_mode = mode;
+ }
+
+ void set_display_timecode_in_stream(bool enable) {
+ display_timecode_in_stream = enable;
+ }
+
+ void set_display_timecode_on_stdout(bool enable) {
+ display_timecode_on_stdout = enable;
+ }
+
+ int64_t get_num_connected_clients() const {
+ return httpd.get_num_connected_clients();
+ }
+
+ std::vector<Theme::MenuEntry> get_theme_menu() { return theme->get_theme_menu(); }
+
+ void theme_menu_entry_clicked(int lua_ref) { return theme->theme_menu_entry_clicked(lua_ref); }
+
+ void set_theme_menu_callback(std::function<void()> callback)
+ {
+ theme->set_theme_menu_callback(callback);
+ }
+
+ void wait_for_next_frame();
private:
- void configure_card(unsigned card_index, const QSurfaceFormat &format, CaptureInterface *capture);
+ struct CaptureCard;
+
+ enum class CardType {
+ LIVE_CARD,
+ FAKE_CAPTURE,
+ FFMPEG_INPUT,
+ CEF_INPUT,
+ };
+ void configure_card(unsigned card_index, bmusb::CaptureInterface *capture, CardType card_type, DeckLinkOutput *output);
+ void set_output_card_internal(int card_index); // Should only be called from the mixer thread.
void bm_frame(unsigned card_index, uint16_t timecode,
- FrameAllocator::Frame video_frame, size_t video_offset, VideoFormat video_format,
- FrameAllocator::Frame audio_frame, size_t audio_offset, AudioFormat audio_format);
+ bmusb::FrameAllocator::Frame video_frame, size_t video_offset, bmusb::VideoFormat video_format,
+ bmusb::FrameAllocator::Frame audio_frame, size_t audio_offset, bmusb::AudioFormat audio_format);
+ void bm_hotplug_add(libusb_device *dev);
+ void bm_hotplug_remove(unsigned card_index);
void place_rectangle(movit::Effect *resample_effect, movit::Effect *padding_effect, float x0, float y0, float x1, float y1);
void thread_func();
+ void handle_hotplugged_cards();
+ void schedule_audio_resampling_tasks(unsigned dropped_frames, int num_samples_per_frame, int length_per_frame, bool is_preroll, std::chrono::steady_clock::time_point frame_timestamp);
+ std::string get_timecode_text() const;
+ void render_one_frame(int64_t duration);
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; }
+ void trim_queue(CaptureCard *card, size_t safe_queue_length);
+ std::pair<std::string, std::string> get_channels_json();
+ std::pair<std::string, std::string> get_channel_color_http(unsigned channel_idx);
HTTPD httpd;
- unsigned num_cards;
+ unsigned num_cards, num_video_inputs, num_html_inputs = 0;
- QSurface *mixer_surface, *h264_encoder_surface;
+ QSurface *mixer_surface, *h264_encoder_surface, *decklink_output_surface;
std::unique_ptr<movit::ResourcePool> resource_pool;
std::unique_ptr<Theme> theme;
std::atomic<unsigned> audio_source_channel{0};
+ std::atomic<int> master_clock_channel{0}; // Gets overridden by <output_card_index> if set.
+ int output_card_index = -1; // -1 for none.
+ uint32_t output_video_mode = -1;
+
+ // The mechanics of changing the output card and modes are so intricately connected
+ // with the work the mixer thread is doing. Thus, we don't change it directly,
+ // we just set this variable instead, which signals to the mixer thread that
+ // it should do the change before the next frame. This simplifies locking
+ // considerations immensely.
+ std::atomic<int> desired_output_card_index{-1};
+ std::atomic<uint32_t> desired_output_video_mode{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;
+ std::unique_ptr<ChromaSubsampler> chroma_subsampler;
+ std::unique_ptr<v210Converter> v210_converter;
+ std::unique_ptr<VideoEncoder> video_encoder;
+
+ std::unique_ptr<TimecodeRenderer> timecode_renderer;
+ std::atomic<bool> display_timecode_in_stream{false};
+ std::atomic<bool> display_timecode_on_stdout{false};
// Effects part of <display_chain>. Owned by <display_chain>.
- movit::FlatInput *display_input;
+ movit::YCbCrInput *display_input;
int64_t pts_int = 0; // In TIMEBASE units.
- std::mutex bmusb_mutex;
+ mutable std::mutex frame_num_mutex;
+ std::condition_variable frame_num_updated;
+ unsigned frame_num = 0; // Under <frame_num_mutex>.
+
+ // Accumulated errors in number of 1/TIMEBASE audio samples. If OUTPUT_FREQUENCY divided by
+ // frame rate is integer, will always stay zero.
+ unsigned fractional_samples = 0;
+
+ mutable std::mutex card_mutex;
+ bool has_bmusb_thread = false;
struct CaptureCard {
- CaptureInterface *capture;
+ std::unique_ptr<bmusb::CaptureInterface> capture;
+ bool is_fake_capture;
+ CardType type;
+ std::unique_ptr<DeckLinkOutput> output;
+
+ // CEF only delivers frames when it actually has a change.
+ // If we trim the queue for latency reasons, we could thus
+ // end up in a situation trimming a frame that was meant to
+ // be displayed for a long time, which is really suboptimal.
+ // Thus, if we drop the last frame we have, may_have_dropped_last_frame
+ // is set to true, and the next starvation event will trigger
+ // us requestin a CEF repaint.
+ bool is_cef_capture, may_have_dropped_last_frame = false;
+
+ // If this card is used for output (ie., output_card_index points to it),
+ // it cannot simultaneously be uesd for capture, so <capture> gets replaced
+ // by a FakeCapture. However, since reconstructing the real capture object
+ // with all its state can be annoying, it is not being deleted, just stopped
+ // and moved here.
+ std::unique_ptr<bmusb::CaptureInterface> parked_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 contains anything.
+ QSurface *surface = nullptr;
+
+ 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.
+ std::function<void()> upload_func; // Needs to be called to actually upload the texture to OpenGL.
+ unsigned dropped_frames = 0; // Number of dropped frames before this one.
+ std::chrono::steady_clock::time_point received_timestamp = std::chrono::steady_clock::time_point::min();
+ };
+ std::deque<NewFrame> new_frames;
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.
+ 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.
+
int last_timecode = -1; // Unwrapped.
- int64_t next_local_pts = 0; // Beginning of next frame, in TIMEBASE units.
+
+ JitterHistory jitter_history;
+
+ // Metrics.
+ std::vector<std::pair<std::string, std::string>> labels;
+ std::atomic<int64_t> metric_input_received_frames{0};
+ std::atomic<int64_t> metric_input_duped_frames{0};
+ std::atomic<int64_t> metric_input_dropped_frames_jitter{0};
+ std::atomic<int64_t> metric_input_dropped_frames_error{0};
+ std::atomic<int64_t> metric_input_resets{0};
+ std::atomic<int64_t> metric_input_queue_length_frames{0};
+
+ std::atomic<int64_t> metric_input_has_signal_bool{-1};
+ std::atomic<int64_t> metric_input_is_connected_bool{-1};
+ std::atomic<int64_t> metric_input_interlaced_bool{-1};
+ std::atomic<int64_t> metric_input_width_pixels{-1};
+ std::atomic<int64_t> metric_input_height_pixels{-1};
+ std::atomic<int64_t> metric_input_frame_rate_nom{-1};
+ std::atomic<int64_t> metric_input_frame_rate_den{-1};
+ std::atomic<int64_t> metric_input_sample_rate_hz{-1};
};
- CaptureCard cards[MAX_CARDS]; // protected by <bmusb_mutex>
+ JitterHistory output_jitter_history;
+ CaptureCard cards[MAX_VIDEO_CARDS]; // Protected by <card_mutex>.
+ YCbCrInterpretation ycbcr_interpretation[MAX_VIDEO_CARDS]; // Protected by <card_mutex>.
+ std::unique_ptr<AudioMixer> audio_mixer; // Same as global_audio_mixer (see audio_mixer.h).
+ bool input_card_is_master_clock(unsigned card_index, unsigned master_card_index) const;
+ struct OutputFrameInfo {
+ int dropped_frames; // Since last frame.
+ int num_samples; // Audio samples needed for this output frame.
+ int64_t frame_duration; // In TIMEBASE units.
+ bool is_preroll;
+ std::chrono::steady_clock::time_point frame_timestamp;
+ };
+ OutputFrameInfo get_one_frame_from_each_card(unsigned master_card_index, bool master_card_is_output, CaptureCard::NewFrame new_frames[MAX_VIDEO_CARDS], bool has_new_frame[MAX_VIDEO_CARDS]);
InputState input_state;
+ // Cards we have been noticed about being hotplugged, but haven't tried adding yet.
+ // Protected by its own mutex.
+ std::mutex hotplug_mutex;
+ std::vector<libusb_device *> hotplugged_cards;
+
class OutputChannel {
public:
~OutputChannel();
- void output_frame(DisplayFrame frame);
+ void output_frame(DisplayFrame &&frame);
bool get_display_frame(DisplayFrame *frame);
- void set_frame_ready_callback(new_frame_ready_callback_t callback);
+ void add_frame_ready_callback(void *key, new_frame_ready_callback_t callback);
+ void remove_frame_ready_callback(void *key);
+ void set_transition_names_updated_callback(transition_names_updated_callback_t callback);
+ void set_name_updated_callback(name_updated_callback_t callback);
+ void set_color_updated_callback(color_updated_callback_t callback);
private:
friend class Mixer;
+ unsigned channel;
Mixer *parent = nullptr; // Not owned.
std::mutex frame_mutex;
DisplayFrame current_frame, ready_frame; // protected by <frame_mutex>
bool has_current_frame = false, has_ready_frame = false; // protected by <frame_mutex>
- new_frame_ready_callback_t new_frame_ready_callback;
- bool has_new_frame_ready_callback = false;
+ std::map<void *, new_frame_ready_callback_t> new_frame_ready_callbacks; // protected by <frame_mutex>
+ transition_names_updated_callback_t transition_names_updated_callback;
+ name_updated_callback_t name_updated_callback;
+ color_updated_callback_t color_updated_callback;
+
+ std::vector<std::string> last_transition_names;
+ std::string last_name, last_color;
};
OutputChannel output_channel[NUM_OUTPUTS];
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;
+ bool adjust_rate;
+ std::chrono::steady_clock::time_point frame_timestamp;
};
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];
+ bool is_mode_scanning[MAX_VIDEO_CARDS]{ false };
+ std::vector<uint32_t> mode_scanlist[MAX_VIDEO_CARDS];
+ unsigned mode_scanlist_index[MAX_VIDEO_CARDS]{ 0 };
+ std::chrono::steady_clock::time_point last_mode_scan_change[MAX_VIDEO_CARDS];
};
extern Mixer *global_mixer;
projects / nageru / blobdiff