#include <thread>
#include <vector>
+#include <movit/image_format.h>
+
#include "audio_mixer.h"
#include "bmusb/bmusb.h"
#include "defs.h"
#include "theme.h"
#include "timebase.h"
#include "video_encoder.h"
+#include "ycbcr_interpretation.h"
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
+// 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
// 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.
+// 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.
//
-// 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.
+// 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;
- 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.
+ 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 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;
+ 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 {
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].set_frame_ready_callback(callback);
+ 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?
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);
}
// Note: You can also get this through the global variable global_audio_mixer.
- AudioMixer *get_audio_mixer() { return &audio_mixer; }
- const AudioMixer *get_audio_mixer() const { return &audio_mixer; }
+ AudioMixer *get_audio_mixer() { return audio_mixer.get(); }
+ const AudioMixer *get_audio_mixer() const { return audio_mixer.get(); }
void schedule_cut()
{
return cards[card_index].output != nullptr;
}
+ bool card_is_ffmpeg(unsigned card_index) const {
+ assert(card_index < num_cards + num_video_inputs);
+ return cards[card_index].type == CardType::FFMPEG_INPUT;
+ }
+
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();
cards[card_index].capture->set_audio_input(input);
}
+ std::string get_ffmpeg_filename(unsigned card_index) const;
+
+ 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);
}
desired_output_card_index = card_index;
}
+ 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 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, bmusb::CaptureInterface *capture, bool is_fake_capture, DeckLinkOutput *output);
+ 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,
bmusb::FrameAllocator::Frame video_frame, size_t video_offset, bmusb::VideoFormat video_format,
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);
+ 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 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, *decklink_output_surface;
std::unique_ptr<movit::ResourcePool> resource_pool;
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 are so intricately connected
+ // 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;
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.
+ 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;
- std::mutex card_mutex;
+ mutable std::mutex card_mutex;
bool has_bmusb_thread = false;
struct CaptureCard {
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
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::queue<NewFrame> new_frames;
+ std::deque<NewFrame> new_frames;
bool should_quit = false;
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.
+
+ 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};
};
+ JitterHistory output_jitter_history;
CaptureCard cards[MAX_VIDEO_CARDS]; // Protected by <card_mutex>.
- AudioMixer audio_mixer; // Same as global_audio_mixer (see audio_mixer.h).
+ 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]);
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);
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;
+ 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;
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;
};
extern Mixer *global_mixer;
-extern bool uses_mlock;
#endif // !defined(_MIXER_H)
projects / nageru / blobdiff