rx-player 4.0.0-dev.2023102700 → 4.0.0-dev.2023110700

package/dist/es2017/default_config.d.ts

@@ -0,0 +1,1081 @@
+/**
+ * Configuration file for the whole player.
+ * Feel free to tweak those values if you know what you're doing.
+ *
+ * Please not that you will need to re-build the whole project to take these
+ * modifications into account.
+ *
+ * @type {Object}
+ */
+declare const DEFAULT_CONFIG: {
+    /**
+     * Volume set on unMute if the volume is set to 0 and either:
+     *   - mute has never been called before
+     *   - mute has last been called while the volume was already set to 0 (either
+     *     via setVolume, or a previous mute call)
+     * @type {Number}
+     */
+    DEFAULT_UNMUTED_VOLUME: number;
+    /**
+     * Default time interval after which a request will timeout, in ms.
+     * @type {Number}
+     */
+    DEFAULT_REQUEST_TIMEOUT: number;
+    /**
+     * Can be either:
+     *   - "native": Subtitles are all displayed in a <track> element
+     *   - "html": Subtitles are all displayed in a <div> separated from the video
+     *     element. Can be useful to display richer TTML subtitles, for example.
+     * @type {Object|null}
+     */
+    DEFAULT_TEXT_TRACK_MODE: "native" | "html";
+    /**
+     * Default behavior for the `enableFastSwitching` loadVideo options.
+     *
+     * Fast-switching allows to provide quicker transitions from lower quality
+     * segments to higher quality segments but might be badly supported on some
+     * devices.
+     * When enabled, the RxPlayer might replace segments of a lower-quality
+     * (with a lower bitrate) with segments of a higher quality (with a higher
+     * bitrate). This allows to have a fast transition when network conditions
+     * improve.
+     * When disabled, segments of a lower-quality will not be replaced.
+     */
+    DEFAULT_ENABLE_FAST_SWITCHING: boolean;
+    /**
+     * In some cases after switching the current track or bitrate, the RxPlayer
+     * could be led to go into the `"RELOADING"` state, which corresponds to
+     * visually a black screen (with nothing audible) before restarting playback.
+     *
+     * We could want to seek back some milliseconds when doing that.
+     * For example, when switching the current audio track, it might make sense
+     * to restart some time before, so the beginning of the sentence can be heard
+     * again in the new language.
+     *
+     * This config property allows to set the relative position the RxPlayer will
+     * seek to after reloading, in seconds.
+     *
+     * For example: a value of `-0.7` means that will seek back 700 milliseconds
+     * when reloading due to a track or bitrate switch with necessitated a
+     * reloading.
+     */
+    DELTA_POSITION_AFTER_RELOAD: {
+        /** Relative position when switching the bitrate */
+        bitrateSwitch: number;
+        /**
+         * Relative position when switching the track.
+         *
+         * From tests, I noticed that seeking back was only really "pleasant" when
+         * switching the audio track.
+         *
+         * E.g. switching the video track often means changing the camera angle or
+         * even totally changing what is being seen and rely much less on temporal
+         * context than when an audio track is switched.
+         * As such, I decided to only set a sensible seek-back behavior when
+         * switching the audio track, and only a minimal one (to still ensure
+         * nothing was missed) for video.
+         *
+         * "Other" mainly concern text track, where seeking back could even be
+         * annoying, so that behavior has been disabled in that case.
+         */
+        trackSwitch: {
+            audio: number;
+            video: number;
+            other: number;
+        };
+    };
+    /**
+     * Behavior of the RxPlayer when encountering a whole other codec on a already
+     * existing audio or video SourceBuffer.
+     *
+     * Can be either:
+     *
+     *    - "continue": Segments linked to the new codec will continue to be
+     *      pushed to that same SourceBuffer. The RxPlayer will still try to call
+     *      the `changeType` API on the SourceBuffer before pushing those
+     *      segments but continue even if this call failed.
+     *
+     *    - "reload": Every time a new incompatible codec is encountered on a
+     *      given SourceBuffer, we will reload the MediaSource.
+     */
+    DEFAULT_CODEC_SWITCHING_BEHAVIOR: "reload" | "continue";
+    /**
+     * If set to true, video through loadVideo will auto play by default
+     * @type {Boolean}
+     */
+    DEFAULT_AUTO_PLAY: boolean;
+    /**
+     * Default buffer goal in seconds.
+     * Once enough content has been downloaded to fill the buffer up to
+     * ``current position + DEFAULT_WANTED_BUFFER_AHEAD", we will stop downloading
+     * content.
+     * @type {Number}
+     */
+    DEFAULT_WANTED_BUFFER_AHEAD: number;
+    /**
+     * Default max buffer size ahead of the current position in seconds.
+     * The buffer _after_ this limit will be garbage collected.
+     * Set to Infinity for no limit.
+     * @type {Number}
+     */
+    DEFAULT_MAX_BUFFER_AHEAD: number;
+    /**
+     * Default max buffer size ahead of the current position in seconds.
+     * The buffer _before_ this limit will be garbage collected.
+     * Set to Infinity for no limit.
+     * @type {Number}
+     */
+    DEFAULT_MAX_BUFFER_BEHIND: number;
+    /**
+     * Default video buffer memory limit in kilobytes.
+     * Once enough video content has been downloaded to fill the buffer up to
+     * DEFAULT_MAX_VIDEO_BUFFER_SIZE , we will stop downloading
+     * content.
+     * @type {Number}
+     */
+    DEFAULT_MAX_VIDEO_BUFFER_SIZE: number;
+    /**
+     * Maximum possible buffer ahead for each type of buffer, to avoid too much
+     * memory usage when playing for a long time.
+     * Equal to Infinity if not defined here.
+     * @type {Object}
+     */
+    MAXIMUM_MAX_BUFFER_AHEAD: Partial<Record<"audio" | "video" | "text", number>>;
+    /**
+     * Maximum possible buffer behind for each type of buffer, to avoid too much
+     * memory usage when playing for a long time.
+     * Equal to Infinity if not defined here.
+     * @type {Object}
+     */
+    MAXIMUM_MAX_BUFFER_BEHIND: Partial<Record<"audio" | "video" | "text", number>>;
+    /**
+     * Default bitrate ceils initially set as the first content begins.
+     *
+     * If no track is found with a bitrate inferior or equal to the
+     * bitrate there, the one with the lowest bitrate will be taken instead.
+     *
+     * Set to 0 for the lowest bitrate, Infinity for the highest.
+     *
+     * These values are only useful for the first content played, as consecutive
+     * play will always take the last set one.
+     * @type {Object}
+     */
+    DEFAULT_BASE_BANDWIDTH: number;
+    /**
+     * Delay after which, if the page is hidden, the user is considered inactive
+     * on the current video.
+     *
+     * Allow to enforce specific optimizations when the page is not shown.
+     * @see DEFAULT_THROTTLE_WHEN_HIDDEN
+     * @type {Number}
+     */
+    INACTIVITY_DELAY: number;
+    /**
+     * If true, if the video is considered in a "hidden" state for a delay specified by
+     * the INACTIVITY DELAY config property, we throttle automatically to the video
+     * representation with the lowest bitrate.
+     * @type {Boolean}
+     */
+    DEFAULT_THROTTLE_VIDEO_BITRATE_WHEN_HIDDEN: boolean;
+    /**
+     * Default video resolution limit behavior.
+     *
+     * This option allows for example to throttle the video resolution so it
+     * does not exceed the screen resolution.
+     *
+     * Here set to "none" by default to disable throttling.
+     * @type {Boolean}
+     */
+    DEFAULT_VIDEO_RESOLUTION_LIMIT: "none";
+    /**
+     * Default initial live gap considered if no presentation delay has been
+     * suggested, in seconds.
+     * @type {Number}
+     */
+    DEFAULT_LIVE_GAP: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Maximum time, in seconds, the player should automatically skip when stalled
+     * because of a current hole in the buffer.
+     * Bear in mind that this might seek over not-yet-downloaded/pushed segments.
+     * @type {Number}
+     */
+    BUFFER_DISCONTINUITY_THRESHOLD: number;
+    /**
+     * When encountering small discontinuities, the RxPlayer may want, in specific
+     * conditions, ignore those and let the browser seek over them iself (this
+     * allows for example to avoid conflicts when both the browser and the
+     * RxPlayer want to seek at a different position, sometimes leading to a
+     * seeking loop).
+     * In this case, we however still want to seek it ourselves if the browser
+     * doesn't take the opportunity soon enough.
+     *
+     * This value specifies a delay after which a discontinuity ignored by the
+     * RxPlayer is finally considered.
+     * We want to maintain high enough to be sure the browser will not seek yet
+     * small enough so this (arguably rare) situation won't lead to too much
+     * waiting time.
+     */
+    FORCE_DISCONTINUITY_SEEK_DELAY: number;
+    /**
+     * Ratio used to know if an already loaded segment should be re-buffered.
+     * We re-load the given segment if the current one times that ratio is
+     * inferior to the new one.
+     * @type {Number}
+     */
+    BITRATE_REBUFFERING_RATIO: number;
+    /**
+     * Those are used when a "QuotaExceededError" error is received after
+     * appending a new segment in the SourceBuffer.
+     *
+     * This error can arise when the browser's buffer is considered full.
+     * In this case, the player goes into manual garbage collection (GC) mode.
+     * @type {Object}
+     */
+    BUFFER_GC_GAPS: {
+        /**
+         * _Low_ gap (from current position) from which the buffer will be _garbage
+         * collected_ (read removed from the buffer) when a QuotaExceededError is
+         * received.
+         * In seconds.
+         * @type {Number}
+         */
+        CALM: number;
+        /**
+         * _High_ gap (from current position) from which the buffer will be _garbage
+         * collected_ (read removed from the buffer) when a QuotaExceededError is
+         * received, if the low one does not clean up any buffer.
+         * In seconds.
+         * @type {Number}
+         */
+        BEEFY: number;
+    };
+    /**
+     * The default number of times a manifest request will be re-performed
+     * when loaded/refreshed if the request finishes on an error which
+     * justify an retry.
+     *
+     * Note that some errors do not use this counter:
+     *   - if the error is not due to the xhr, no retry will be peformed
+     *   - if the error is an HTTP error code, but not a 500-smthg or a 404, no
+     *     retry will be performed.
+     * @type Number
+     */
+    DEFAULT_MAX_MANIFEST_REQUEST_RETRY: number;
+    /**
+     * Default delay, in seconds, during which a CDN will be "downgraded".
+     *
+     * For example in case of media content being available on multiple CDNs, the
+     * RxPlayer may decide that a CDN is less reliable (for example, it returned a
+     * server error) and should thus be avoided, at least for some time
+     *
+     * This value is the amount of time this CDN will be "less considered" than the
+     * alternatives.
+     */
+    DEFAULT_CDN_DOWNGRADE_TIME: number;
+    /**
+     * The default number of times a segment request will be re-performed when
+     * on error which justify a retry.
+     *
+     * Note that some errors do not use this counter:
+     *   - if the error is not due to the xhr, no retry will be peformed
+     *   - if the error is an HTTP error code, but not a 500-smthg or a 404, no
+     *     retry will be performed.
+     * @type Number
+     */
+    DEFAULT_MAX_REQUESTS_RETRY_ON_ERROR: number;
+    /**
+     * Initial backoff delay when a segment / manifest download fails, in
+     * milliseconds.
+     *
+     * This delay will then grow exponentally by power of twos (200, 400, 800
+     * etc.)
+     *
+     * Please note that this delay is not exact, as it will be fuzzed.
+     * @type {Number}
+     */
+    INITIAL_BACKOFF_DELAY_BASE: {
+        REGULAR: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Maximum backoff delay when a segment / manifest download fails, in
+     * milliseconds.
+     *
+     * Please note that this delay is not exact, as it will be fuzzed.
+     * @type {Number}
+     */
+    MAX_BACKOFF_DELAY_BASE: {
+        REGULAR: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Minimum interval at which playback information samples will be taken. This
+     * variable is for the "regular" mediasource strategy (that is, not for the
+     * directfile API.
+     *
+     * At each of these interval, various different modules in the RxPlayer will
+     * run based on the information communicated.
+     *
+     * Keep in mind this is the minimum interval. This logic will also be
+     * triggered when various events of the media element are received.
+     * @type {Number}
+     */
+    SAMPLING_INTERVAL_MEDIASOURCE: number;
+    /**
+     * Same than SAMPLING_INTERVAL_MEDIASOURCE but for lowLatency mode.
+     * @type {Number}
+     */
+    SAMPLING_INTERVAL_LOW_LATENCY: number;
+    /**
+     * Same than SAMPLING_INTERVAL_MEDIASOURCE but for the directfile API.
+     * @type {Number}
+     */
+    SAMPLING_INTERVAL_NO_MEDIASOURCE: number;
+    /**
+     * Amount of buffer to have ahead of the current position before we may
+     * consider buffer-based adaptive estimates, in seconds.
+     *
+     * For example setting it to `10` means that we need to have ten seconds of
+     * buffer ahead of the current position before relying on buffer-based
+     * adaptive estimates.
+     *
+     * To avoid getting in-and-out of the buffer-based logic all the time, it
+     * should be set higher than `ABR_EXIT_BUFFER_BASED_ALGO`.
+     */
+    ABR_ENTER_BUFFER_BASED_ALGO: number;
+    /**
+     * Below this amount of buffer ahead of the current position, in seconds, we
+     * will stop using buffer-based estimate in our adaptive logic to select a
+     * quality.
+     *
+     * For example setting it to `5` means that if we have less than 5 seconds of
+     * buffer ahead of the current position, we should stop relying on
+     * buffer-based estimates to choose a quality.
+     *
+     * To avoid getting in-and-out of the buffer-based logic all the time, it
+     * should be set lower than `ABR_ENTER_BUFFER_BASED_ALGO`.
+     */
+    ABR_EXIT_BUFFER_BASED_ALGO: number;
+    /**
+     * Minimum number of bytes sampled before we trust the estimate.
+     * If we have not sampled much data, our estimate may not be accurate
+     * enough to trust.
+     * If the total of bytes sampled is less than this value, we use a
+     * default estimate.
+     * This specific value is based on experimentations.
+     * @type {Number}
+     */
+    ABR_MINIMUM_TOTAL_BYTES: number;
+    /**
+     * Minimum number of bytes, under which samples are discarded.
+     * Our models do not include latency information, so connection startup time
+     * (time to first byte) is considered part of the download time.
+     * Because of this, we should ignore very small downloads which would cause
+     * our estimate to be too low.
+     * This specific value is based on experimentation.
+     * @type {Number}
+     */
+    ABR_MINIMUM_CHUNK_SIZE: number;
+    /**
+     * Factor with which is multiplied the bandwidth estimate when the ABR is in
+     * starvation mode.
+     * @type {Object}
+     */
+    ABR_STARVATION_FACTOR: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Factor with which is multiplied the bandwidth estimate when the ABR is not
+     * in starvation mode.
+     * @type {Object}
+     */
+    ABR_REGULAR_FACTOR: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * If a media buffer has less than ABR_STARVATION_GAP in seconds ahead of the
+     * current position in its buffer, the adaptive logic will go into starvation
+     * mode.
+     *
+     * It gets out of starvation mode when the OUT_OF_STARVATION_GAP value is
+     * reached.
+     *
+     * Under this starvation mode:
+     *
+     *   - the bandwidth considered will be a little lower than the one estimated
+     *
+     *   - the time the next important request take will be checked
+     *     multiple times to detect when/if it takes too much time.
+     *     If the request is considered too long, the bitrate will be hastily
+     *     re-calculated from this single request.
+     *
+     * @type {Object}
+     */
+    ABR_STARVATION_GAP: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    OUT_OF_STARVATION_GAP: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * This is a security to avoid going into starvation mode when the content is
+     * ending (@see ABR_STARVATION_GAP).
+     * Basically, we subtract that value from the global duration of the content
+     * and we never enter "starvation mode" if the currently available buffer
+     * (which equals to the current position + the available buffer ahead of it)
+     * is equal or higher than this value.
+     * @type {Number}
+     */
+    ABR_STARVATION_DURATION_DELTA: number;
+    /**
+     * Half-life, in seconds for a fastly-evolving exponential weighted moving
+     * average.
+     * The lower it is, the faster the ABR logic will react to the bandwidth
+     * falling quickly.
+     * Should be kept to a lower number than ABR_SLOW_EMA for coherency reasons.
+     * @type {Number}
+     */
+    ABR_FAST_EMA: number;
+    /**
+     * Half-life, in seconds for a slowly-evolving exponential weighted moving
+     * average.
+     * The lower it is, the faster the ABR logic is going to react to recent
+     * bandwidth variation, on the higher and on the lower side.
+     * Should be kept to a higher number than ABR_FAST_EMA for coherency reasons.
+     * @type {Number}
+     */
+    ABR_SLOW_EMA: number;
+    /**
+     * Number of seconds ahead in the buffer after which playback will resume when
+     * seeking on an unbuffered part of the content.
+     * @type {Number}
+     */
+    RESUME_GAP_AFTER_SEEKING: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Number of seconds ahead in the buffer after which playback will resume when
+     * the player was rebuffering due to a low readyState.
+     * @type {Number}
+     */
+    RESUME_GAP_AFTER_NOT_ENOUGH_DATA: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Number of seconds ahead in the buffer after which playback will resume
+     * after the player went through a buffering step.
+     * @type {Number}
+     */
+    RESUME_GAP_AFTER_BUFFERING: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Maximum number of seconds in the buffer based on which a "rebuffering"
+     * strategy will be considered:
+     * The player will pause playback to get enough time building a sufficient
+     * buffer. This mostly happen when seeking in an unbuffered part or when not
+     * enough buffer is ahead of the current position.
+     * @type {Number}
+     */
+    REBUFFERING_GAP: {
+        DEFAULT: number;
+        LOW_LATENCY: number;
+    };
+    /**
+     * Amount of time (in seconds) with data ahead of the current position, at
+     * which we always consider the browser to be able to play.
+     *
+     * If the media element has this amount of data in advance or more but
+     * playback cannot begin, the player will consider it "freezing".
+     */
+    MINIMUM_BUFFER_AMOUNT_BEFORE_FREEZING: number;
+    /**
+     * A media whose position inexplicably does not increment despite playing is
+     * called as "freezing" in the RxPlayer.
+     *
+     * If the media is still "freezing" after waiting for `UNFREEZING_SEEK_DELAY`
+     * milliseconds, the RxPlayer will try to un-freeze the situation by interacting
+     * with the media element.
+     *
+     * Those interactions can be costly in time before playback continue, so it
+     * should be set at a sufficiently high value to avoid false positives.
+     */
+    UNFREEZING_SEEK_DELAY: number;
+    /**
+     * A media whose position inexplicably does not increment despite playing is
+     * called as "freezing" in the RxPlayer.
+     *
+     * A small freezing interval may be normal as the browser may take time before
+     * playing, e.g. after a seek.
+     *
+     * If the media is still "freezing" after waiting for `FREEZING_STALLED_DELAY`
+     * milliseconds, the RxPlayer will emit a BUFFERING state through its API to
+     * notify that the player cannot currently advance.
+     */
+    FREEZING_STALLED_DELAY: number;
+    /**
+     * A media whose position inexplicably does not increment despite playing is
+     * called as "freezing" in the RxPlayer.
+     *
+     * If the media is frozen for a sufficiently large time
+     * (@see UNFREEZING_SEEK_DELAY), the RxPlayer will perform a seek corresponding
+     * to its current position plus `UNFREEZING_DELTA_POSITION` seconds.
+     *
+     * This should be kept short enough as the goal is just to un-freeze lower-level
+     * buffers.
+     */
+    UNFREEZING_DELTA_POSITION: number;
+    /**
+     * Maximum authorized difference between what we calculated to be the
+     * beginning or end of the segment in a media buffer and what we
+     * actually are noticing now.
+     *
+     * If the segment seems to have removed more than this size in seconds, we
+     * will infer that the segment has been garbage collected and we might try to
+     * re-download it.
+     * @type {Number}
+     */
+    MAX_TIME_MISSING_FROM_COMPLETE_SEGMENT: number;
+    /**
+     * The maximum authorized difference, in seconds, between the real buffered
+     * time of a given chunk and what the segment information of the Manifest
+     * tells us.
+     *
+     * Setting a value too high can lead to parts of the media buffer being
+     * linked to the wrong segments and to segments wrongly believed to be still
+     * complete (instead of garbage collected).
+     *
+     * Setting a value too low can lead to parts of the media buffer not being
+     * linked to the concerned segment and to segments wrongly believed to be
+     * partly garbage collected (instead of complete segments).
+     * @type {Number}
+     */
+    MAX_MANIFEST_BUFFERED_START_END_DIFFERENCE: number;
+    /**
+     * The maximum authorized difference, in seconds, between the duration a
+     * segment should have according to the Manifest and the actual duration it
+     * seems to have once pushed to the media buffer.
+     *
+     * Setting a value too high can lead to parts of the media buffer being
+     * linked to the wrong segments and to segments wrongly believed to be still
+     * complete (instead of garbage collected).
+     *
+     * Setting a value too low can lead to parts of the media buffer not being
+     * linked to the concerned segment and to segments wrongly believed to be
+     * partly garbage collected (instead of complete segments). This last point
+     * could lead to unnecessary segment re-downloading.
+     * @type {Number}
+     */
+    MAX_MANIFEST_BUFFERED_DURATION_DIFFERENCE: number;
+    /**
+     * Minimum duration in seconds a segment should be into a buffered range to be
+     * considered as part of that range.
+     * Segments which have less than this amount of time "linked" to a buffered
+     * range will be deleted.
+     *
+     * Setting a value too low can lead in worst-case scenarios to segments being
+     * wrongly linked to the next or previous range it is truly linked too (if
+     * those ranges are too close).
+     *
+     * Setting a value too high can lead to part of the buffer not being assigned
+     * any segment. It also limits the minimum duration a segment can be.
+     *
+     * TODO As of now, this limits the minimum size a complete segment can be. A
+     * better logic would be to also consider the duration of a segment. Though
+     * this logic could lead to bugs with the current code.
+     * @type {Number}
+     */
+    MINIMUM_SEGMENT_SIZE: number;
+    /**
+     * Append windows allow to filter media data from segments if they are outside
+     * a given limit.
+     * Coded frames with presentation timestamp within this range are allowed to
+     * be appended to the media buffer while coded frames outside this range are
+     * filtered out.
+     *
+     * Those are often set to be the start and end of the "Period" the segment is
+     * in.
+     * However, we noticed that some browsers were too aggressive when the exact
+     * limits were set: more data than needed was removed, often leading to
+     * discontinuities.
+     *
+     * Those securities are added to the set windows (substracted from the window
+     * start and added to the window end) to avoid those problems.
+     * @type {Object}
+     */
+    APPEND_WINDOW_SECURITIES: {
+        START: number;
+        END: number;
+    };
+    /**
+     * Maximum interval at which text tracks are refreshed in an "html"
+     * textTrackMode.
+     *
+     * The text tracks are also refreshed on various video events, this interval
+     * will only trigger a refresh if none of those events was received during
+     * that timespan.
+     *
+     * Note that if the TextTrack cue did not change between two intervals or
+     * events, the DOM won't be refreshed.
+     * The TextTrack cues structure is also optimized for fast retrieval.
+     * We should thus not have much of a performance impact here if we set a low
+     * interval.
+     *
+     * @type {Number}
+     */
+    MAXIMUM_HTML_TEXT_TRACK_UPDATE_INTERVAL: number;
+    /**
+     * On browsers with no ResizeObserver API, this will be the interval in
+     * milliseconds at which we should check if the text track element has
+     * changed its size, and updates proportional text-track data accordingly
+     * (like a proportional font-size).
+     *
+     * This is only used:
+     *   - in an "html" textTrackMode
+     *   - when some styling is proportional in the text track data
+     *
+     * Putting a value too low will render faster but might use to much proc time.
+     * Putting a value too high might provoke a re-render too late after the user
+     * changed the element's size (e.g. when going to fullscreen mode).
+     *
+     * @type {Number}
+     */
+    TEXT_TRACK_SIZE_CHECKS_INTERVAL: number;
+    /**
+     * The Buffer padding is a time offset from the current time that affects
+     * the buffer.
+     *
+     * Basically, from a given time, if the current buffer gap number (time
+     * between the current time and the end of the downloaded buffer) is above
+     * the padding described here (of the corresponding type), we won't
+     * reschedule segments for that range.
+     *
+     * This is to avoid excessive re-buffering.
+     *
+     * Keeping the padding too low would increase the risk of re-bufferings.
+     *
+     * Keeping the padding too high would delay visible quality increase.
+     *
+     * @type {Object}
+     */
+    BUFFER_PADDING: {
+        audio: number;
+        video: number;
+        other: number;
+    };
+    /**
+     * Segments of different types are downloaded by steps:
+     *
+     *   - first the audio/video/text Segments which are immediately needed
+     *
+     *   - then once every of those Segments have been downloaded, less-needed
+     *     Segments
+     *
+     *   - then once every of those less-needed Segments have been downloaded,
+     *     even less-needed Segments
+     *
+     *   - etc.
+     *
+     * This stepped download strategy allows to make a better use of network
+     * ressources.
+     *
+     * For example, if more than sufficient audio buffer has been downloaded but
+     * the immediately-needed video Segment is still pending its request, we might
+     * be in a situation of rebuffering.
+     * In that case, a better strategy would be to make sure every network
+     * ressource is allocated for this video Segment before rebuffering happens.
+     *
+     * This is where those steps become useful.
+     *
+     * --
+     *
+     * The numbers defined in this Array describe what the steps are.
+     *
+     * Each number is linked to a distance from the current playing position, in
+     * seconds.
+     * Distances which will be used as limit points, from which a new step is
+     * reached (see example).
+     *
+     * In the RxPlayer's code, each step is then translated in to a priority
+     * number.
+     * The lower is that number, the lower is the step and the lower is the step,
+     * the higher is the priority.
+     *
+     * Note: You can set an empty array to deactivate the steps feature (every
+     * Segments have the same priority).
+     *
+     * @example
+     *
+     * let's imagine the following SEGMENT_PRIORITIES_STEPS array:
+     * [5, 11, 17, 25]
+     *
+     * To link each Segments to a corresponding priority number (and thus to a
+     * specific step), we have to consider the distance between the current
+     * position and the start time of the Segment.
+     *
+     * We have in our example 5 groups, which correspond to the following possible
+     * distances:
+     *   1. inferior to 5 => first step (priority number = 0)
+     *   2. between 5 and 11 => second step (priority number = 1)
+     *   3. between 11 and 17 => third step (priority number = 2)
+     *   4. between 17 and 25 => fourth step (priority number = 3)
+     *   5. superior to 25 => fifth step (priority number = 4)
+     *
+     * Segments corresponding to a lower-step will need to all be downloaded
+     * before Segments of a newer step begin.
+     *
+     * @type {Array.<Number>}
+     */
+    SEGMENT_PRIORITIES_STEPS: number[];
+    /**
+     * Some segment requests are said to be "high priority".
+     *
+     * Requests in that category once done will cancel any segment request that
+     * has a low priority number (see `SEGMENT_PRIORITIES_STEPS`) - meaning a
+     * priority number equal to `MIN_CANCELABLE_PRIORITY` or more.
+     *
+     * Enter here the last priority number that is considered high priority
+     * (beginning by the first step, which has the priority number `0`).
+     * @type {number}
+     */
+    MAX_HIGH_PRIORITY_LEVEL: number;
+    /**
+     * Enter here the first priority step (see `SEGMENT_PRIORITIES_STEPS`) that
+     * will be considered as low priority.
+     *
+     * Segment requests with a low priority will be cancelled if a high priority
+     * segment request (see MAX_HIGH_PRIORITY_LEVEL) is scheduled while they are
+     * pending.
+     *
+     * This number should be strictly superior to the value indicated in
+     * `MAX_HIGH_PRIORITY_LEVEL`.
+     * @type {number}
+     */
+    MIN_CANCELABLE_PRIORITY: number;
+    /**
+     * Codecs used in the videoCapabilities of the MediaKeySystemConfiguration
+     * (DRM).
+     *
+     * Defined in order of importance (first will be tested first etc.)
+     * @type {Array.<string>}
+     */
+    EME_DEFAULT_VIDEO_CODECS: string[];
+    /**
+     * Codecs used in the audioCapabilities of the MediaKeySystemConfiguration
+     * (DRM).
+     *
+     * Defined in order of importance (first will be tested first etc.)
+     * @type {Array.<string>}
+     */
+    EME_DEFAULT_AUDIO_CODECS: string[];
+    /**
+     * Robustnesses used in the {audio,video}Capabilities of the
+     * MediaKeySystemConfiguration (DRM).
+     *
+     * Only used for widevine keysystems.
+     *
+     * Defined in order of importance (first will be tested first etc.)
+     * @type {Array.<string>}
+     */
+    EME_DEFAULT_WIDEVINE_ROBUSTNESSES: string[];
+    /**
+     * Robustnesses used in the {audio,video}Capabilities of the
+     * MediaKeySystemConfiguration (DRM).
+     *
+     * Only used for "com.microsoft.playready.recommendation" keysystems.
+     *
+     * Defined in order of importance (first will be tested first etc.)
+     * @type {Array.<string>}
+     */
+    EME_DEFAULT_PLAYREADY_RECOMMENDATION_ROBUSTNESSES: string[];
+    /**
+     * Link canonical key systems names to their respective reverse domain name,
+     * used in the EME APIs.
+     * This allows to have a simpler API, where users just need to set "widevine"
+     * or "playready" as a keySystem.
+     * @type {Object}
+     */
+    EME_KEY_SYSTEMS: Partial<Record<string, string[]>>;
+    /**
+     * The Manifest parsing logic has a notion of "unsafeMode" which allows to
+     * speed-up this process a lot with a small risk of de-synchronization with
+     * what actually is on the server.
+     * Because using that mode is risky, and can lead to all sort of problems, we
+     * regularly should fall back to a regular "safe" parsing every once in a
+     * while.
+     * This value defines how many consecutive time maximum the "unsafeMode"
+     * parsing can be done.
+     */
+    MAX_CONSECUTIVE_MANIFEST_PARSING_IN_UNSAFE_MODE: number;
+    /**
+     * Minimum time spent parsing the Manifest before we can authorize parsing
+     * it in an "unsafeMode", to speed-up the process with a little risk.
+     * Please note that this parsing time also sometimes includes idle time such
+     * as when the parser is waiting for a request to finish.
+     */
+    MIN_MANIFEST_PARSING_TIME_TO_ENTER_UNSAFE_MODE: number;
+    /**
+     * Minimum amount of <S> elements in a DASH MPD's <SegmentTimeline> element
+     * necessary to begin parsing the current SegmentTimeline element in an
+     * unsafe manner (meaning: with risks of de-synchronization).
+     * This is only done when the "unsafeMode" parsing mode is enabled.
+     */
+    MIN_DASH_S_ELEMENTS_TO_PARSE_UNSAFELY: number;
+    /**
+     * When we detect that the local Manifest might be out-of-sync with the
+     * server's one, we schedule a Manifest refresh.
+     * However, as this "unsynchronization" is only a theory and as we do not want
+     * to send too many Manifest requests, we keep a delay between the last
+     * Manifest refresh done and that one.
+     * This value indicates which delay we want. Note that the Manifest could
+     * still be refreshed before this delay for other reasons.
+     * @type {Number}
+     */
+    OUT_OF_SYNC_MANIFEST_REFRESH_DELAY: number;
+    /**
+     * When a partial Manifest update (that is an update with a partial sub-set
+     * of the Manifest) fails, we will perform an update with the whole Manifest
+     * instead.
+     * To not overload the client - as parsing a Manifest can be resource heavy -
+     * we set a minimum delay to wait before doing the corresponding request.
+     * @type {Number}
+     */
+    FAILED_PARTIAL_UPDATE_MANIFEST_REFRESH_DELAY: number;
+    /**
+     * DASH Manifest based on a SegmentTimeline should normally have an
+     * MPD@minimumUpdatePeriod attribute which should be sufficient to
+     * know when to refresh it.
+     * However, there is a specific case, for when it is equal to 0.
+     * As of DASH-IF IOP (valid in v4.3), when a DASH's MPD set a
+     * MPD@minimumUpdatePeriod to `0`, a client should not refresh the MPD
+     * unless told to do so through inband events, in the stream.
+     * In reality however, we found it to not always be the case (even with
+     * DASH-IF own streams) and moreover to not always be the best thing to do.
+     * We prefer to refresh in average at a regular interval when we do not have
+     * this information.
+     * /!\ This value is expressed in seconds.
+     */
+    DASH_FALLBACK_LIFETIME_WHEN_MINIMUM_UPDATE_PERIOD_EQUAL_0: number;
+    /**
+     * Default value for the maximum number of simultaneous MediaKeySessions that
+     * will be kept in a cache (linked to the MediaKeys instance) to avoid doing
+     * superfluous license requests.
+     *
+     * If this number is reached, any new session creation will close the oldest
+     * one.
+     * Another value can be configured through the API, in which case this default
+     * will be overwritten.
+     * @type {Number}
+     */
+    EME_DEFAULT_MAX_SIMULTANEOUS_MEDIA_KEY_SESSIONS: number;
+    /**
+     * When playing contents with a persistent license, we will usually store some
+     * information related to that MediaKeySession, to be able to play it at a
+     * later time.
+     *
+     * Those information are removed once a MediaKeySession is not considered
+     * as "usable" anymore. But to know that, the RxPlayer has to load it.
+     *
+     * But the RxPlayer does not re-load every persisted MediaKeySession every
+     * time to check each one of them one by one, as this would not be a
+     * performant thing to do.
+     *
+     * So this is only done when and if the corresponding content is encountered
+     * again and only if it contains the same initialization data.
+     *
+     * We have to consider that those "information" contain binary data which can
+     * be of arbitrary length. Size taken by an array of them can relatively
+     * rapidly take a lot of space in JS memory.
+     *
+     * So to avoid this storage to take too much space (would it be in the chosen
+     * browser's storage or in JS memory), we now set a higher bound for the
+     * amount of MediaKeySession information that can be stored at the same time.
+     *
+     * I set the value of 1000 here, as it seems big enough to not be considered a
+     * problem (though it can become one, when contents have a lot of keys per
+     * content), and still low enough so it should not cause much problem (my
+     * method to choose that number was to work with power of 10s and choosing the
+     * amount which seemed the most sensible one).
+     *
+     * This wasn't battle-tested however.
+     */
+    EME_MAX_STORED_PERSISTENT_SESSION_INFORMATION: number;
+    /**
+     * After loading a persistent MediaKeySession, the RxPlayer needs to ensure
+     * that its keys still allow to decrypt a content.
+     *
+     * However on some browsers, the `keyStatuses` property that we used to check
+     * the keys' satuses linked to that session can be empty for some time after
+     * the loading operation is done.
+     *
+     * This value allows to configure a delay in milliseconds that will be the
+     * maximum time we will wait after a persistent session is loaded.
+     * If after that time, the `keyStatuses` property is still empty, we will
+     * consider that session as not usable.
+     */
+    EME_WAITING_DELAY_LOADED_SESSION_EMPTY_KEYSTATUSES: number;
+    /**
+     * The player relies on browser events and properties to update its status to
+     * "ENDED".
+     *
+     * Sadly in some cases, like in Chrome 54, this event is never triggered on
+     * some contents probably due to a browser bug.
+     *
+     * This threshold resolves this issue by forcing the status to "ENDED" when:
+     *   1. the player is stalling
+     *   2. the absolute difference between current playback time and duration is
+     *      under this value
+     *
+     * If set to null, this workaround is disabled and the player only relies on
+     * browser events.
+     *
+     * @type {Number|null}
+     */
+    FORCED_ENDED_THRESHOLD: number;
+    /**
+     * Maximum duration from the current position we will let in the buffer when
+     * switching an Adaptation/Representations of a given type.
+     *
+     * For example, if we have ``text: { before: 1, after: 4 }``, it means that
+     * when switching subtitles, we will let 1 second before and 4 second after
+     * the current position in the previous language (until the new segments
+     * overwrite it).
+     * This is to allow smooth transitions and avoid de-synchronization that
+     * can happen when removing the content being decoded.
+     * @type {Object}
+     */
+    ADAP_REP_SWITCH_BUFFER_PADDINGS: {
+        video: {
+            before: number;
+            after: number;
+        };
+        audio: {
+            before: number;
+            after: number;
+        };
+        text: {
+            before: number;
+            after: number;
+        };
+    };
+    /**
+     * Interval, in milliseconds, at which we should manually flush
+     * SourceBuffers.
+     * Some browsers (happened with firefox 66) sometimes "forget" to send us
+     * `update` or `updateend` events.
+     * In that case, we're completely unable to continue the queue here and
+     * stay locked in a waiting state.
+     * This interval is here to check at regular intervals if the underlying
+     * SourceBuffer is currently updating.
+     * @type {Number}
+     */
+    SOURCE_BUFFER_FLUSHING_INTERVAL: number;
+    /**
+     * Any already-pushed segment starting before or at the current position +
+     * CONTENT_REPLACEMENT_PADDING won't be replaced by new segments.
+     *
+     * This allows to avoid overwriting segments that are currently being decoded
+     * as we encountered many decoding issues when doing so.
+     * @type {Number} - in seconds
+     */
+    CONTENT_REPLACEMENT_PADDING: number;
+    /**
+     * For video and audio segments, determines two thresholds below which :
+     * - The segment is considered as loaded from cache
+     * - The segment may be loaded from cache depending on the previous request
+     */
+    CACHE_LOAD_DURATION_THRESHOLDS: {
+        video: number;
+        audio: number;
+    };
+    /** Interval we will use to poll for checking if an event shall be emitted */
+    STREAM_EVENT_EMITTER_POLL_INTERVAL: number;
+    /**
+     * In Javascript, numbers are encoded in a way that a floating number may be
+     * represented internally with a rounding error. When multiplying times in
+     * seconds by the timescale, we've encoutered cases were the rounding error
+     * was amplified by a factor which is about the timescale.
+     * Example :
+     * (192797480.641122).toFixed(20) = 192797480.64112201333045959473
+     * (error is 0.0000000133...)
+     * 192797480.641122 * 10000000 = 1927974806411220.2 (error is 0.2)
+     * 192797480.641122 * 10000000 * 4 = 7711899225644881 (error is 1)
+     * The error is much more significant here, once the timescale has been
+     * applied.
+     * Thus, we consider that our max tolerable rounding error is 1ms.
+     * It is much more than max rounding errors when seen into practice,
+     * and not significant from the media loss perspective.
+     */
+    DEFAULT_MAXIMUM_TIME_ROUNDING_ERROR: number;
+    /**
+     * RxPlayer's media buffers have a linked history registering recent events
+     * that happened on those.
+     * The reason is to implement various heuristics in case of weird browser
+     * behavior.
+     *
+     * The `BUFFERED_HISTORY_RETENTION_TIME` is the minimum age an entry of
+     * that history can have before being removed from the history.
+     */
+    BUFFERED_HISTORY_RETENTION_TIME: number;
+    /**
+     * RxPlayer's media buffers have a linked history registering recent events
+     * that happened on those.
+     * The reason is to implement various heuristics in case of weird browser
+     * behavior.
+     *
+     * The `BUFFERED_HISTORY_RETENTION_TIME` is the maximum number of entries
+     * there can be in that history.
+     */
+    BUFFERED_HISTORY_MAXIMUM_ENTRIES: number;
+    /**
+     * Minimum buffer in seconds ahead relative to current time
+     * we should be able to download, even in cases of saturated memory.
+     */
+    MIN_BUFFER_AHEAD: number;
+    /**
+     * Distance in seconds behind the current position
+     * the player will free up to in the case we agressively free up memory
+     * It is set to avoid playback issues
+     */
+    UPTO_CURRENT_POSITION_CLEANUP: number;
+    /**
+     * Default "switching mode" used when locking video Representations.
+     * That is, which behavior the RxPlayer should have by default when
+     * explicitely and manually switching from a previous set of video
+     * Representations to a new one.
+     */
+    DEFAULT_VIDEO_REPRESENTATIONS_SWITCHING_MODE: "seamless";
+    /**
+     * Default "switching mode" used when locking audio Representations.
+     * That is, which behavior the RxPlayer should have by default when
+     * explicitely and manually switching from a previous set of audio
+     * Representations to a new one.
+     */
+    DEFAULT_AUDIO_REPRESENTATIONS_SWITCHING_MODE: "seamless";
+    /**
+     * Default "switching mode" used when switching between video tracks.
+     * That is, which behavior the RxPlayer should have by default when
+     * explicitely and manually switching from a previous video track to a new
+     * one.
+     */
+    DEFAULT_VIDEO_TRACK_SWITCHING_MODE: "reload";
+    /**
+     * Default "switching mode" used when switching between audio tracks.
+     * That is, which behavior the RxPlayer should have by default when
+     * explicitely and manually switching from a previous audio track to a new
+     * one.
+     */
+    DEFAULT_AUDIO_TRACK_SWITCHING_MODE: "seamless";
+};
+export type IDefaultConfig = typeof DEFAULT_CONFIG;
+export default DEFAULT_CONFIG;