rx-player 4.0.0-dev.2023121900 → 4.0.0-dev.2024022100

package/dist/_esm5.processed/default_config.d.ts

@@ -1,1143 +0,0 @@
-/**
- * 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";
-    /**
-     * Strategy to adopt when manually setting the current bitrate.
-     * Can be either:
-     *   - "seamless": transitions are very smooth but not immediate.
-     *   - "direct": the quality switch happens immediately but to achieve that,
-     *     the player will need to set a new MediaSource on the media element in
-     *     some cases. This often leads to a black screen + unavailable APIs
-     *     during a short moment.
-     * @type {string}
-     */
-    DEFAULT_MANUAL_BITRATE_SWITCHING_MODE: "direct" | "seamless";
-    /**
-     * 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;
-    /**
-     * Strategy to adopt when manually switching of audio adaptation.
-     * Can be either:
-     *    - "seamless": transitions are smooth but could be not immediate.
-     *    - "direct": that strategy will perform a very small seek that result
-     *    most of the time by a flush of the current buffered data, by doing
-     *    that we allow quicker transition between audio track, but we could
-     *    see appear a RELOADING or a SEEKING state.
-     */
-    DEFAULT_AUDIO_TRACK_SWITCHING_MODE: "direct" | "seamless";
-    /**
-     * 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;
-    /**
-     * If set to false, "native" subtitles (in a <track> element) will be hidden
-     * by default.
-     * @type {Boolean}
-     */
-    DEFAULT_SHOW_NATIVE_SUBTITLE: boolean;
-    /**
-     * If set to true, the player will by default stop immediately and unload the
-     * content on reaching the end of the media.
-     *
-     * If set to false, it will not unload nor stop by default, leaving the user
-     * free to seek in the already-loaded content.
-     *
-     * Set to `true` for legacy reasons.
-     * @type {Boolean}
-     */
-    DEFAULT_STOP_AT_END: 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" | "image" | "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" | "image" | "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_INITIAL_BITRATES: {
-        audio: number;
-        video: number;
-        other: number;
-    };
-    /**
-     * Default bitrate floor initially set to dictate the minimum bitrate the
-     * adaptive logic can automatically switch to.
-     *
-     * If no track is found with a quality superior or equal to the
-     * bitrate there, the lowest bitrate will be taken instead.
-     *
-     * Set to Infinity to discard any limit in the ABR strategy.
-     * @type {Object}
-     */
-    DEFAULT_MIN_BITRATES: Record<"audio" | "video" | "other", number>;
-    /**
-     * Default bitrate ceil initially set to dictate the maximum bitrate the
-     * adaptive logic can automatically switch to.
-     *
-     * If no track is found with a quality inferior or equal to the
-     * bitrate there, the lowest bitrate will be taken instead.
-     *
-     * Set to Infinity to discard any limit in the ABR strategy.
-     * @type {Object}
-     */
-    DEFAULT_MAX_BITRATES: Record<"audio" | "video" | "other", 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 player is 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_WHEN_HIDDEN: boolean;
-    /**
-     * 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;
-    /**
-     * If true, the video representations you can switch to in adaptive mode
-     * are limited by the video element's width.
-     *
-     * Basically in that case, we won't switch to a video Representation with
-     * a width higher than the current width of the video HTMLElement.
-     * @type {Boolean}
-     */
-    DEFAULT_LIMIT_VIDEO_WIDTH: boolean;
-    /**
-     * 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.
-     *   - if it has a high chance of being due to the user being offline, a
-     *     separate counter is used (see DEFAULT_MAX_REQUESTS_RETRY_ON_OFFLINE).
-     * @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.
-     *   - if it has a high chance of being due to the user being offline, a
-     *     separate counter is used (see DEFAULT_MAX_REQUESTS_RETRY_ON_OFFLINE).
-     * @type Number
-     */
-    DEFAULT_MAX_REQUESTS_RETRY_ON_ERROR: number;
-    /**
-     * Under some circonstances, we're able to tell that the user is offline (see
-     * the compat files).
-     * When this happens, and xhr requests fails due to an error event (you might
-     * still be able to perform xhr offline, e.g. on localhost), you might want to
-     * retry indefinitely or with a higher number of retry than if the error is
-     * due to a CDN problem.
-     *
-     * A capped exponential backoff will still be used (like for an error code).
-     * @type {Number}
-     */
-    DEFAULT_MAX_REQUESTS_RETRY_ON_OFFLINE: 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" key system.
-     *
-     * Defined in order of importance (first will be tested first etc.)
-     * @type {Array.<string>}
-     */
-    EME_DEFAULT_PLAYREADY_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 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}
-     */
-    ADAPTATION_SWITCH_BUFFER_PADDINGS: {
-        video: {
-            before: number;
-            after: number;
-        };
-        audio: {
-            before: number;
-            after: number;
-        };
-        text: {
-            before: number;
-            after: number;
-        };
-        image: {
-            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;
-};
-export type IDefaultConfig = typeof DEFAULT_CONFIG;
-export default DEFAULT_CONFIG;