// Copyright 2018 The Fuchsia Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

library fuchsia.media;

using fuchsia.media.audio;

///
/// AudioRenderers can be in one of two states at any time: _configurable_ or _operational_. A
/// renderer is considered operational whenever it has packets queued to be rendered; otherwise it
/// is _configurable_. Once an AudioRenderer enters the operational state, calls to "configuring"
/// methods are disallowed and will cause the audio service to disconnect the client's connection.
/// The following are considered configuring methods: `AddPayloadBuffer`, `SetPcmStreamType`,
/// `SetStreamType`, `SetPtsUnits`, `SetPtsContinuityThreshold`.
///
/// If an AudioRenderer must be reconfigured, the client must ensure that no packets are still
/// enqueued when these "configuring" methods are called. Thus it is best practice to call
/// `DiscardAllPackets` on the AudioRenderer (and ideally `Stop` before `DiscardAllPackets`), prior
/// to reconfiguring the renderer.
///
protocol AudioRenderer {
    compose StreamBufferSet;
    compose StreamSink;

    /// Binds to the gain control for this AudioRenderer.
    BindGainControl(request<fuchsia.media.audio.GainControl> gain_control_request);

    /// Sets the units used by the presentation (media) timeline. By default, PTS units are
    /// nanoseconds (as if this were called with numerator of 1e9 and denominator of 1).
    SetPtsUnits(uint32 tick_per_second_numerator,
                uint32 tick_per_second_denominator);

    /// Sets the maximum threshold (in seconds) between explicit user-provided PTS
    /// and expected PTS (determined using interpolation). Beyond this threshold,
    /// a stream is no longer considered 'continuous' by the renderer.
    ///
    /// Defaults to an interval of half a PTS 'tick', using the currently-defined PTS units.
    /// Most users should not need to change this value from its default.
    ///
    /// Example:
    /// A user is playing back 48KHz audio from a container, which also contains
    /// video and needs to be synchronized with the audio. The timestamps are
    /// provided explicitly per packet by the container, and expressed in mSec
    /// units. This means that a single tick of the media timeline (1 mSec)
    /// represents exactly 48 frames of audio. The application in this scenario
    /// delivers packets of audio to the AudioRenderer, each with exactly 470
    /// frames of audio, and each with an explicit timestamp set to the best
    /// possible representation of the presentation time (given this media
    /// clock's resolution). So, starting from zero, the timestamps would be..
    ///
    /// [ 0, 10, 20, 29, 39, 49, 59, 69, 78, 88, ... ]
    ///
    /// In this example, attempting to use the presentation time to compute the
    /// starting frame number of the audio in the packet would be wrong the
    /// majority of the time. The first timestamp is correct (by definition), but
    /// it will be 24 packets before the timestamps and frame numbers come back
    /// into alignment (the 24th packet would start with the 11280th audio frame
    /// and have a PTS of exactly 235).
    ///
    /// One way to fix this situation is to set the PTS continuity threshold
    /// (henceforth, CT) for the stream to be equal to 1/2 of the time taken by
    /// the number of frames contained within a single tick of the media clock,
    /// rounded up. In this scenario, that would be 24.0 frames of audio, or 500
    /// uSec. Any packets whose expected PTS was within +/-CT frames of the
    /// explicitly provided PTS would be considered to be a continuation of the
    /// previous frame of audio. For this example, calling 'SetPtsContinuityThreshold(0.0005)'
    /// would work well.
    ///
    /// Other possible uses:
    /// Users who are scheduling audio explicitly, relative to a clock which has
    /// not been configured as the reference clock, can use this value to control
    /// the maximum acceptable synchronization error before a discontinuity is
    /// introduced. E.g., if a user is scheduling audio based on a recovered
    /// common media clock, and has not published that clock as the reference
    /// clock, and they set the CT to 20mSec, then up to 20mSec of drift error
    /// can accumulate before the AudioRenderer deliberately inserts a
    /// presentation discontinuity to account for the error.
    ///
    /// Users whose need to deal with a container where their timestamps may be
    /// even less correct than +/- 1/2 of a PTS tick may set this value to
    /// something larger. This should be the maximum level of inaccuracy present
    /// in the container timestamps, if known. Failing that, it could be set to
    /// the maximum tolerable level of drift error before absolute timestamps are
    /// explicitly obeyed. Finally, a user could set this number to a very large
    /// value (86400.0 seconds, for example) to effectively cause *all*
    /// timestamps to be ignored after the first, thus treating all audio as
    /// continuous with previously delivered packets. Conversely, users who wish
    /// to *always* explicitly schedule their audio packets exactly may specify
    /// a CT of 0.
    ///
    /// Note: explicitly specifying high-frequency PTS units reduces the default
    /// continuity threshold accordingly. Internally, this threshold is stored as an
    /// integer of 1/8192 subframes. The default threshold is computed as follows:
    ///     RoundUp((AudioFPS/PTSTicksPerSec) * 4096) / (AudioFPS * 8192)
    /// For this reason, specifying PTS units with a frequency greater than 8192x
    /// the frame rate (or NOT calling SetPtsUnits, which accepts the default PTS
    /// unit of 1 nanosec) will result in a default continuity threshold of zero.
    ///
    SetPtsContinuityThreshold(float32 threshold_seconds);

    /// Retrieves the stream's reference clock. The returned handle will have READ, DUPLICATE
    /// and TRANSFER rights, and will refer to a zx::clock that is MONOTONIC and CONTINUOUS.
    ///
    GetReferenceClock() -> (handle<clock> reference_clock);

    /// Sets the reference clock that controls this renderer's playback rate. If the input
    /// parameter is valid, it must have READ, DUPLICATE, TRANSFER rights and refer to a clock
    /// that is MONOTONIC and CONTINUOUS. If instead ZX_HANDLE_INVALID is passed, then this
    /// renderer will use the so-called 'optimal' clock generated by AudioCore for this stream.
    ///
    /// Although |SetReferenceClock| is not considered a configuring method, calling it when
    /// packets are queued (even if paused) will generally cause a reference timeline
    /// discontinuity, with packets either discarded or delayed (based on the discontinuity).
    /// Cases that PRESERVE continuity would include setting as reference either the 'optimal'
    /// clock provided by AudioCore, or the clone of a client-controlled reference clock.
    ///
    SetReferenceClock(handle<clock>? reference_clock);

    /// Sets the usage of the render stream. This method may not be called after
    /// `SetPcmStreamType` is called. The default usage is `MEDIA`.
    SetUsage(AudioRenderUsage usage);

    /// Sets the type of the stream to be delivered by the client. Using this method implies
    /// that the stream encoding is `AUDIO_ENCODING_LPCM`.
    ///
    /// This must be called before `Play` or `PlayNoReply`. After a call to `SetPcmStreamType`,
    /// the client must then send an `AddPayloadBuffer` request, then the various `StreamSink`
    /// methods such as `SendPacket`/`SendPacketNoReply`.
    SetPcmStreamType(AudioStreamType type);

    /// Enable or disable notifications about changes to the minimum clock lead
    /// time (in nanoseconds) for this AudioRenderer. Calling this method with
    /// 'enabled' set to true will trigger an immediate `OnMinLeadTimeChanged`
    /// event with the current minimum lead time for the AudioRenderer. If the
    /// value changes, an `OnMinLeadTimeChanged` event will be raised with the
    /// new value. This behavior will continue until the user calls
    /// `EnableMinLeadTimeEvents(false)`.
    ///
    /// The minimum clock lead time is the amount of time ahead of the reference
    /// clock's understanding of "now" that packets needs to arrive (relative to
    /// the playback clock transformation) in order for the mixer to be able to
    /// mix packet. For example...
    ///
    /// ++ Let the PTS of packet X be P(X)
    /// ++ Let the function which transforms PTS -> RefClock be R(p) (this
    ///    function is determined by the call to Play(...)
    /// ++ Let the minimum lead time be MLT
    ///
    /// If R(P(X)) < RefClock.Now() + MLT
    /// Then the packet is late, and some (or all) of the packet's payload will
    /// need to be skipped in order to present the packet at the scheduled time.
    ///
    // TODO(mpuryear): What should the units be here?  Options include...
    //
    // 1) Normalized to nanoseconds (this is the current API)
    // 2) Reference clock units (what happens if the reference clock changes?)
    // 3) PTS units (what happens when the user changes the PTS units?)
    //
    // TODO(mpuryear): Should `EnableMinLeadTimeEvents` have an optional -> ()
    // return value for synchronization purposes?  Probably not; users should be
    // able to send a disable request and clear their event handler if they no
    // longer want notifications. Their in-process dispatcher framework can
    // handle draining and dropping any lead time changed events that were
    // already in flight when the disable message was sent.
    //
    EnableMinLeadTimeEvents(bool enabled);
    -> OnMinLeadTimeChanged(int64 min_lead_time_nsec);

    // TODO(mpuryear): Eliminate this method when possible. Right now, it is
    // used by code requiring synchronous FIDL interfaces to talk to
    // AudioRenderers.
    ///
    /// While it is possible to call `GetMinLeadTime` before `SetPcmStreamType`,
    /// there's little reason to do so. This is because lead time is a function
    /// of format/rate, so lead time will be recalculated after `SetPcmStreamType`.
    /// If min lead time events are enabled before `SetPcmStreamType` (with
    /// `EnableMinLeadTimeEvents(true)`), then an event will be generated in
    /// response to `SetPcmStreamType`.
    GetMinLeadTime() -> (int64 min_lead_time_nsec);

    /// Immediately put the AudioRenderer into a playing state. Start the advance
    /// of the media timeline, using specific values provided by the caller (or
    /// default values if not specified). In an optional callback, return the
    /// timestamp values ultimately used -- these set the ongoing relationship
    /// between the media and reference timelines (i.e., how to translate between
    /// the domain of presentation timestamps, and the realm of local system
    /// time).
    ///
    /// Local system time is specified in units of nanoseconds; media_time is
    /// specified in the units defined by the user in the `SetPtsUnits` function,
    /// or nanoseconds if `SetPtsUnits` is not called.
    ///
    /// The act of placing an AudioRenderer into the playback state establishes a
    /// relationship between 1) the user-defined media (or presentation) timeline
    /// for this particular AudioRenderer, and 2) the real-world system reference
    /// timeline. To communicate how to translate between timelines, the Play()
    /// callback provides an equivalent timestamp in each time domain. The first
    /// value ('reference_time') is given in terms of the local system clock; the
    /// second value ('media_time') is what media instant exactly corresponds to
    /// that local time. Restated, the frame at 'media_time' in the audio stream
    /// should be presented at system local time 'reference_time'.
    ///
    /// Note: on calling this API, media_time immediately starts advancing. It is
    /// possible (if uncommon) for a caller to specify a system time that is
    /// far in the past, or far into the future. This, along with the specified
    /// media time, is simply used to determine what media time corresponds to
    /// 'now', and THAT media time is then intersected with presentation
    /// timestamps of packets already submitted, to determine which media frames
    /// should be presented next.
    ///
    /// With the corresponding reference_time and media_time values, a user can
    /// translate arbitrary time values from one timeline into the other. After
    /// calling `SetPtsUnits(pts_per_sec_numerator, pts_per_sec_denominator)` and
    /// given the 'ref_start' and 'media_start' values from `Play`, then for
    /// any 'ref_time':
    ///
    /// media_time = ( (ref_time - ref_start) / 1e9
    ///                * (pts_per_sec_numerator / pts_per_sec_denominator) )
    ///              + media_start
    ///
    /// Conversely, for any presentation timestamp 'media_time':
    ///
    /// ref_time = ( (media_time - media_start)
    ///              * (pts_per_sec_denominator / pts_per_sec_numerator)
    ///              * 1e9 )
    ///            + ref_start
    ///
    /// Users, depending on their use case, may optionally choose not to specify
    /// one or both of these timestamps. A timestamp may be omitted by supplying
    /// the special value '`NO_TIMESTAMP`'. The AudioRenderer automatically deduces
    /// any omitted timestamp value using the following rules:
    ///
    /// Reference Time
    /// If 'reference_time' is omitted, the AudioRenderer will select a "safe"
    /// reference time to begin presentation, based on the minimum lead times for
    /// the output devices that are currently bound to this AudioRenderer. For
    /// example, if an AudioRenderer is bound to an internal audio output
    /// requiring at least 3 mSec of lead time, and an HDMI output requiring at
    /// least 75 mSec of lead time, the AudioRenderer might (if 'reference_time'
    /// is omitted) select a reference time 80 mSec from now.
    ///
    /// Media Time
    /// If media_time is omitted, the AudioRenderer will select one of two
    /// values.
    /// - If the AudioRenderer is resuming from the paused state, and packets
    /// have not been discarded since being paused, then the AudioRenderer will
    /// use a media_time corresponding to the instant at which the presentation
    /// became paused.
    /// - If the AudioRenderer is being placed into a playing state for the first
    /// time following startup or a 'discard packets' operation, the initial
    /// media_time will be set to the PTS of the first payload in the pending
    /// packet queue. If the pending queue is empty, initial media_time will be
    /// set to zero.
    ///
    /// Return Value
    /// When requested, the AudioRenderer will return the 'reference_time' and
    /// 'media_time' which were selected and used (whether they were explicitly
    /// specified or not) in the return value of the play call.
    ///
    /// Examples
    /// 1. A user has queued some audio using `SendPacket` and simply wishes them
    /// to start playing as soon as possible. The user may call Play without
    /// providing explicit timestamps -- `Play(NO_TIMESTAMP, NO_TIMESTAMP)`.
    ///
    /// 2. A user has queued some audio using `SendPacket`, and wishes to start
    /// playback at a specified 'reference_time', in sync with some other media
    /// stream, either initially or after discarding packets. The user would call
    /// `Play(reference_time, NO_TIMESTAMP)`.
    ///
    /// 3. A user has queued some audio using `SendPacket`. The first of these
    /// packets has a PTS of zero, and the user wishes playback to begin as soon
    /// as possible, but wishes to skip all of the audio content between PTS 0
    /// and PTS 'media_time'. The user would call
    /// `Play(NO_TIMESTAMP, media_time)`.
    ///
    /// 4. A user has queued some audio using `SendPacket` and want to present
    /// this media in synch with another player in a different device. The
    /// coordinator of the group of distributed players sends an explicit
    /// message to each player telling them to begin presentation of audio at
    /// PTS 'media_time', at the time (based on the group's shared reference
    /// clock) 'reference_time'. Here the user would call
    /// `Play(reference_time, media_time)`.
    ///
    // TODO(mpuryear): Define behavior in the case that a user calls `Play()`
    // while the system is already playing. We should probably do nothing but
    // return a valid correspondence pair in response -- unless both reference
    // and media times are provided (and do not equate to the current timeline
    // relationship), in which case we should introduce a discontinuity.
    //
    // TODO(mpuryear): Collapse these if we ever have optional retvals in FIDL
    Play(int64 reference_time, int64 media_time)
        -> (int64 reference_time, int64 media_time);
    PlayNoReply(int64 reference_time, int64 media_time);

    /// Immediately put the AudioRenderer into the paused state and then report
    /// the relationship between the media and reference timelines which was
    /// established (if requested).
    ///
    // TODO(mpuryear): Define behavior in the case that a user calls `Pause()`
    // while the system is already in the paused state. We should probably do
    // nothing but provide a valid correspondence pair in response.
    //
    // TODO(mpuryear): Collapse these if we ever have optional retvals in FIDL
    Pause() -> (int64 reference_time, int64 media_time);
    PauseNoReply();

    // TODO(mpuryear): Spec methods/events which can be used for unintentional
    // discontinuity/underflow detection.
    //
    // TODO(mpuryear): Spec methods/events which can be used to report routing
    // changes. (Presuming that they belong at this level at all; they may
    // belong on some sort of policy object).
    //
    // TODO(mpuryear): Spec methods/events which can be used to report policy
    // induced gain/ducking changes. (Presuming that they belong at this level
    // at all; they may belong on some sort of policy object).
};