// Copyright 2017 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;
using zx;

/// Configuration for a capturer which will receive a loopback stream
/// a system output.
type LoopbackAudioCapturerConfiguration = table {};

/// Configuration for a capturer which will receive a stream from an
/// input device.
type InputAudioCapturerConfiguration = table {
    1: usage AudioCaptureUsage;
};

/// Configuration for an audio Capturer.
type AudioCapturerConfiguration = strict union {
    1: loopback LoopbackAudioCapturerConfiguration;
    2: input InputAudioCapturerConfiguration;
};

// TODO(mpuryear): Routing policy needs to become more capable than this.
// Clients will need to be able to request sets of inputs/outputs/renderers,
// make changes to theses sets, have their requests vetted by policy (do they
// have the permission to capture this private stream, do they have the
// permission to capture at this frame rate, etc...).  Eventually, this
// functionality will need to be expressed at the AudioPolicy level, not here.
// TODO(mpuryear): can we come up with better names than these?  Both are
// really async modes under the hood).
// TODO(mpuryear): Add a new method to replace `DiscardAllPackets`, which will
// be removed from `StreamSource`.
// TODO(mpuryear): Fix obsolete docs.
// TODO(mpuryear): Specify the way in which timestamps relative to a different
// clock (such as an audio domain clock) may be delivered to a client.
/// AudioCapturer
///
/// An AudioCapturer is an interface returned from an fuchsia.media.Audio's
/// CreateAudioCapturer method, which may be used by clients to capture audio
/// from either the current default audio input device, or the current default
/// audio output device depending on the flags passed during creation.
///
/// ** Format support **
///
/// See (Get|Set)StreamType below. By default, the captured stream type will be
/// initially determined by the currently configured stream type of the source
/// that the AudioCapturer was bound to at creation time. Users may either fetch
/// this type using GetStreamType, or they may choose to have the media
/// resampled or converted to a type of their choosing by calling SetStreamType.
/// Note: the stream type may only be set while the system is not running,
/// meaning that there are no pending capture regions (specified using CaptureAt)
/// and that the system is not currently running in 'async' capture mode.
///
/// ** Buffers and memory management **
///
/// Audio data is captured into a shared memory buffer (a VMO) supplied by the
/// user to the AudioCapturer during the AddPayloadBuffer call.  Please note the
/// following requirements related to the management of the payload buffer.
///
/// + The payload buffer must be supplied before any capture operation may
///   start.  Any attempt to start capture (via either CaptureAt or
///   StartAsyncCapture) before a payload buffer has been established is an
///   error.
/// + The payload buffer may not be changed while there are any capture
///   operations pending.
/// + The stream type may not be changed after the payload buffer has been set.
/// + The payload buffer must be an integral number of audio frame sizes (in
///   bytes)
/// + When running in 'async' mode (see below), the payload buffer must be at
///   least as large as twice the frames_per_packet size specified during
///   StartAsyncCapture.
/// + The handle to the payload buffer supplied by the user must be readable,
///   writable, mappable and transferable.
/// + Users should always treat the payload buffer as read-only.
///
/// ** Synchronous vs. Asynchronous capture mode **
///
/// The AudioCapturer interface can be used in one of two mutually exclusive
/// modes: Synchronous and Asynchronous.  A description of each mode and their
/// tradeoffs is given below.
///
/// ** Synchronous mode **
///
/// By default, AudioCapturer instances are running in 'sync' mode.  They will
/// only capture data when a user supplies at least one region to capture into
/// using the CaptureAt method.  Regions supplied in this way will be filled in
/// the order that they are received and returned to the client as StreamPackets
/// via the return value of the CaptureAt method.  If an AudioCapturer instance
/// has data to capture, but no place to put it (because there are no more
/// pending regions to fill), the next payload generated will indicate that their
/// has been an overflow by setting the Discontinuity flag on the next produced
/// StreamPacket.  Synchronous mode may not be used in conjunction with
/// Asynchronous mode.  It is an error to attempt to call StartAsyncCapture while
/// the system still regions supplied by CaptureAt waiting to be filled.
///
/// If a user has supplied regions to be filled by the AudioCapturer instance in
/// the past, but wishes to reclaim those regions, they may do so using the
/// DiscardAllPackets method.  Calling the DiscardAllPackets method will cause
/// all pending regions to be returned, but with `NO_TIMESTAMP` as their
/// StreamPacket's PTS.  See "Timing and Overflows", below, for a discussion of
/// timestamps and discontinuity flags. After a DiscardAllPackets operation,
/// an OnEndOfStream event will be produced.  While an AudioCapturer will never
/// overwrite any region of the payload buffer after a completed region is
/// returned, it may overwrite the unfilled portions of a partially filled
/// buffer which has been returned as a result of a DiscardAllPackets operation.
///
/// ** Asynchronous mode **
///
/// While running in 'async' mode, clients do not need to explicitly supply
/// shared buffer regions to be filled by the AudioCapturer instance. Instead, a
/// client enters into 'async' mode by calling StartAsyncCapture and supplying a
/// callback interface and the number of frames to capture per-callback. Once
/// running in async mode, the AudioCapturer instance will identify which
/// payload buffer regions to capture into, capture the specified number of
/// frames, then deliver those frames as StreamPackets using the OnPacketCapture
/// FIDL event. Users may stop capturing and return the AudioCapturer instance to
/// 'sync' mode using the StopAsyncCapture method.
///
/// It is considered an error to attempt any of the following operations.
///
/// + To attempt to enter 'async' capture mode when no payload buffer has been
///   established.
/// + To specify a number of frames to capture per payload which does not permit
///   at least two contiguous capture payloads to exist in the established
///   shared payload buffer simultaneously.
/// + To send a region to capture into using the CaptureAt method while the
///   AudioCapturer instance is running in 'async' mode.
/// + To attempt to call DiscardAllPackets while the AudioCapturer instance is
///   running in 'async' mode.
/// + To attempt to re-start 'async' mode capturing without having first
///   stopped.
/// + To attempt any operation except for SetGain while in the process of
///   stopping.
///
/// ** Synchronizing with a StopAsyncCapture operation **
///
/// Stopping asynchronous capture mode and returning to synchronous capture mode
/// is an operation which takes time.  Aside from SetGain, users may not call any
/// other methods on the AudioCapturer interface after calling StopAsyncCapture
/// (including calling StopAsyncCapture again) until after the stop operation has
/// completed.  Because of this, it is important for users to be able to
/// synchronize with the stop operation.  Two mechanisms are provided for doing
/// so.
///
/// The first is to use StopAsyncCapture (not the NoReply variant). When the user's
/// callback has been called, they can be certain that stop operation is complete
/// and that the AudioCapturer instance has returned to synchronous operation
/// mode.
///
/// The second way to determine that a stop operation has completed is to use the
/// flags on the packets which get delivered via the user-supplied
/// AudioCapturerCallback interface after calling StopAsyncCapture.  When
/// asked to stop, any partially filled packet will be returned to the user, and
/// the final packet returned will always have the end-of-stream flag (kFlagsEos)
/// set on it to indicate that this is the final frame in the sequence.  If
/// there is no partially filled packet to return, the AudioCapturer will
/// synthesize an empty packet with no timestamp, and offset/length set to zero,
/// in order to deliver a packet with the end-of-stream flag set on it.  Once
/// users have seen the end-of-stream flag after calling stop, the AudioCapturer
/// has finished the stop operation and returned to synchronous operating mode.
///
/// ** Timing and Overflows **
///
/// All media packets produced by an AudioCapturer instance will have their PTS
/// field filled out with the capture time of the audio expressed as a timestamp
/// given by the reference clock timeline. Note: this timestamp is actually a
/// capture timestamp, not a presentation timestamp (it is more of a CTS than a
/// PTS) and is meant to represent the underlying system's best estimate of the
/// capture time of the first frame of audio, including all outboard and hardware
/// introduced buffering delay.  As a result, all timestamps produced by an
/// AudioCapturer should be expected to be in the past relative to 'now' on the
/// stream's reference clock timeline.
///
/// The one exception to the "everything has an explicit timestamp" rule is when
/// discarding submitted regions while operating in synchronous mode. Discarded
/// packets have no data in them, but FIDL demands that all pending
/// method-return-value callbacks be executed.  Because of this, the regions will
/// be returned to the user, but their timestamps will be set to
/// `NO_TIMESTAMP`, and their payload sizes will be set to zero.  Any
/// partially filled payload will have a valid timestamp, but a payload size
/// smaller than originally requested.  The final discarded payload (if there
/// were any to discard) will be followed by an OnEndOfStream event.
///
/// Two StreamPackets delivered by an AudioCapturer instance are 'continuous' if
/// the first frame of audio contained in the second packet was capture exactly
/// one nominal frame time after the final frame of audio in the first packet.
/// If this relationship does not hold, the second StreamPacket will have the
/// 'kFlagDiscontinuous' flag set in it's flags field.
///
/// Even though explicit timestamps are provided on every StreamPacket produced,
/// users who have very precise timing requirements are encouraged to always
/// reason about time by counting frames delivered since the last discontinuity,
/// rather than simply using the raw capture timestamps. This is because the
/// explicit timestamps written on continuous packets may have a small amount of
/// rounding error based on whether or not the units of the capture timeline
/// reference clock are divisible by the chosen audio frame rate.
///
/// Users should always expect the first StreamPacket produced by an
/// AudioCapturer to have the discontinuous flag set on it (as there is no
/// previous packet to be continuous with). Similarly, the first StreamPacket
/// after a DiscardAllPackets or a Stop/Start cycle will always be
/// discontinuous. After that, there are only two reasons that a StreamPacket
/// will ever be discontinuous:
///
/// 1. The user is operating in synchronous mode and does not supply regions to
///    be filled quickly enough.  If the next continuous frame of data has not
///    been captured by the time it needs to be purged from the source buffers,
///    an overflow has occurred and the AudioCapturer will flag the next captured
///    region as discontinuous.
/// 2. The user is operating in asynchronous mode and some internal error
///    prevents the AudioCapturer instance from capturing the next frame of audio
///    in a continuous fashion.  This might be high system load or a hardware
///    error, but in general it is something which should never normally happen.
///    In practice, however, if it does, the next produced packet will be flagged
///    as being discontinuous.
///
/// ** Synchronous vs. Asynchronous Trade-offs **
///
/// The choice of operating in synchronous vs. asynchronous mode is up to the
/// user, and depending on the user's requirements, there are some advantages and
/// disadvantages to each choice.
///
/// Synchronous mode requires only a single Zircon channel under the hood and can
/// achieve some small savings because of this.  In addition, the user has
/// complete control over the buffer management.  Users specify exactly where
/// audio will be captured to and in what order.  Because of this, if users do
/// not need to always be capturing, it is simple to stop and restart the capture
/// later (just by ceasing to supply packets, then resuming later on).  Payloads
/// do not need to be uniform in size either, clients may specify payloads of
/// whatever granularity is appropriate.
///
/// The primary downside of operating in synchronous mode is that two messages
/// will need to be sent for every packet to be captured.  One to inform the
/// AudioCapturer of the instance to capture into, and one to inform the user
/// that the packet has been captured.  This may end up increasing overhead and
/// potentially complicating client designs.
///
/// Asynchronous mode has the advantage requiring only 1/2 of the messages,
/// however, when operating in 'async' mode, AudioCapturer instances have no way
/// of knowing if a user is processing the StreamPackets being sent in a timely
/// fashion, and no way of automatically detecting an overflow condition.  Users
/// of 'async' mode should be careful to use a buffer large enough to ensure that
/// they will be able to process their data before an AudioCapturer will be
/// forced to overwrite it.
///
// ** Future Directions (aka TODOs) **
//
// + Consider adding a 'zero message' capture mode where the AudioCapturer
//   simply supplies a linear transformation and some buffer parameters (max
//   audio hold time) each time that it is started in 'async' mode, or each
//   time an internal overflow occurs in 'async' mode.  Based on this
//   information, client should know where the capture write pointer is at all
//   times as a function of the transformation removing the need to send any
//   buffer position messages.  This would reduce the operational overhead just
//   about as low as it could go, and could allow for the lowest possible
//   latency for capture clients.  OTOH - it might be better to achieve this
//   simply by allowing clients to be granted direct, exclusive access to the
//   driver level of capture if no resampling, reformatting, or sharing is
//   needed.
// + Consider providing some mechanism by which users may specify the exact
//   time at which they want to capture data.
// + Allow for more complex routing/mixing/AEC scenarios and place this under
//   the control of the policy manager.
// + Define and enforce access permissions and downsampling requirements for
//   sensitive content.  Enforce using the policy manager.
// + Consider allowing the mixer to produce compressed audio.
//

protocol AudioCapturer {
    compose StreamBufferSet;
    compose StreamSource;

    /// Sets the stream type of the stream to be delivered.  Causes the source
    /// material to be reformatted/resampled if needed in order to produce the
    /// requested stream type. Must be called before the payload buffer is
    /// established.
    SetPcmStreamType(struct {
        stream_type AudioStreamType;
    });

    /// Explicitly specifies a region of the shared payload buffer for the audio
    /// input to capture into.
    CaptureAt(struct {
        payload_buffer_id uint32;
        payload_offset uint32;
        frames uint32;
    }) -> (struct {
        captured_packet StreamPacket;
    });

    /// Places the AudioCapturer into 'async' capture mode and begin to produce
    /// packets of exactly 'frames_per_packet' number of frames each. The
    /// OnPacketProduced event (of StreamSink) will be used to inform the client
    /// of produced packets.
    StartAsyncCapture(struct {
        frames_per_packet uint32;
    });

    /// Stops capturing in 'async' capture mode and (optionally) deliver a callback
    /// that may be used by the client if explicit synchronization is needed.
    StopAsyncCapture() -> ();
    StopAsyncCaptureNoReply();

    /// Binds to the gain control for this AudioCapturer.
    BindGainControl(resource struct {
        gain_control_request server_end:fuchsia.media.audio.GainControl;
    });

    /// 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() -> (resource struct {
        reference_clock zx.handle:CLOCK;
    });

    /// Sets the reference clock that controls this capturer's playback rate. If the input
    /// parameter is a valid zx::clock, it must have READ, DUPLICATE, TRANSFER rights and
    /// refer to a clock that is both MONOTONIC and CONTINUOUS. If instead an invalid clock
    /// is passed (such as the uninitialized `zx::clock()`), this indicates that the stream
    /// will use a 'flexible' clock generated by AudioCore that tracks the audio device.
    ///
    /// `SetReferenceClock` cannot be called after the capturer payload buffer has been
    /// added. It also cannot be called a second time (even before capture).
    /// If the client wants a reference clock that is initially `CLOCK_MONOTONIC` but may
    /// diverge at some later time, they should create a clone of the monotonic clock, set
    /// this as the stream's reference clock, then rate-adjust it subsequently as needed.
    ///
    SetReferenceClock(resource struct {
        reference_clock zx.handle:<CLOCK, optional>;
    });

    /// Sets the usage of the capture stream.  This may be changed on the fly, but
    /// packets in flight may be affected by the new usage.  By default the
    /// Capturer is created with the FOREGROUND usage.
    SetUsage(struct {
        usage AudioCaptureUsage;
    });

    // ////////////////////////////////////////////////////////////////////////
    // StreamBufferSet methods
    // See stream.fidl.

    // ////////////////////////////////////////////////////////////////////////
    // StreamSource methods
    // See stream.fidl.

    // ////////////////////////////////////////////////////////////////////////
    // Methods to be deprecated
    // These methods will go away.

    /// Gets the currently configured stream type. Note: for an AudioCapturer
    /// which was just created and has not yet had its stream type explicitly
    /// set, this will retrieve the stream type -- at the time the AudioCapturer
    /// was created -- of the source (input or looped-back output) to which the
    /// AudioCapturer is bound. Even if this matches the client's desired format,
    /// `SetPcmStreamType` must still be called.
    ///
    // TODO(mpuryear): Get rid of this. Eventually, AudioCapturers will be
    // bindable to a set of inputs/outputs/renderers, so the concept of a
    // "native" stream type will go away. Mechanisms will need to be put in
    // place to allow users to enumerate the configuration of these bind-able
    // endpoints (and perhaps to exercise control over them), but it will be
    // the user of the AudioCapturer's job to specify the format they want.
    GetStreamType() -> (struct {
        stream_type StreamType;
    });
};