| // 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 point in time, either |
| // the configurable state or the operational state. A renderer is considered |
| // to be operational any time it has packets queued and waiting to be |
| // rendered; otherwise it is considered to be in the configurable state. When an |
| // AudioRenderer has entered the operational state of its life, any attempt to |
| // call a config method in the interface is considered to be illegal and will |
| // result in termination of the interface's connection to the audio service. |
| // |
| // If an AudioRenderer must be reconfigured, it is best practice to always call |
| // `DiscardAllPackets` on the AudioRenderer, before starting to reconfigure it. |
| // |
| |
| [FragileBase] |
| protocol AudioRenderer { |
| compose StreamBufferSet; |
| compose StreamSink; |
| |
| // Sets the type of the stream to be delivered by the client. Using this |
| // method implies that the stream encoding is AUDIO_ENCODING_LPCM. |
| SetPcmStreamType(AudioStreamType type); |
| |
| // Sets the stream type to be delivered by the client. This method is used |
| // for compressed pass-through. The media_specific field must be of type |
| // audio. |
| // NOTE: Not currently implemented. |
| SetStreamType(StreamType type); |
| |
| // Sets the units used by the presentation (media) timeline. By default, PTS |
| // units are nanoseconds (as if this were called with values of 1e9 and 1). |
| SetPtsUnits(uint32 tick_per_second_numerator, |
| uint32 tick_per_second_denominator); |
| |
| // Sets the maximum threshold (in frames) between an explicit PTS (user- |
| // provided) and an expected PTS (determined using interpolation). Beyond |
| // this threshold, a stream is no longer considered 'continuous' by the |
| // renderer. |
| // |
| // Defaults to RoundUp((AudioFPS/PTSTicksPerSec) / 2.0) / AudioFPS |
| // 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. |
| // |
| // 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. |
| // |
| SetPtsContinuityThreshold(float32 threshold_seconds); |
| |
| // Set the reference clock used to control playback rate. |
| // |
| // TODO(mpuryear): refine this type when we solidly define what a clock |
| // handle is/looks like. Also should we allow users to lock their rates to |
| // CLOCK_MONOTONIC instead of following the default (perhaps dynamic) system |
| // rate? |
| SetReferenceClock(handle reference_clock); |
| |
| // 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 'kNoTimestamp'. 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(kNoTimestamp, kNoTimestamp)`. |
| // |
| // 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, kNoTimestamp)`. |
| // |
| // 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(kNoTimestamp, 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(); |
| |
| // 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. |
| GetMinLeadTime() -> (int64 min_lead_time_nsec); |
| |
| // Binds to the gain control for this AudioRenderer. |
| BindGainControl(request<fuchsia.media.audio.GainControl> gain_control_request); |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // StreamBufferSet methods |
| // See stream.fidl. |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // StreamSink methods |
| // See stream.fidl. |
| |
| // 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). |
| }; |
