platform_hardware_interfaces/audio/7.0/IStreamOut.hal
Treehugger Robot 0144b4ff47 Merge "audio: Specify frame counter behavior for streams" am: 56ea2c54d7 am: 2cc5c067a3 am: a2f6cf19f8
Original change: https://android-review.googlesource.com/c/platform/hardware/interfaces/+/1623059

MUST ONLY BE SUBMITTED BY AUTOMERGER

Change-Id: I2a3b8e750f0d120d25c19a1e91567b7b85f5c117
2021-03-11 04:00:55 +00:00

394 lines
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Text

/*
* Copyright (C) 2020 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
package android.hardware.audio@7.0;
import android.hardware.audio.common@7.0;
import IStream;
import IStreamOutCallback;
import IStreamOutEventCallback;
interface IStreamOut extends IStream {
/**
* Return the audio hardware driver estimated latency in milliseconds.
*
* @return latencyMs latency in milliseconds.
*/
getLatency() generates (uint32_t latencyMs);
/**
* This method is used in situations where audio mixing is done in the
* hardware. This method serves as a direct interface with hardware,
* allowing to directly set the volume as apposed to via the framework.
* This method might produce multiple PCM outputs or hardware accelerated
* codecs, such as MP3 or AAC.
*
* Optional method
*
* @param left left channel attenuation, 1.0f is unity, 0.0f is zero.
* @param right right channel attenuation, 1.0f is unity, 0.0f is zero.
* @return retval operation completion status.
* If a volume is outside [0,1], return INVALID_ARGUMENTS
*/
setVolume(float left, float right) generates (Result retval);
/**
* Called when the metadata of the stream's source has been changed.
*
* Optional method
*
* @param sourceMetadata Description of the audio that is played by the clients.
* @return retval operation completion status.
* If any of the metadata fields contains an invalid value,
* returns INVALID_ARGUMENTS.
* If method isn't supported by the HAL returns NOT_SUPPORTED.
*/
updateSourceMetadata(SourceMetadata sourceMetadata) generates (Result retval);
/**
* Commands that can be executed on the driver writer thread.
*/
enum WriteCommand : int32_t {
WRITE,
GET_PRESENTATION_POSITION,
GET_LATENCY
};
/**
* Data structure passed back to the client via status message queue
* of 'write' operation.
*
* Possible values of 'retval' field:
* - OK, write operation was successful;
* - INVALID_ARGUMENTS, stream was not configured properly;
* - INVALID_STATE, stream is in a state that doesn't allow writes;
* - INVALID_OPERATION, retrieving presentation position isn't supported.
*/
struct WriteStatus {
Result retval;
WriteCommand replyTo; // discriminator
union Reply {
uint64_t written; // WRITE command, amount of bytes written, >= 0.
struct PresentationPosition { // same as generated by
uint64_t frames; // getPresentationPosition.
TimeSpec timeStamp;
} presentationPosition;
uint32_t latencyMs; // Same as generated by getLatency.
} reply;
};
/**
* Set up required transports for passing audio buffers to the driver.
*
* The transport consists of three message queues:
* -- command queue is used to instruct the writer thread what operation
* to perform;
* -- data queue is used for passing audio data from the client
* to the driver;
* -- status queue is used for reporting operation status
* (e.g. amount of bytes actually written or error code).
*
* The driver operates on a dedicated thread. The client must ensure that
* the thread is given an appropriate priority and assigned to correct
* scheduler and cgroup. For this purpose, the method returns the identifier
* of the driver thread.
*
* @param frameSize the size of a single frame, in bytes.
* @param framesCount the number of frames in a buffer.
* @return retval OK if both message queues were created successfully.
* INVALID_STATE if the method was already called.
* INVALID_ARGUMENTS if there was a problem setting up
* the queues.
* @return commandMQ a message queue used for passing commands.
* @return dataMQ a message queue used for passing audio data in the format
* specified at the stream opening.
* @return statusMQ a message queue used for passing status from the driver
* using WriteStatus structures.
* @return threadId identifier of the driver's dedicated thread; the caller
* may adjust the thread priority to match the priority
* of the thread that provides audio data.
*/
prepareForWriting(uint32_t frameSize, uint32_t framesCount)
generates (
Result retval,
fmq_sync<WriteCommand> commandMQ,
fmq_sync<uint8_t> dataMQ,
fmq_sync<WriteStatus> statusMQ,
int32_t threadId);
/**
* Return the number of audio frames written by the audio DSP to DAC since
* the output has exited standby.
*
* Optional method
*
* @return retval operation completion status.
* @return dspFrames number of audio frames written.
*/
getRenderPosition() generates (Result retval, uint32_t dspFrames);
/**
* Get the local time at which the next write to the audio driver will be
* presented. The units are microseconds, where the epoch is decided by the
* local audio HAL.
*
* Optional method
*
* @return retval operation completion status.
* @return timestampUs time of the next write.
*/
getNextWriteTimestamp() generates (Result retval, int64_t timestampUs);
/**
* Set the callback interface for notifying completion of non-blocking
* write and drain.
*
* Calling this function implies that all future 'write' and 'drain'
* must be non-blocking and use the callback to signal completion.
*
* 'clearCallback' method needs to be called in order to release the local
* callback proxy on the server side and thus dereference the callback
* implementation on the client side.
*
* @return retval operation completion status.
*/
setCallback(IStreamOutCallback callback) generates (Result retval);
/**
* Clears the callback previously set via 'setCallback' method.
*
* Warning: failure to call this method results in callback implementation
* on the client side being held until the HAL server termination.
*
* If no callback was previously set, the method should be a no-op
* and return OK.
*
* @return retval operation completion status: OK or NOT_SUPPORTED.
*/
clearCallback() generates (Result retval);
/**
* Set the callback interface for notifying about an output stream event.
*
* Calling this method with a null pointer will result in releasing
* the local callback proxy on the server side and thus dereference
* the callback implementation on the client side.
*
* @return retval operation completion status.
*/
setEventCallback(IStreamOutEventCallback callback)
generates (Result retval);
/**
* Returns whether HAL supports pausing and resuming of streams.
*
* @return supportsPause true if pausing is supported.
* @return supportsResume true if resume is supported.
*/
supportsPauseAndResume()
generates (bool supportsPause, bool supportsResume);
/**
* Notifies to the audio driver to stop playback however the queued buffers
* are retained by the hardware. Useful for implementing pause/resume. Empty
* implementation if not supported however must be implemented for hardware
* with non-trivial latency. In the pause state, some audio hardware may
* still be using power. Client code may consider calling 'suspend' after a
* timeout to prevent that excess power usage.
*
* Implementation of this function is mandatory for offloaded playback.
*
* @return retval operation completion status.
*/
pause() generates (Result retval);
/**
* Notifies to the audio driver to resume playback following a pause.
* Returns error INVALID_STATE if called without matching pause.
*
* Implementation of this function is mandatory for offloaded playback.
*
* @return retval operation completion status.
*/
resume() generates (Result retval);
/**
* Returns whether HAL supports draining of streams.
*
* @return supports true if draining is supported.
*/
supportsDrain() generates (bool supports);
/**
* Requests notification when data buffered by the driver/hardware has been
* played. If 'setCallback' has previously been called to enable
* non-blocking mode, then 'drain' must not block, instead it must return
* quickly and completion of the drain is notified through the callback. If
* 'setCallback' has not been called, then 'drain' must block until
* completion.
*
* If 'type' is 'ALL', the drain completes when all previously written data
* has been played.
*
* If 'type' is 'EARLY_NOTIFY', the drain completes shortly before all data
* for the current track has played to allow time for the framework to
* perform a gapless track switch.
*
* Drain must return immediately on 'stop' and 'flush' calls.
*
* Implementation of this function is mandatory for offloaded playback.
*
* @param type type of drain.
* @return retval operation completion status.
*/
drain(AudioDrain type) generates (Result retval);
/**
* Notifies to the audio driver to flush (that is, drop) the queued
* data. Stream must already be paused before calling 'flush'. For
* compressed and offload streams the frame count returned by
* 'getPresentationPosition' must reset after flush.
*
* Optional method
*
* Implementation of this function is mandatory for offloaded playback.
*
* @return retval operation completion status.
*/
flush() generates (Result retval);
/**
* Return a recent count of the number of audio frames presented to an
* external observer. This excludes frames which have been written but are
* still in the pipeline. The count must not reset to zero when a PCM output
* enters standby. For compressed and offload streams it is recommended that
* HAL resets the frame count.
*
* This method also returns the value of CLOCK_MONOTONIC as of this
* presentation count. The returned count is expected to be 'recent', but
* does not need to be the most recent possible value. However, the
* associated time must correspond to whatever count is returned.
* Example: assume that N+M frames have been presented, where M is a 'small'
* number. Then it is permissible to return N instead of N+M, and the
* timestamp must correspond to N rather than N+M. The terms 'recent' and
* 'small' are not defined. They reflect the quality of the implementation.
*
* @return retval operation completion status.
* @return frames count of presented audio frames.
* @return timeStamp associated clock time.
*/
getPresentationPosition()
generates (Result retval, uint64_t frames, TimeSpec timeStamp);
/**
* Selects a presentation for decoding from a next generation media stream
* (as defined per ETSI TS 103 190-2) and a program within the presentation.
*
* Optional method
*
* @param presentationId selected audio presentation.
* @param programId refinement for the presentation.
* @return retval operation completion status.
*/
selectPresentation(int32_t presentationId, int32_t programId)
generates (Result retval);
/**
* Returns the Dual Mono mode presentation setting.
*
* Optional method
*
* @return retval operation completion status.
* @return mode current setting of Dual Mono mode.
*/
getDualMonoMode() generates (Result retval, DualMonoMode mode);
/**
* Sets the Dual Mono mode presentation on the output device.
*
* The Dual Mono mode is generally applied to stereo audio streams
* where the left and right channels come from separate sources.
*
* Optional method
*
* @param mode selected Dual Mono mode.
* @return retval operation completion status.
*/
setDualMonoMode(DualMonoMode mode) generates (Result retval);
/**
* Returns the Audio Description Mix level in dB.
*
* The level is applied to streams incorporating a secondary Audio
* Description stream. It specifies the relative level of mixing for
* the Audio Description with a reference to the Main Audio.
*
* Optional method
*
* The value of the relative level is in the range from negative infinity
* to +48.
*
* @return retval operation completion status.
* @return leveldB the current Audio Description Mix Level in dB.
*/
getAudioDescriptionMixLevel() generates (Result retval, float leveldB);
/**
* Sets the Audio Description Mix level in dB.
*
* For streams incorporating a secondary Audio Description stream
* the relative level of mixing of the Audio Description to the Main Audio
* is controlled by this method.
*
* Optional method
*
* The value of the relative level must be in the range from negative
* infinity to +48.
*
* @param leveldB Audio Description Mix Level in dB
* @return retval operation completion status.
*/
setAudioDescriptionMixLevel(float leveldB) generates (Result retval);
/**
* Retrieves current playback rate parameters.
*
* Optional method
*
* @return retval operation completion status.
* @return playbackRate current playback parameters
*/
getPlaybackRateParameters()
generates (Result retval, PlaybackRate playbackRate);
/**
* Sets the playback rate parameters that control playback behavior.
* This is normally used when playing encoded content and decoding
* is performed in hardware. Otherwise, the framework can apply
* necessary transformations.
*
* Optional method
*
* If the HAL supports setting the playback rate, it is recommended
* to support speed and pitch values at least in the range
* from 0.5f to 2.0f, inclusive (see the definition of PlaybackRate struct).
*
* @param playbackRate playback parameters
* @return retval operation completion status.
*/
setPlaybackRateParameters(PlaybackRate playbackRate)
generates (Result retval);
};