b6a8851f30
This adds the following method: IStreamOut.selectPresentation This corresponds to the following legacy parameters: AUDIO_PARAMETER_STREAM_PRESENTATION_ID AUDIO_PARAMETER_STREAM_PROGRAM_ID Bug: 63901775 Test: make Change-Id: I9ca6ead72b1ef80d2de582a6e4b051ee32fe1857
279 lines
11 KiB
Text
279 lines
11 KiB
Text
/*
|
|
* Copyright (C) 2018 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@4.0;
|
|
|
|
import android.hardware.audio.common@4.0;
|
|
import IStream;
|
|
import IStreamOutCallback;
|
|
|
|
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);
|
|
|
|
/**
|
|
* 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;
|
|
};
|
|
|
|
/**
|
|
* Called when the metadata of the stream's source has been changed.
|
|
* @param sourceMetadata Description of the audio that is played by the clients.
|
|
*/
|
|
updateSourceMetadata(SourceMetadata sourceMetadata);
|
|
|
|
/**
|
|
* 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 identifiers
|
|
* 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 threadInfo identifiers of the driver's dedicated thread.
|
|
*/
|
|
prepareForWriting(uint32_t frameSize, uint32_t framesCount)
|
|
generates (
|
|
Result retval,
|
|
fmq_sync<WriteCommand> commandMQ,
|
|
fmq_sync<uint8_t> dataMQ,
|
|
fmq_sync<WriteStatus> statusMQ,
|
|
ThreadInfo threadInfo);
|
|
|
|
/**
|
|
* 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);
|
|
|
|
/**
|
|
* 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 the queued data. Stream must
|
|
* already be paused before calling '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 is not reset to zero when output enters
|
|
* standby. 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.
|
|
*
|
|
* Optional method
|
|
*
|
|
* @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);
|
|
};
|