/* * 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 commandMQ, fmq_sync dataMQ, fmq_sync 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); };