Audio: Copy HAL V6 into V7

This is an automated copy performed using copyHAL.sh script.

Bug: 142480271
Test: m
Change-Id: Ifd91cc0bb59608cd92d1d8e4e76c3abea0a8da5e
diff --git a/audio/7.0/IStreamOut.hal b/audio/7.0/IStreamOut.hal
new file mode 100644
index 0000000..208beb6
--- /dev/null
+++ b/audio/7.0/IStreamOut.hal
@@ -0,0 +1,378 @@
+/*
+ * 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);
+
+    /**
+     * 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);
+
+    /**
+     * 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 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);
+
+    /**
+     * 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);
+};