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+/*
+ * Copyright (C) 2007 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.
+ */
+
+#ifndef ANDROID_AUDIOTRACK_H
+#define ANDROID_AUDIOTRACK_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <media/IAudioFlinger.h>
+#include <media/IAudioTrack.h>
+#include <media/AudioSystem.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <binder/IMemory.h>
+#include <utils/threads.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class audio_track_cblk_t;
+
+// ----------------------------------------------------------------------------
+
+class AudioTrack : virtual public RefBase
+{
+public:
+ enum channel_index {
+ MONO = 0,
+ LEFT = 0,
+ RIGHT = 1
+ };
+
+ /* Events used by AudioTrack callback function (audio_track_cblk_t).
+ */
+ enum event_type {
+ EVENT_MORE_DATA = 0, // Request to write more data to PCM buffer.
+ EVENT_UNDERRUN = 1, // PCM buffer underrun occured.
+ EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from loop start if loop count was not 0.
+ EVENT_MARKER = 3, // Playback head is at the specified marker position (See setMarkerPosition()).
+ EVENT_NEW_POS = 4, // Playback head is at a new position (See setPositionUpdatePeriod()).
+ EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer.
+ };
+
+ /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+ * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
+ */
+
+ class Buffer
+ {
+ public:
+ enum {
+ MUTE = 0x00000001
+ };
+ uint32_t flags; // 0 or MUTE
+ audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
+ // accessed directly by WebKit ANP callback
+ int channelCount; // will be removed in the future, do not use
+
+ size_t frameCount; // number of sample frames corresponding to size;
+ // on input it is the number of frames desired,
+ // on output is the number of frames actually filled
+
+ size_t size; // input/output in byte units
+ union {
+ void* raw;
+ short* i16; // signed 16-bit
+ int8_t* i8; // unsigned 8-bit, offset by 0x80
+ };
+ };
+
+
+ /* As a convenience, if a callback is supplied, a handler thread
+ * is automatically created with the appropriate priority. This thread
+ * invokes the callback when a new buffer becomes available or various conditions occur.
+ * Parameters:
+ *
+ * event: type of event notified (see enum AudioTrack::event_type).
+ * user: Pointer to context for use by the callback receiver.
+ * info: Pointer to optional parameter according to event type:
+ * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
+ * more bytes than indicated by 'size' field and update 'size' if fewer bytes are
+ * written.
+ * - EVENT_UNDERRUN: unused.
+ * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
+ * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
+ * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
+ * - EVENT_BUFFER_END: unused.
+ */
+
+ typedef void (*callback_t)(int event, void* user, void *info);
+
+ /* Returns the minimum frame count required for the successful creation of
+ * an AudioTrack object.
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - NO_INIT: audio server or audio hardware not initialized
+ */
+
+ static status_t getMinFrameCount(int* frameCount,
+ audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
+ uint32_t sampleRate = 0);
+
+ /* Constructs an uninitialized AudioTrack. No connection with
+ * AudioFlinger takes place.
+ */
+ AudioTrack();
+
+ /* Creates an audio track and registers it with AudioFlinger.
+ * Once created, the track needs to be started before it can be used.
+ * Unspecified values are set to the audio hardware's current
+ * values.
+ *
+ * Parameters:
+ *
+ * streamType: Select the type of audio stream this track is attached to
+ * (e.g. AUDIO_STREAM_MUSIC).
+ * sampleRate: Track sampling rate in Hz.
+ * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
+ * 16 bits per sample).
+ * channelMask: Channel mask: see audio_channels_t.
+ * frameCount: Minimum size of track PCM buffer in frames. This defines the
+ * latency of the track. The actual size selected by the AudioTrack could be
+ * larger if the requested size is not compatible with current audio HAL
+ * latency.
+ * flags: Reserved for future use.
+ * cbf: Callback function. If not null, this function is called periodically
+ * to request new PCM data.
+ * user: Context for use by the callback receiver.
+ * notificationFrames: The callback function is called each time notificationFrames PCM
+ * frames have been consumed from track input buffer.
+ * sessionId: Specific session ID, or zero to use default.
+ */
+
+ AudioTrack( audio_stream_type_t streamType,
+ uint32_t sampleRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ int channelMask = 0,
+ int frameCount = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ int notificationFrames = 0,
+ int sessionId = 0);
+
+ // DEPRECATED
+ explicit AudioTrack( int streamType,
+ uint32_t sampleRate = 0,
+ int format = AUDIO_FORMAT_DEFAULT,
+ int channelMask = 0,
+ int frameCount = 0,
+ uint32_t flags = (uint32_t) AUDIO_POLICY_OUTPUT_FLAG_NONE,
+ callback_t cbf = 0,
+ void* user = 0,
+ int notificationFrames = 0,
+ int sessionId = 0);
+
+ /* Creates an audio track and registers it with AudioFlinger. With this constructor,
+ * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
+ * identified by the argument sharedBuffer. This prototype is for static buffer playback.
+ * PCM data must be present in memory before the AudioTrack is started.
+ * The write() and flush() methods are not supported in this case.
+ * It is recommended to pass a callback function to be notified of playback end by an
+ * EVENT_UNDERRUN event.
+ */
+
+ AudioTrack( audio_stream_type_t streamType,
+ uint32_t sampleRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ int channelMask = 0,
+ const sp<IMemory>& sharedBuffer = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ int notificationFrames = 0,
+ int sessionId = 0);
+
+ /* Terminates the AudioTrack and unregisters it from AudioFlinger.
+ * Also destroys all resources associated with the AudioTrack.
+ */
+ ~AudioTrack();
+
+
+ /* Initialize an uninitialized AudioTrack.
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful initialization
+ * - INVALID_OPERATION: AudioTrack is already initialized
+ * - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+ * - NO_INIT: audio server or audio hardware not initialized
+ * */
+ status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
+ uint32_t sampleRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ int channelMask = 0,
+ int frameCount = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ int notificationFrames = 0,
+ const sp<IMemory>& sharedBuffer = 0,
+ bool threadCanCallJava = false,
+ int sessionId = 0);
+
+
+ /* Result of constructing the AudioTrack. This must be checked
+ * before using any AudioTrack API (except for set()), because using
+ * an uninitialized AudioTrack produces undefined results.
+ * See set() method above for possible return codes.
+ */
+ status_t initCheck() const;
+
+ /* Returns this track's estimated latency in milliseconds.
+ * This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
+ * and audio hardware driver.
+ */
+ uint32_t latency() const;
+
+ /* getters, see constructors and set() */
+
+ audio_stream_type_t streamType() const;
+ audio_format_t format() const;
+ int channelCount() const;
+ uint32_t frameCount() const;
+
+ /* Return channelCount * (bit depth per channel / 8).
+ * channelCount is determined from channelMask, and bit depth comes from format.
+ */
+ size_t frameSize() const;
+
+ sp<IMemory>& sharedBuffer();
+
+
+ /* After it's created the track is not active. Call start() to
+ * make it active. If set, the callback will start being called.
+ */
+ void start();
+
+ /* Stop a track. If set, the callback will cease being called and
+ * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
+ * and will fill up buffers until the pool is exhausted.
+ */
+ void stop();
+ bool stopped() const;
+
+ /* Flush a stopped track. All pending buffers are discarded.
+ * This function has no effect if the track is not stopped.
+ */
+ void flush();
+
+ /* Pause a track. If set, the callback will cease being called and
+ * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
+ * and will fill up buffers until the pool is exhausted.
+ */
+ void pause();
+
+ /* Mute or unmute this track.
+ * While muted, the callback, if set, is still called.
+ */
+ void mute(bool);
+ bool muted() const;
+
+ /* Set volume for this track, mostly used for games' sound effects
+ * left and right volumes. Levels must be >= 0.0 and <= 1.0.
+ */
+ status_t setVolume(float left, float right);
+ void getVolume(float* left, float* right) const;
+
+ /* Set the send level for this track. An auxiliary effect should be attached
+ * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
+ */
+ status_t setAuxEffectSendLevel(float level);
+ void getAuxEffectSendLevel(float* level) const;
+
+ /* Set sample rate for this track, mostly used for games' sound effects
+ */
+ status_t setSampleRate(int sampleRate);
+ uint32_t getSampleRate() const;
+
+ /* Enables looping and sets the start and end points of looping.
+ *
+ * Parameters:
+ *
+ * loopStart: loop start expressed as the number of PCM frames played since AudioTrack start.
+ * loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start.
+ * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
+ * pending or active loop. loopCount = -1 means infinite looping.
+ *
+ * For proper operation the following condition must be respected:
+ * (loopEnd-loopStart) <= framecount()
+ */
+ status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
+
+ /* Sets marker position. When playback reaches the number of frames specified, a callback with
+ * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
+ * notification callback.
+ * If the AudioTrack has been opened with no callback function associated, the operation will fail.
+ *
+ * Parameters:
+ *
+ * marker: marker position expressed in frames.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - INVALID_OPERATION: the AudioTrack has no callback installed.
+ */
+ status_t setMarkerPosition(uint32_t marker);
+ status_t getMarkerPosition(uint32_t *marker) const;
+
+
+ /* Sets position update period. Every time the number of frames specified has been played,
+ * a callback with event type EVENT_NEW_POS is called.
+ * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
+ * callback.
+ * If the AudioTrack has been opened with no callback function associated, the operation will fail.
+ *
+ * Parameters:
+ *
+ * updatePeriod: position update notification period expressed in frames.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - INVALID_OPERATION: the AudioTrack has no callback installed.
+ */
+ status_t setPositionUpdatePeriod(uint32_t updatePeriod);
+ status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
+
+ /* Sets playback head position within AudioTrack buffer. The new position is specified
+ * in number of frames.
+ * This method must be called with the AudioTrack in paused or stopped state.
+ * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
+ * Therefore using this method makes sense only when playing a "static" audio buffer
+ * as opposed to streaming.
+ * The getPosition() method on the other hand returns the total number of frames played since
+ * playback start.
+ *
+ * Parameters:
+ *
+ * position: New playback head position within AudioTrack buffer.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - INVALID_OPERATION: the AudioTrack is not stopped.
+ * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
+ */
+ status_t setPosition(uint32_t position);
+ status_t getPosition(uint32_t *position);
+
+ /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
+ * rewriting the buffer before restarting playback after a stop.
+ * This method must be called with the AudioTrack in paused or stopped state.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - INVALID_OPERATION: the AudioTrack is not stopped.
+ */
+ status_t reload();
+
+ /* Returns a handle on the audio output used by this AudioTrack.
+ *
+ * Parameters:
+ * none.
+ *
+ * Returned value:
+ * handle on audio hardware output
+ */
+ audio_io_handle_t getOutput();
+
+ /* Returns the unique session ID associated with this track.
+ *
+ * Parameters:
+ * none.
+ *
+ * Returned value:
+ * AudioTrack session ID.
+ */
+ int getSessionId() const;
+
+ /* Attach track auxiliary output to specified effect. Use effectId = 0
+ * to detach track from effect.
+ *
+ * Parameters:
+ *
+ * effectId: effectId obtained from AudioEffect::id().
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - INVALID_OPERATION: the effect is not an auxiliary effect.
+ * - BAD_VALUE: The specified effect ID is invalid
+ */
+ status_t attachAuxEffect(int effectId);
+
+ /* Obtains a buffer of "frameCount" frames. The buffer must be
+ * filled entirely, and then released with releaseBuffer().
+ * If the track is stopped, obtainBuffer() returns
+ * STOPPED instead of NO_ERROR as long as there are buffers available,
+ * at which point NO_MORE_BUFFERS is returned.
+ * Buffers will be returned until the pool (buffercount())
+ * is exhausted, at which point obtainBuffer() will either block
+ * or return WOULD_BLOCK depending on the value of the "blocking"
+ * parameter.
+ *
+ * Interpretation of waitCount:
+ * +n limits wait time to n * WAIT_PERIOD_MS,
+ * -1 causes an (almost) infinite wait time,
+ * 0 non-blocking.
+ */
+
+ enum {
+ NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
+ STOPPED = 1
+ };
+
+ status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+
+ /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
+ void releaseBuffer(Buffer* audioBuffer);
+
+ /* As a convenience we provide a write() interface to the audio buffer.
+ * This is implemented on top of obtainBuffer/releaseBuffer. For best
+ * performance use callbacks. Returns actual number of bytes written >= 0,
+ * or one of the following negative status codes:
+ * INVALID_OPERATION AudioTrack is configured for shared buffer mode
+ * BAD_VALUE size is invalid
+ * STOPPED AudioTrack was stopped during the write
+ * NO_MORE_BUFFERS when obtainBuffer() returns same
+ * or any other error code returned by IAudioTrack::start() or restoreTrack_l().
+ */
+ ssize_t write(const void* buffer, size_t size);
+
+ /*
+ * Dumps the state of an audio track.
+ */
+ status_t dump(int fd, const Vector<String16>& args) const;
+
+protected:
+ /* copying audio tracks is not allowed */
+ AudioTrack(const AudioTrack& other);
+ AudioTrack& operator = (const AudioTrack& other);
+
+ /* a small internal class to handle the callback */
+ class AudioTrackThread : public Thread
+ {
+ public:
+ AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false);
+ private:
+ friend class AudioTrack;
+ virtual bool threadLoop();
+ virtual status_t readyToRun();
+ virtual void onFirstRef();
+ AudioTrack& mReceiver;
+ };
+
+ // body of AudioTrackThread::threadLoop()
+ bool processAudioBuffer(const sp<AudioTrackThread>& thread);
+
+ status_t createTrack_l(audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ audio_policy_output_flags_t flags,
+ const sp<IMemory>& sharedBuffer,
+ audio_io_handle_t output);
+ void flush_l();
+ status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
+ audio_io_handle_t getOutput_l();
+ status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
+ bool stopped_l() const { return !mActive; }
+
+ sp<IAudioTrack> mAudioTrack;
+ sp<IMemory> mCblkMemory;
+ sp<AudioTrackThread> mAudioTrackThread;
+
+ float mVolume[2];
+ float mSendLevel;
+ uint32_t mFrameCount;
+
+ audio_track_cblk_t* mCblk;
+ audio_format_t mFormat;
+ audio_stream_type_t mStreamType;
+ uint8_t mChannelCount;
+ uint8_t mMuted;
+ uint8_t mReserved;
+ uint32_t mChannelMask;
+ status_t mStatus;
+ uint32_t mLatency;
+
+ bool mActive; // protected by mLock
+
+ callback_t mCbf; // callback handler for events, or NULL
+ void* mUserData;
+ uint32_t mNotificationFramesReq; // requested number of frames between each notification callback
+ uint32_t mNotificationFramesAct; // actual number of frames between each notification callback
+ sp<IMemory> mSharedBuffer;
+ int mLoopCount;
+ uint32_t mRemainingFrames;
+ uint32_t mMarkerPosition;
+ bool mMarkerReached;
+ uint32_t mNewPosition;
+ uint32_t mUpdatePeriod;
+ bool mFlushed; // FIXME will be made obsolete by making flush() synchronous
+ audio_policy_output_flags_t mFlags;
+ int mSessionId;
+ int mAuxEffectId;
+ mutable Mutex mLock;
+ status_t mRestoreStatus;
+ bool mIsTimed;
+ int mPreviousPriority; // before start()
+ int mPreviousSchedulingGroup;
+};
+
+class TimedAudioTrack : public AudioTrack
+{
+public:
+ TimedAudioTrack();
+
+ /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
+ status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
+
+ /* queue a buffer obtained via allocateTimedBuffer for playback at the
+ given timestamp. PTS units a microseconds on the media time timeline.
+ The media time transform (set with setMediaTimeTransform) set by the
+ audio producer will handle converting from media time to local time
+ (perhaps going through the common time timeline in the case of
+ synchronized multiroom audio case) */
+ status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+
+ /* define a transform between media time and either common time or
+ local time */
+ enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TargetTimeline target);
+};
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOTRACK_H