| /* |
| * 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 <cutils/sched_policy.h> |
| #include <media/AudioSystem.h> |
| #include <media/AudioTimestamp.h> |
| #include <media/IAudioTrack.h> |
| #include <media/AudioResamplerPublic.h> |
| #include <media/Modulo.h> |
| #include <utils/threads.h> |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| |
| struct audio_track_cblk_t; |
| class AudioTrackClientProxy; |
| class StaticAudioTrackClientProxy; |
| |
| // ---------------------------------------------------------------------------- |
| |
| class AudioTrack : public RefBase |
| { |
| public: |
| |
| /* Events used by AudioTrack callback function (callback_t). |
| * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. |
| */ |
| enum event_type { |
| EVENT_MORE_DATA = 0, // Request to write more data to buffer. |
| // This event only occurs for TRANSFER_CALLBACK. |
| // If this event is delivered but the callback handler |
| // does not want to write more data, the handler must |
| // ignore the event by setting frameCount to zero. |
| // This might occur, for example, if the application is |
| // waiting for source data or is at the end of stream. |
| // |
| // For data filling, it is preferred that the callback |
| // does not block and instead returns a short count on |
| // the amount of data actually delivered |
| // (or 0, if no data is currently available). |
| EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for |
| // static tracks. |
| EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from |
| // loop start if loop count was not 0 for a static track. |
| 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 has completed for a static track. |
| EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and |
| // voluntary invalidation by mediaserver, or mediaserver crash. |
| EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played |
| // back (after stop is called) for an offloaded track. |
| #if 0 // FIXME not yet implemented |
| EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change |
| // in the mapping from frame position to presentation time. |
| // See AudioTimestamp for the information included with event. |
| #endif |
| }; |
| |
| /* Client should declare a Buffer and pass the address to obtainBuffer() |
| * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. |
| */ |
| |
| class Buffer |
| { |
| public: |
| // FIXME use m prefix |
| size_t frameCount; // number of sample frames corresponding to size; |
| // on input to obtainBuffer() it is the number of frames desired, |
| // on output from obtainBuffer() it is the number of available |
| // [empty slots for] frames to be filled |
| // on input to releaseBuffer() it is currently ignored |
| |
| size_t size; // input/output in bytes == frameCount * frameSize |
| // on input to obtainBuffer() it is ignored |
| // on output from obtainBuffer() it is the number of available |
| // [empty slots for] bytes to be filled, |
| // which is frameCount * frameSize |
| // on input to releaseBuffer() it is the number of bytes to |
| // release |
| // FIXME This is redundant with respect to frameCount. Consider |
| // removing size and making frameCount the primary field. |
| |
| union { |
| void* raw; |
| short* i16; // signed 16-bit |
| int8_t* i8; // unsigned 8-bit, offset by 0x80 |
| }; // input to obtainBuffer(): unused, output: pointer to buffer |
| }; |
| |
| /* 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 const uint32_t containing the marker position in frames. |
| * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. |
| * - EVENT_BUFFER_END: unused. |
| * - EVENT_NEW_IAUDIOTRACK: unused. |
| * - EVENT_STREAM_END: unused. |
| * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. |
| */ |
| |
| 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 |
| * - BAD_VALUE: unsupported configuration |
| * frameCount is guaranteed to be non-zero if status is NO_ERROR, |
| * and is undefined otherwise. |
| * FIXME This API assumes a route, and so should be deprecated. |
| */ |
| |
| static status_t getMinFrameCount(size_t* frameCount, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate); |
| |
| /* How data is transferred to AudioTrack |
| */ |
| enum transfer_type { |
| TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters |
| TRANSFER_CALLBACK, // callback EVENT_MORE_DATA |
| TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() |
| TRANSFER_SYNC, // synchronous write() |
| TRANSFER_SHARED, // shared memory |
| }; |
| |
| /* Constructs an uninitialized AudioTrack. No connection with |
| * AudioFlinger takes place. Use set() after this. |
| */ |
| AudioTrack(); |
| |
| /* Creates an AudioTrack object and registers it with AudioFlinger. |
| * Once created, the track needs to be started before it can be used. |
| * Unspecified values are set to appropriate default values. |
| * |
| * Parameters: |
| * |
| * streamType: Select the type of audio stream this track is attached to |
| * (e.g. AUDIO_STREAM_MUSIC). |
| * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. |
| * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. |
| * 0 will not work with current policy implementation for direct output |
| * selection where an exact match is needed for sampling rate. |
| * format: Audio format. For mixed tracks, any PCM format supported by server is OK. |
| * For direct and offloaded tracks, the possible format(s) depends on the |
| * output sink. |
| * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. |
| * frameCount: Minimum size of track PCM buffer in frames. This defines the |
| * application's contribution to 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 |
| * configuration. Zero means to use a default value. |
| * flags: See comments on audio_output_flags_t in <system/audio.h>. |
| * cbf: Callback function. If not null, this function is called periodically |
| * to provide new data in TRANSFER_CALLBACK mode |
| * and inform of marker, position updates, etc. |
| * 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 by server. |
| * Zero means to use a default value, which is typically: |
| * - fast tracks: HAL buffer size, even if track frameCount is larger |
| * - normal tracks: 1/2 of track frameCount |
| * A positive value means that many frames at initial source sample rate. |
| * A negative value for this parameter specifies the negative of the |
| * requested number of notifications (sub-buffers) in the entire buffer. |
| * For fast tracks, the FastMixer will process one sub-buffer at a time. |
| * The size of each sub-buffer is determined by the HAL. |
| * To get "double buffering", for example, one should pass -2. |
| * The minimum number of sub-buffers is 1 (expressed as -1), |
| * and the maximum number of sub-buffers is 8 (expressed as -8). |
| * Negative is only permitted for fast tracks, and if frameCount is zero. |
| * TODO It is ugly to overload a parameter in this way depending on |
| * whether it is positive, negative, or zero. Consider splitting apart. |
| * sessionId: Specific session ID, or zero to use default. |
| * transferType: How data is transferred to AudioTrack. |
| * offloadInfo: If not NULL, provides offload parameters for |
| * AudioSystem::getOutputForAttr(). |
| * uid: User ID of the app which initially requested this AudioTrack |
| * for power management tracking, or -1 for current user ID. |
| * pid: Process ID of the app which initially requested this AudioTrack |
| * for power management tracking, or -1 for current process ID. |
| * pAttributes: If not NULL, supersedes streamType for use case selection. |
| * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack |
| binder to AudioFlinger. |
| It will return an error instead. The application will recreate |
| the track based on offloading or different channel configuration, etc. |
| * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow |
| * maxRequiredSpeed playback. Values less than 1.0f and greater than |
| * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks |
| * and direct or offloaded tracks, this parameter is ignored. |
| * threadCanCallJava: Not present in parameter list, and so is fixed at false. |
| */ |
| |
| AudioTrack( audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount = 0, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| int32_t notificationFrames = 0, |
| audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, |
| transfer_type transferType = TRANSFER_DEFAULT, |
| const audio_offload_info_t *offloadInfo = NULL, |
| uid_t uid = AUDIO_UID_INVALID, |
| pid_t pid = -1, |
| const audio_attributes_t* pAttributes = NULL, |
| bool doNotReconnect = false, |
| float maxRequiredSpeed = 1.0f); |
| |
| /* Creates an audio track and registers it with AudioFlinger. |
| * With this constructor, the track is configured for static buffer mode. |
| * Data to be rendered is passed in a shared memory buffer |
| * identified by the argument sharedBuffer, which should be non-0. |
| * If sharedBuffer is zero, this constructor is equivalent to the previous constructor |
| * but without the ability to specify a non-zero value for the frameCount parameter. |
| * The memory should be initialized to the desired data before calling start(). |
| * The write() method is 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, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| const sp<IMemory>& sharedBuffer, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| int32_t notificationFrames = 0, |
| audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, |
| transfer_type transferType = TRANSFER_DEFAULT, |
| const audio_offload_info_t *offloadInfo = NULL, |
| uid_t uid = AUDIO_UID_INVALID, |
| pid_t pid = -1, |
| const audio_attributes_t* pAttributes = NULL, |
| bool doNotReconnect = false, |
| float maxRequiredSpeed = 1.0f); |
| |
| /* Terminates the AudioTrack and unregisters it from AudioFlinger. |
| * Also destroys all resources associated with the AudioTrack. |
| */ |
| protected: |
| virtual ~AudioTrack(); |
| public: |
| |
| /* Initialize an AudioTrack that was created using the AudioTrack() constructor. |
| * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. |
| * set() is not multi-thread safe. |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful initialization |
| * - INVALID_OPERATION: AudioTrack is already initialized |
| * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) |
| * - NO_INIT: audio server or audio hardware not initialized |
| * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. |
| * If sharedBuffer is non-0, the frameCount parameter is ignored and |
| * replaced by the shared buffer's total allocated size in frame units. |
| * |
| * Parameters not listed in the AudioTrack constructors above: |
| * |
| * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. |
| * |
| * Internal state post condition: |
| * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes |
| */ |
| status_t set(audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount = 0, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| int32_t notificationFrames = 0, |
| const sp<IMemory>& sharedBuffer = 0, |
| bool threadCanCallJava = false, |
| audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, |
| transfer_type transferType = TRANSFER_DEFAULT, |
| const audio_offload_info_t *offloadInfo = NULL, |
| uid_t uid = AUDIO_UID_INVALID, |
| pid_t pid = -1, |
| const audio_attributes_t* pAttributes = NULL, |
| bool doNotReconnect = false, |
| float maxRequiredSpeed = 1.0f); |
| |
| /* Result of constructing the AudioTrack. This must be checked for successful initialization |
| * 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 { return mStatus; } |
| |
| /* 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 { return mLatency; } |
| |
| /* Returns the number of application-level buffer underruns |
| * since the AudioTrack was created. |
| */ |
| uint32_t getUnderrunCount() const; |
| |
| /* getters, see constructors and set() */ |
| |
| audio_stream_type_t streamType() const; |
| audio_format_t format() const { return mFormat; } |
| |
| /* Return frame size in bytes, which for linear PCM is |
| * channelCount * (bit depth per channel / 8). |
| * channelCount is determined from channelMask, and bit depth comes from format. |
| * For non-linear formats, the frame size is typically 1 byte. |
| */ |
| size_t frameSize() const { return mFrameSize; } |
| |
| uint32_t channelCount() const { return mChannelCount; } |
| size_t frameCount() const { return mFrameCount; } |
| |
| /* |
| * Return the period of the notification callback in frames. |
| * This value is set when the AudioTrack is constructed. |
| * It can be modified if the AudioTrack is rerouted. |
| */ |
| uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } |
| |
| /* Return effective size of audio buffer that an application writes to |
| * or a negative error if the track is uninitialized. |
| */ |
| ssize_t getBufferSizeInFrames(); |
| |
| /* Returns the buffer duration in microseconds at current playback rate. |
| */ |
| status_t getBufferDurationInUs(int64_t *duration); |
| |
| /* Set the effective size of audio buffer that an application writes to. |
| * This is used to determine the amount of available room in the buffer, |
| * which determines when a write will block. |
| * This allows an application to raise and lower the audio latency. |
| * The requested size may be adjusted so that it is |
| * greater or equal to the absolute minimum and |
| * less than or equal to the getBufferCapacityInFrames(). |
| * It may also be adjusted slightly for internal reasons. |
| * |
| * Return the final size or a negative error if the track is unitialized |
| * or does not support variable sizes. |
| */ |
| ssize_t setBufferSizeInFrames(size_t size); |
| |
| /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ |
| sp<IMemory> sharedBuffer() const { return mSharedBuffer; } |
| |
| /* After it's created the track is not active. Call start() to |
| * make it active. If set, the callback will start being called. |
| * If the track was previously paused, volume is ramped up over the first mix buffer. |
| */ |
| status_t start(); |
| |
| /* Stop a track. |
| * In static buffer mode, the track is stopped immediately. |
| * In streaming mode, the callback will cease being called. Note that obtainBuffer() still |
| * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. |
| * In streaming mode the stop does not occur immediately: any data remaining in the buffer |
| * is first drained, mixed, and output, and only then is the track marked as stopped. |
| */ |
| void stop(); |
| bool stopped() const; |
| |
| /* Flush a stopped or paused track. All previously buffered data is discarded immediately. |
| * This has the effect of draining the buffers without mixing or output. |
| * Flush is intended for streaming mode, for example before switching to non-contiguous content. |
| * This function is a no-op if the track is not stopped or paused, or uses a static buffer. |
| */ |
| void flush(); |
| |
| /* Pause a track. After pause, the callback will cease being called and |
| * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works |
| * and will fill up buffers until the pool is exhausted. |
| * Volume is ramped down over the next mix buffer following the pause request, |
| * and then the track is marked as paused. It can be resumed with ramp up by start(). |
| */ |
| void pause(); |
| |
| /* Set volume for this track, mostly used for games' sound effects |
| * left and right volumes. Levels must be >= 0.0 and <= 1.0. |
| * This is the older API. New applications should use setVolume(float) when possible. |
| */ |
| status_t setVolume(float left, float right); |
| |
| /* Set volume for all channels. This is the preferred API for new applications, |
| * especially for multi-channel content. |
| */ |
| status_t setVolume(float volume); |
| |
| /* 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 source sample rate for this track in Hz, mostly used for games' sound effects. |
| * Zero is not permitted. |
| */ |
| status_t setSampleRate(uint32_t sampleRate); |
| |
| /* Return current source sample rate in Hz. |
| * If specified as zero in constructor or set(), this will be the sink sample rate. |
| */ |
| uint32_t getSampleRate() const; |
| |
| /* Return the original source sample rate in Hz. This corresponds to the sample rate |
| * if playback rate had normal speed and pitch. |
| */ |
| uint32_t getOriginalSampleRate() const; |
| |
| /* Set source playback rate for timestretch |
| * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster |
| * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch |
| * |
| * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX |
| * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX |
| * |
| * Speed increases the playback rate of media, but does not alter pitch. |
| * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. |
| */ |
| status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); |
| |
| /* Return current playback rate */ |
| const AudioPlaybackRate& getPlaybackRate() const; |
| |
| /* Enables looping and sets the start and end points of looping. |
| * Only supported for static buffer mode. |
| * |
| * Parameters: |
| * |
| * loopStart: loop start in frames relative to start of buffer. |
| * loopEnd: loop end in frames relative to start of buffer. |
| * 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: |
| * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). |
| * |
| * If the loop period (loopEnd - loopStart) is too small for the implementation to support, |
| * setLoop() will return BAD_VALUE. loopCount must be >= -1. |
| * |
| */ |
| 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. To set a marker at a position which would compute as 0, |
| * a workaround is to set the marker at a nearby position such as ~0 or 1. |
| * If the AudioTrack has been opened with no callback function associated, the operation will |
| * fail. |
| * |
| * Parameters: |
| * |
| * marker: marker position expressed in wrapping (overflow) frame units, |
| * like the return value of getPosition(). |
| * |
| * 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. |
| * Extremely small values may be rounded up to a value the implementation can support. |
| * |
| * 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. |
| * Only supported for static buffer mode. |
| * |
| * Parameters: |
| * |
| * position: New playback head position in frames relative to start of buffer. |
| * 0 <= position <= frameCount(). Note that end of buffer is permitted, |
| * but will result in an immediate underrun if started. |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. |
| * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack |
| * buffer |
| */ |
| status_t setPosition(uint32_t position); |
| |
| /* Return the total number of frames played since playback start. |
| * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. |
| * It is reset to zero by flush(), reload(), and stop(). |
| * |
| * Parameters: |
| * |
| * position: Address where to return play head position. |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - BAD_VALUE: position is NULL |
| */ |
| status_t getPosition(uint32_t *position); |
| |
| /* For static buffer mode only, this returns the current playback position in frames |
| * relative to start of buffer. It is analogous to the position units used by |
| * setLoop() and setPosition(). After underrun, the position will be at end of buffer. |
| */ |
| status_t getBufferPosition(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. |
| * Not allowed in streaming mode. |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. |
| */ |
| status_t reload(); |
| |
| /* Returns a handle on the audio output used by this AudioTrack. |
| * |
| * Parameters: |
| * none. |
| * |
| * Returned value: |
| * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the |
| * track needed to be re-created but that failed |
| */ |
| private: |
| audio_io_handle_t getOutput() const; |
| public: |
| |
| /* Selects the audio device to use for output of this AudioTrack. A value of |
| * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. |
| * |
| * Parameters: |
| * The device ID of the selected device (as returned by the AudioDevicesManager API). |
| * |
| * Returned value: |
| * - NO_ERROR: successful operation |
| * TODO: what else can happen here? |
| */ |
| status_t setOutputDevice(audio_port_handle_t deviceId); |
| |
| /* Returns the ID of the audio device selected for this AudioTrack. |
| * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. |
| * |
| * Parameters: |
| * none. |
| */ |
| audio_port_handle_t getOutputDevice(); |
| |
| /* Returns the ID of the audio device actually used by the output to which this AudioTrack is |
| * attached. |
| * A value of AUDIO_PORT_HANDLE_NONE indicates the audio track is not attached to any output. |
| * |
| * Parameters: |
| * none. |
| */ |
| audio_port_handle_t getRoutedDeviceId(); |
| |
| /* Returns the unique session ID associated with this track. |
| * |
| * Parameters: |
| * none. |
| * |
| * Returned value: |
| * AudioTrack session ID. |
| */ |
| audio_session_t getSessionId() const { return mSessionId; } |
| |
| /* 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); |
| |
| /* Public API for TRANSFER_OBTAIN mode. |
| * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. |
| * After filling these slots with data, the caller should release them with releaseBuffer(). |
| * If the track buffer is not full, obtainBuffer() returns as many contiguous |
| * [empty slots for] frames as are available immediately. |
| * |
| * If nonContig is non-NULL, it is an output parameter that will be set to the number of |
| * additional non-contiguous frames that are predicted to be available immediately, |
| * if the client were to release the first frames and then call obtainBuffer() again. |
| * This value is only a prediction, and needs to be confirmed. |
| * It will be set to zero for an error return. |
| * |
| * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK |
| * regardless of the value of waitCount. |
| * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a |
| * maximum timeout based on waitCount; see chart below. |
| * Buffers will be returned until the pool |
| * is exhausted, at which point obtainBuffer() will either block |
| * or return WOULD_BLOCK depending on the value of the "waitCount" |
| * parameter. |
| * |
| * Interpretation of waitCount: |
| * +n limits wait time to n * WAIT_PERIOD_MS, |
| * -1 causes an (almost) infinite wait time, |
| * 0 non-blocking. |
| * |
| * Buffer fields |
| * On entry: |
| * frameCount number of [empty slots for] frames requested |
| * size ignored |
| * raw ignored |
| * After error return: |
| * frameCount 0 |
| * size 0 |
| * raw undefined |
| * After successful return: |
| * frameCount actual number of [empty slots for] frames available, <= number requested |
| * size actual number of bytes available |
| * raw pointer to the buffer |
| */ |
| status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, |
| size_t *nonContig = NULL); |
| |
| private: |
| /* If nonContig is non-NULL, it is an output parameter that will be set to the number of |
| * additional non-contiguous frames that are predicted to be available immediately, |
| * if the client were to release the first frames and then call obtainBuffer() again. |
| * This value is only a prediction, and needs to be confirmed. |
| * It will be set to zero for an error return. |
| * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), |
| * in case the requested amount of frames is in two or more non-contiguous regions. |
| * FIXME requested and elapsed are both relative times. Consider changing to absolute time. |
| */ |
| status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, |
| struct timespec *elapsed = NULL, size_t *nonContig = NULL); |
| public: |
| |
| /* Public API for TRANSFER_OBTAIN mode. |
| * Release a filled buffer of frames for AudioFlinger to process. |
| * |
| * Buffer fields: |
| * frameCount currently ignored but recommend to set to actual number of frames filled |
| * size actual number of bytes filled, must be multiple of frameSize |
| * raw ignored |
| */ |
| void releaseBuffer(const Buffer* audioBuffer); |
| |
| /* As a convenience we provide a write() interface to the audio buffer. |
| * Input parameter 'size' is in byte units. |
| * 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 static buffer or streaming mode |
| * BAD_VALUE size is invalid |
| * WOULD_BLOCK when obtainBuffer() returns same, or |
| * AudioTrack was stopped during the write |
| * DEAD_OBJECT when AudioFlinger dies or the output device changes and |
| * the track cannot be automatically restored. |
| * The application needs to recreate the AudioTrack |
| * because the audio device changed or AudioFlinger died. |
| * This typically occurs for direct or offload tracks |
| * or if mDoNotReconnect is true. |
| * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). |
| * Default behavior is to only return when all data has been transferred. Set 'blocking' to |
| * false for the method to return immediately without waiting to try multiple times to write |
| * the full content of the buffer. |
| */ |
| ssize_t write(const void* buffer, size_t size, bool blocking = true); |
| |
| /* |
| * Dumps the state of an audio track. |
| * Not a general-purpose API; intended only for use by media player service to dump its tracks. |
| */ |
| status_t dump(int fd, const Vector<String16>& args) const; |
| |
| /* |
| * Return the total number of frames which AudioFlinger desired but were unavailable, |
| * and thus which resulted in an underrun. Reset to zero by stop(). |
| */ |
| uint32_t getUnderrunFrames() const; |
| |
| /* Get the flags */ |
| audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } |
| |
| /* Set parameters - only possible when using direct output */ |
| status_t setParameters(const String8& keyValuePairs); |
| |
| /* Sets the volume shaper object */ |
| VolumeShaper::Status applyVolumeShaper( |
| const sp<VolumeShaper::Configuration>& configuration, |
| const sp<VolumeShaper::Operation>& operation); |
| |
| /* Gets the volume shaper state */ |
| sp<VolumeShaper::State> getVolumeShaperState(int id); |
| |
| /* Get parameters */ |
| String8 getParameters(const String8& keys); |
| |
| /* Poll for a timestamp on demand. |
| * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, |
| * or if you need to get the most recent timestamp outside of the event callback handler. |
| * Caution: calling this method too often may be inefficient; |
| * if you need a high resolution mapping between frame position and presentation time, |
| * consider implementing that at application level, based on the low resolution timestamps. |
| * Returns NO_ERROR if timestamp is valid. |
| * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after |
| * start/ACTIVE, when the number of frames consumed is less than the |
| * overall hardware latency to physical output. In WOULD_BLOCK cases, |
| * one might poll again, or use getPosition(), or use 0 position and |
| * current time for the timestamp. |
| * DEAD_OBJECT if AudioFlinger dies or the output device changes and |
| * the track cannot be automatically restored. |
| * The application needs to recreate the AudioTrack |
| * because the audio device changed or AudioFlinger died. |
| * This typically occurs for direct or offload tracks |
| * or if mDoNotReconnect is true. |
| * INVALID_OPERATION wrong state, or some other error. |
| * |
| * The timestamp parameter is undefined on return, if status is not NO_ERROR. |
| */ |
| status_t getTimestamp(AudioTimestamp& timestamp); |
| private: |
| status_t getTimestamp_l(AudioTimestamp& timestamp); |
| public: |
| |
| /* Return the extended timestamp, with additional timebase info and improved drain behavior. |
| * |
| * This is similar to the AudioTrack.java API: |
| * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) |
| * |
| * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method |
| * |
| * 1. stop() by itself does not reset the frame position. |
| * A following start() resets the frame position to 0. |
| * 2. flush() by itself does not reset the frame position. |
| * The frame position advances by the number of frames flushed, |
| * when the first frame after flush reaches the audio sink. |
| * 3. BOOTTIME clock offsets are provided to help synchronize with |
| * non-audio streams, e.g. sensor data. |
| * 4. Position is returned with 64 bits of resolution. |
| * |
| * Parameters: |
| * timestamp: A pointer to the caller allocated ExtendedTimestamp. |
| * |
| * Returns NO_ERROR on success; timestamp is filled with valid data. |
| * BAD_VALUE if timestamp is NULL. |
| * WOULD_BLOCK if called immediately after start() when the number |
| * of frames consumed is less than the |
| * overall hardware latency to physical output. In WOULD_BLOCK cases, |
| * one might poll again, or use getPosition(), or use 0 position and |
| * current time for the timestamp. |
| * If WOULD_BLOCK is returned, the timestamp is still |
| * modified with the LOCATION_CLIENT portion filled. |
| * DEAD_OBJECT if AudioFlinger dies or the output device changes and |
| * the track cannot be automatically restored. |
| * The application needs to recreate the AudioTrack |
| * because the audio device changed or AudioFlinger died. |
| * This typically occurs for direct or offloaded tracks |
| * or if mDoNotReconnect is true. |
| * INVALID_OPERATION if called on a offloaded or direct track. |
| * Use getTimestamp(AudioTimestamp& timestamp) instead. |
| */ |
| status_t getTimestamp(ExtendedTimestamp *timestamp); |
| private: |
| status_t getTimestamp_l(ExtendedTimestamp *timestamp); |
| public: |
| |
| /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this |
| * AudioTrack is routed is updated. |
| * Replaces any previously installed callback. |
| * Parameters: |
| * callback: The callback interface |
| * Returns NO_ERROR if successful. |
| * INVALID_OPERATION if the same callback is already installed. |
| * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable |
| * BAD_VALUE if the callback is NULL |
| */ |
| status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); |
| |
| /* remove an AudioDeviceCallback. |
| * Parameters: |
| * callback: The callback interface |
| * Returns NO_ERROR if successful. |
| * INVALID_OPERATION if the callback is not installed |
| * BAD_VALUE if the callback is NULL |
| */ |
| status_t removeAudioDeviceCallback( |
| const sp<AudioSystem::AudioDeviceCallback>& callback); |
| |
| /* Obtain the pending duration in milliseconds for playback of pure PCM |
| * (mixable without embedded timing) data remaining in AudioTrack. |
| * |
| * This is used to estimate the drain time for the client-server buffer |
| * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. |
| * One may optionally request to find the duration to play through the HAL |
| * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, |
| * INVALID_OPERATION may be returned if the kernel location is unavailable. |
| * |
| * Returns NO_ERROR if successful. |
| * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained |
| * or the AudioTrack does not contain pure PCM data. |
| * BAD_VALUE if msec is nullptr or location is invalid. |
| */ |
| status_t pendingDuration(int32_t *msec, |
| ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); |
| |
| /* hasStarted() is used to determine if audio is now audible at the device after |
| * a start() command. The underlying implementation checks a nonzero timestamp position |
| * or increment for the audible assumption. |
| * |
| * hasStarted() returns true if the track has been started() and audio is audible |
| * and no subsequent pause() or flush() has been called. Immediately after pause() or |
| * flush() hasStarted() will return false. |
| * |
| * If stop() has been called, hasStarted() will return true if audio is still being |
| * delivered or has finished delivery (even if no audio was written) for both offloaded |
| * and normal tracks. This property removes a race condition in checking hasStarted() |
| * for very short clips, where stop() must be called to finish drain. |
| * |
| * In all cases, hasStarted() may turn false briefly after a subsequent start() is called |
| * until audio becomes audible again. |
| */ |
| bool hasStarted(); // not const |
| |
| bool isPlaying() { |
| AutoMutex lock(mLock); |
| return mState == STATE_ACTIVE || mState == STATE_STOPPING; |
| } |
| 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); |
| |
| // Do not call Thread::requestExitAndWait() without first calling requestExit(). |
| // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. |
| virtual void requestExit(); |
| |
| void pause(); // suspend thread from execution at next loop boundary |
| void resume(); // allow thread to execute, if not requested to exit |
| void wake(); // wake to handle changed notification conditions. |
| |
| private: |
| void pauseInternal(nsecs_t ns = 0LL); |
| // like pause(), but only used internally within thread |
| |
| friend class AudioTrack; |
| virtual bool threadLoop(); |
| AudioTrack& mReceiver; |
| virtual ~AudioTrackThread(); |
| Mutex mMyLock; // Thread::mLock is private |
| Condition mMyCond; // Thread::mThreadExitedCondition is private |
| bool mPaused; // whether thread is requested to pause at next loop entry |
| bool mPausedInt; // whether thread internally requests pause |
| nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored |
| bool mIgnoreNextPausedInt; // skip any internal pause and go immediately |
| // to processAudioBuffer() as state may have changed |
| // since pause time calculated. |
| }; |
| |
| // body of AudioTrackThread::threadLoop() |
| // returns the maximum amount of time before we would like to run again, where: |
| // 0 immediately |
| // > 0 no later than this many nanoseconds from now |
| // NS_WHENEVER still active but no particular deadline |
| // NS_INACTIVE inactive so don't run again until re-started |
| // NS_NEVER never again |
| static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; |
| nsecs_t processAudioBuffer(); |
| |
| // caller must hold lock on mLock for all _l methods |
| |
| status_t createTrack_l(); |
| |
| // can only be called when mState != STATE_ACTIVE |
| void flush_l(); |
| |
| void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); |
| |
| // FIXME enum is faster than strcmp() for parameter 'from' |
| status_t restoreTrack_l(const char *from); |
| |
| uint32_t getUnderrunCount_l() const; |
| |
| bool isOffloaded() const; |
| bool isDirect() const; |
| bool isOffloadedOrDirect() const; |
| |
| bool isOffloaded_l() const |
| { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } |
| |
| bool isOffloadedOrDirect_l() const |
| { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| |
| AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } |
| |
| bool isDirect_l() const |
| { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } |
| |
| // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) |
| bool isPurePcmData_l() const |
| { return audio_is_linear_pcm(mFormat) |
| && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } |
| |
| // increment mPosition by the delta of mServer, and return new value of mPosition |
| Modulo<uint32_t> updateAndGetPosition_l(); |
| |
| // check sample rate and speed is compatible with AudioTrack |
| bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; |
| |
| void restartIfDisabled(); |
| |
| // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 |
| sp<IAudioTrack> mAudioTrack; |
| sp<IMemory> mCblkMemory; |
| audio_track_cblk_t* mCblk; // re-load after mLock.unlock() |
| audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() |
| |
| sp<AudioTrackThread> mAudioTrackThread; |
| bool mThreadCanCallJava; |
| |
| float mVolume[2]; |
| float mSendLevel; |
| mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it |
| uint32_t mOriginalSampleRate; |
| AudioPlaybackRate mPlaybackRate; |
| float mMaxRequiredSpeed; // use PCM buffer size to allow this speed |
| |
| // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. |
| // This allocated buffer size is maintained by the proxy. |
| size_t mFrameCount; // maximum size of buffer |
| |
| size_t mReqFrameCount; // frame count to request the first or next time |
| // a new IAudioTrack is needed, non-decreasing |
| |
| // The following AudioFlinger server-side values are cached in createAudioTrack_l(). |
| // These values can be used for informational purposes until the track is invalidated, |
| // whereupon restoreTrack_l() calls createTrack_l() to update the values. |
| uint32_t mAfLatency; // AudioFlinger latency in ms |
| size_t mAfFrameCount; // AudioFlinger frame count |
| uint32_t mAfSampleRate; // AudioFlinger sample rate |
| |
| // constant after constructor or set() |
| audio_format_t mFormat; // as requested by client, not forced to 16-bit |
| audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies |
| // this AudioTrack has valid attributes |
| uint32_t mChannelCount; |
| audio_channel_mask_t mChannelMask; |
| sp<IMemory> mSharedBuffer; |
| transfer_type mTransfer; |
| audio_offload_info_t mOffloadInfoCopy; |
| const audio_offload_info_t* mOffloadInfo; |
| audio_attributes_t mAttributes; |
| |
| size_t mFrameSize; // frame size in bytes |
| |
| status_t mStatus; |
| |
| // can change dynamically when IAudioTrack invalidated |
| uint32_t mLatency; // in ms |
| |
| // Indicates the current track state. Protected by mLock. |
| enum State { |
| STATE_ACTIVE, |
| STATE_STOPPED, |
| STATE_PAUSED, |
| STATE_PAUSED_STOPPING, |
| STATE_FLUSHED, |
| STATE_STOPPING, |
| } mState; |
| |
| // for client callback handler |
| callback_t mCbf; // callback handler for events, or NULL |
| void* mUserData; |
| |
| // for notification APIs |
| |
| // next 2 fields are const after constructor or set() |
| uint32_t mNotificationFramesReq; // requested number of frames between each |
| // notification callback, |
| // at initial source sample rate |
| uint32_t mNotificationsPerBufferReq; |
| // requested number of notifications per buffer, |
| // currently only used for fast tracks with |
| // default track buffer size |
| |
| uint32_t mNotificationFramesAct; // actual number of frames between each |
| // notification callback, |
| // at initial source sample rate |
| bool mRefreshRemaining; // processAudioBuffer() should refresh |
| // mRemainingFrames and mRetryOnPartialBuffer |
| |
| // used for static track cbf and restoration |
| int32_t mLoopCount; // last setLoop loopCount; zero means disabled |
| uint32_t mLoopStart; // last setLoop loopStart |
| uint32_t mLoopEnd; // last setLoop loopEnd |
| int32_t mLoopCountNotified; // the last loopCount notified by callback. |
| // mLoopCountNotified counts down, matching |
| // the remaining loop count for static track |
| // playback. |
| |
| // These are private to processAudioBuffer(), and are not protected by a lock |
| uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() |
| bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() |
| uint32_t mObservedSequence; // last observed value of mSequence |
| |
| Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units |
| bool mMarkerReached; |
| Modulo<uint32_t> mNewPosition; // in frames |
| uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS |
| |
| Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() |
| // which is count of frames consumed by server, |
| // reset by new IAudioTrack, |
| // whether it is reset by stop() is TBD |
| Modulo<uint32_t> mPosition; // in frames, like mServer except continues |
| // monotonically after new IAudioTrack, |
| // and could be easily widened to uint64_t |
| Modulo<uint32_t> mReleased; // count of frames released to server |
| // but not necessarily consumed by server, |
| // reset by stop() but continues monotonically |
| // after new IAudioTrack to restore mPosition, |
| // and could be easily widened to uint64_t |
| int64_t mStartUs; // the start time after flush or stop. |
| // only used for offloaded and direct tracks. |
| ExtendedTimestamp mStartEts; // Extended timestamp at start for normal |
| // AudioTracks. |
| AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct |
| // AudioTracks. |
| |
| bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid |
| bool mTimestampStartupGlitchReported; // reduce log spam |
| bool mRetrogradeMotionReported; // reduce log spam |
| AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion |
| ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp |
| |
| uint32_t mUnderrunCountOffset; // updated when restoring tracks |
| |
| int64_t mFramesWritten; // total frames written. reset to zero after |
| // the start() following stop(). It is not |
| // changed after restoring the track or |
| // after flush. |
| int64_t mFramesWrittenServerOffset; // An offset to server frames due to |
| // restoring AudioTrack, or stop/start. |
| // This offset is also used for static tracks. |
| int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames |
| // delivered for static tracks). |
| // -1 indicates no previous restore point. |
| |
| audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may |
| // be denied by client or server, such as |
| // AUDIO_OUTPUT_FLAG_FAST. mLock must be |
| // held to read or write those bits reliably. |
| audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const |
| |
| bool mDoNotReconnect; |
| |
| audio_session_t mSessionId; |
| int mAuxEffectId; |
| |
| mutable Mutex mLock; |
| |
| int mPreviousPriority; // before start() |
| SchedPolicy mPreviousSchedulingGroup; |
| bool mAwaitBoost; // thread should wait for priority boost before running |
| |
| // The proxy should only be referenced while a lock is held because the proxy isn't |
| // multi-thread safe, especially the SingleStateQueue part of the proxy. |
| // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, |
| // provided that the caller also holds an extra reference to the proxy and shared memory to keep |
| // them around in case they are replaced during the obtainBuffer(). |
| sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only |
| sp<AudioTrackClientProxy> mProxy; // primary owner of the memory |
| |
| bool mInUnderrun; // whether track is currently in underrun state |
| uint32_t mPausedPosition; |
| |
| // For Device Selection API |
| // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. |
| audio_port_handle_t mSelectedDeviceId; |
| |
| sp<VolumeHandler> mVolumeHandler; |
| |
| private: |
| class DeathNotifier : public IBinder::DeathRecipient { |
| public: |
| DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } |
| protected: |
| virtual void binderDied(const wp<IBinder>& who); |
| private: |
| const wp<AudioTrack> mAudioTrack; |
| }; |
| |
| sp<DeathNotifier> mDeathNotifier; |
| uint32_t mSequence; // incremented for each new IAudioTrack attempt |
| uid_t mClientUid; |
| pid_t mClientPid; |
| |
| sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; |
| audio_port_handle_t mPortId; // unique ID allocated by audio policy |
| }; |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIOTRACK_H |
