Moved headers from include/ to appropriate libs
include/camera -> camera/include/camera
include/media/audiohal -> media/libaudiohal/include
include/media/AudioResampler*.h -> media/libaudioprocessing/include
include/media/Audio*.h,IAudio*.h,IEffect*.h,ToneGenerator.h -> media/libaudioclient/include
include/media/EffectsFactoryApi.h -> media/libeffects/include
include/media/stagefright -> media/libstagefright/include
include/media/nbaio -> media/libnbaio/include
include/media/<rest of files> -> media/libmedia/include
include/cpustats -> media/libcpustats/include/cpustats
Added symlinks from old location to new ones
Bug: 33241851
Test: VNDK linked modules will need to add explicit lib dep.
All other modules should compile the same
Change-Id: I0ecf754a2132640ae781a3cc31428fb8c0bd1669
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 03dce0c..ad130e0 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -26,6 +26,8 @@
"libaudioutils",
],
export_shared_lib_headers: ["libbinder"],
+ local_include_dirs: ["include"],
+ export_include_dirs: ["include"],
// for memory heap analysis
static_libs: [
"libc_malloc_debug_backtrace",
diff --git a/media/libaudioclient/include/AudioBufferProvider.h b/media/libaudioclient/include/AudioBufferProvider.h
new file mode 100644
index 0000000..458d170
--- /dev/null
+++ b/media/libaudioclient/include/AudioBufferProvider.h
@@ -0,0 +1,75 @@
+/*
+ * 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_AUDIO_BUFFER_PROVIDER_H
+#define ANDROID_AUDIO_BUFFER_PROVIDER_H
+
+#include <utils/Errors.h>
+
+namespace android {
+// ----------------------------------------------------------------------------
+
+class AudioBufferProvider
+{
+public:
+
+ // FIXME merge with AudioTrackShared::Buffer, AudioTrack::Buffer, and AudioRecord::Buffer
+ // and rename getNextBuffer() to obtainBuffer()
+ struct Buffer {
+ Buffer() : raw(NULL), frameCount(0) { }
+ union {
+ void* raw;
+ short* i16;
+ int8_t* i8;
+ };
+ size_t frameCount;
+ };
+
+ virtual ~AudioBufferProvider() {}
+
+ // On entry:
+ // buffer != NULL
+ // buffer->raw unused
+ // buffer->frameCount maximum number of desired frames
+ // On successful return:
+ // status NO_ERROR
+ // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames
+ // buffer->frameCount number of contiguous available frames at buffer->raw,
+ // 0 < buffer->frameCount <= entry value
+ // On error return:
+ // status != NO_ERROR
+ // buffer->raw NULL
+ // buffer->frameCount 0
+ virtual status_t getNextBuffer(Buffer* buffer) = 0;
+
+ // Release (a portion of) the buffer previously obtained by getNextBuffer().
+ // It is permissible to call releaseBuffer() multiple times per getNextBuffer().
+ // On entry:
+ // buffer->frameCount number of frames to release, must be <= number of frames
+ // obtained but not yet released
+ // buffer->raw unused
+ // On return:
+ // buffer->frameCount 0; implementation MUST set to zero
+ // buffer->raw undefined; implementation is PERMITTED to set to any value,
+ // so if caller needs to continue using this buffer it must
+ // keep track of the pointer itself
+ virtual void releaseBuffer(Buffer* buffer) = 0;
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif // ANDROID_AUDIO_BUFFER_PROVIDER_H
diff --git a/media/libaudioclient/include/AudioEffect.h b/media/libaudioclient/include/AudioEffect.h
new file mode 100644
index 0000000..bfc068b
--- /dev/null
+++ b/media/libaudioclient/include/AudioEffect.h
@@ -0,0 +1,488 @@
+/*
+ * Copyright (C) 2009 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_AUDIOEFFECT_H
+#define ANDROID_AUDIOEFFECT_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <media/IAudioFlinger.h>
+#include <media/IAudioPolicyService.h>
+#include <media/IEffect.h>
+#include <media/IEffectClient.h>
+#include <media/AudioSystem.h>
+#include <system/audio_effect.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+struct effect_param_cblk_t;
+
+// ----------------------------------------------------------------------------
+
+class AudioEffect : public RefBase
+{
+public:
+
+ /*
+ * Static methods for effects enumeration.
+ */
+
+ /*
+ * Returns the number of effects available. This method together
+ * with queryEffect() is used to enumerate all effects:
+ * The enumeration sequence is:
+ * queryNumberEffects(&num_effects);
+ * for (i = 0; i < num_effects; i++)
+ * queryEffect(i,...);
+ *
+ * Parameters:
+ * numEffects: address where the number of effects should be returned.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * NO_ERROR successful operation.
+ * PERMISSION_DENIED could not get AudioFlinger interface
+ * NO_INIT effect library failed to initialize
+ * BAD_VALUE invalid numEffects pointer
+ *
+ * Returned value
+ * *numEffects: updated with number of effects available
+ */
+ static status_t queryNumberEffects(uint32_t *numEffects);
+
+ /*
+ * Returns an effect descriptor during effect
+ * enumeration.
+ *
+ * Parameters:
+ * index: index of the queried effect.
+ * descriptor: address where the effect descriptor should be returned.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * NO_ERROR successful operation.
+ * PERMISSION_DENIED could not get AudioFlinger interface
+ * NO_INIT effect library failed to initialize
+ * BAD_VALUE invalid descriptor pointer or index
+ * INVALID_OPERATION effect list has changed since last execution of queryNumberEffects()
+ *
+ * Returned value
+ * *descriptor: updated with effect descriptor
+ */
+ static status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor);
+
+
+ /*
+ * Returns the descriptor for the specified effect uuid.
+ *
+ * Parameters:
+ * uuid: pointer to effect uuid.
+ * descriptor: address where the effect descriptor should be returned.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * NO_ERROR successful operation.
+ * PERMISSION_DENIED could not get AudioFlinger interface
+ * NO_INIT effect library failed to initialize
+ * BAD_VALUE invalid uuid or descriptor pointers
+ * NAME_NOT_FOUND no effect with this uuid found
+ *
+ * Returned value
+ * *descriptor updated with effect descriptor
+ */
+ static status_t getEffectDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *descriptor) /*const*/;
+
+
+ /*
+ * Returns a list of descriptors corresponding to the pre processings enabled by default
+ * on an AudioRecord with the supplied audio session ID.
+ *
+ * Parameters:
+ * audioSession: audio session ID.
+ * descriptors: address where the effect descriptors should be returned.
+ * count: as input, the maximum number of descriptor than should be returned
+ * as output, the number of descriptor returned if status is NO_ERROR or the actual
+ * number of enabled pre processings if status is NO_MEMORY
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * NO_ERROR successful operation.
+ * NO_MEMORY the number of descriptor to return is more than the maximum number
+ * indicated by count.
+ * PERMISSION_DENIED could not get AudioFlinger interface
+ * NO_INIT effect library failed to initialize
+ * BAD_VALUE invalid audio session or descriptor pointers
+ *
+ * Returned value
+ * *descriptor updated with descriptors of pre processings enabled by default
+ * *count number of descriptors returned if returned status is NO_ERROR.
+ * total number of pre processing enabled by default if returned status is
+ * NO_MEMORY. This happens if the count passed as input is less than the number
+ * of descriptors to return.
+ * *count is limited to kMaxPreProcessing on return.
+ */
+ static status_t queryDefaultPreProcessing(audio_session_t audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count);
+
+ /*
+ * Events used by callback function (effect_callback_t).
+ */
+ enum event_type {
+ EVENT_CONTROL_STATUS_CHANGED = 0,
+ EVENT_ENABLE_STATUS_CHANGED = 1,
+ EVENT_PARAMETER_CHANGED = 2,
+ EVENT_ERROR = 3
+ };
+
+ /* Callback function notifying client application of a change in effect engine state or
+ * configuration.
+ * An effect engine can be shared by several applications but only one has the control
+ * of the engine activity and configuration at a time.
+ * The EVENT_CONTROL_STATUS_CHANGED event is received when an application loses or
+ * retrieves the control of the effect engine. Loss of control happens
+ * if another application requests the use of the engine by creating an AudioEffect for
+ * the same effect type but with a higher priority. Control is returned when the
+ * application having the control deletes its AudioEffect object.
+ * The EVENT_ENABLE_STATUS_CHANGED event is received by all applications not having the
+ * control of the effect engine when the effect is enabled or disabled.
+ * The EVENT_PARAMETER_CHANGED event is received by all applications not having the
+ * control of the effect engine when an effect parameter is changed.
+ * The EVENT_ERROR event is received when the media server process dies.
+ *
+ * Parameters:
+ *
+ * event: type of event notified (see enum AudioEffect::event_type).
+ * user: Pointer to context for use by the callback receiver.
+ * info: Pointer to optional parameter according to event type:
+ * - EVENT_CONTROL_STATUS_CHANGED: boolean indicating if control is granted (true)
+ * or stolen (false).
+ * - EVENT_ENABLE_STATUS_CHANGED: boolean indicating if effect is now enabled (true)
+ * or disabled (false).
+ * - EVENT_PARAMETER_CHANGED: pointer to a effect_param_t structure.
+ * - EVENT_ERROR: status_t indicating the error (DEAD_OBJECT when media server dies).
+ */
+
+ typedef void (*effect_callback_t)(int32_t event, void* user, void *info);
+
+
+ /* Constructor.
+ * AudioEffect is the base class for creating and controlling an effect engine from
+ * the application process. Creating an AudioEffect object will create the effect engine
+ * in the AudioFlinger if no engine of the specified type exists. If one exists, this engine
+ * will be used. The application creating the AudioEffect object (or a derived class like
+ * Reverb for instance) will either receive control of the effect engine or not, depending
+ * on the priority parameter. If priority is higher than the priority used by the current
+ * effect engine owner, the control will be transfered to the new application. Otherwise
+ * control will remain to the previous application. In this case, the new application will be
+ * notified of changes in effect engine state or control ownership by the effect callback.
+ * After creating the AudioEffect, the application must call the initCheck() method and
+ * check the creation status before trying to control the effect engine (see initCheck()).
+ * If the effect is to be applied to an AudioTrack or MediaPlayer only the application
+ * must specify the audio session ID corresponding to this player.
+ */
+
+ /* Simple Constructor.
+ *
+ * Parameters:
+ *
+ * opPackageName: The package name used for app op checks.
+ */
+ AudioEffect(const String16& opPackageName);
+
+
+ /* Constructor.
+ *
+ * Parameters:
+ *
+ * type: type of effect created: can be null if uuid is specified. This corresponds to
+ * the OpenSL ES interface implemented by this effect.
+ * opPackageName: The package name used for app op checks.
+ * uuid: Uuid of effect created: can be null if type is specified. This uuid corresponds to
+ * a particular implementation of an effect type.
+ * priority: requested priority for effect control: the priority level corresponds to the
+ * value of priority parameter: negative values indicate lower priorities, positive values
+ * higher priorities, 0 being the normal priority.
+ * cbf: optional callback function (see effect_callback_t)
+ * user: pointer to context for use by the callback receiver.
+ * sessionID: audio session this effect is associated to.
+ * If equal to AUDIO_SESSION_OUTPUT_MIX, the effect will be global to
+ * the output mix. Otherwise, the effect will be applied to all players
+ * (AudioTrack or MediaPLayer) within the same audio session.
+ * io: HAL audio output or input stream to which this effect must be attached. Leave at 0 for
+ * automatic output selection by AudioFlinger.
+ */
+
+ AudioEffect(const effect_uuid_t *type,
+ const String16& opPackageName,
+ const effect_uuid_t *uuid = NULL,
+ int32_t priority = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
+ audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
+ audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
+ );
+
+ /* Constructor.
+ * Same as above but with type and uuid specified by character strings
+ */
+ AudioEffect(const char *typeStr,
+ const String16& opPackageName,
+ const char *uuidStr = NULL,
+ int32_t priority = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
+ audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
+ audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
+ );
+
+ /* Terminates the AudioEffect and unregisters it from AudioFlinger.
+ * The effect engine is also destroyed if this AudioEffect was the last controlling
+ * the engine.
+ */
+ ~AudioEffect();
+
+ /* Initialize an uninitialized AudioEffect.
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR or ALREADY_EXISTS: successful initialization
+ * - INVALID_OPERATION: AudioEffect is already initialized
+ * - BAD_VALUE: invalid parameter
+ * - NO_INIT: audio flinger or audio hardware not initialized
+ * */
+ status_t set(const effect_uuid_t *type,
+ const effect_uuid_t *uuid = NULL,
+ int32_t priority = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
+ audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
+ audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
+ );
+
+ /* Result of constructing the AudioEffect. This must be checked
+ * before using any AudioEffect API.
+ * initCheck() can return:
+ * - NO_ERROR: the effect engine is successfully created and the application has control.
+ * - ALREADY_EXISTS: the effect engine is successfully created but the application does not
+ * have control.
+ * - NO_INIT: the effect creation failed.
+ *
+ */
+ status_t initCheck() const;
+
+
+ /* Returns the unique effect Id for the controlled effect engine. This ID is unique
+ * system wide and is used for instance in the case of auxiliary effects to attach
+ * the effect to an AudioTrack or MediaPlayer.
+ *
+ */
+ int32_t id() const { return mId; }
+
+ /* Returns a descriptor for the effect (see effect_descriptor_t in audio_effect.h).
+ */
+ effect_descriptor_t descriptor() const;
+
+ /* Returns effect control priority of this AudioEffect object.
+ */
+ int32_t priority() const { return mPriority; }
+
+
+ /* Enables or disables the effect engine.
+ *
+ * Parameters:
+ * enabled: requested enable state.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - INVALID_OPERATION: the application does not have control of the effect engine or the
+ * effect is already in the requested state.
+ */
+ virtual status_t setEnabled(bool enabled);
+ bool getEnabled() const;
+
+ /* Sets a parameter value.
+ *
+ * Parameters:
+ * param: pointer to effect_param_t structure containing the parameter
+ * and its value (See audio_effect.h).
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation.
+ * - INVALID_OPERATION: the application does not have control of the effect engine.
+ * - BAD_VALUE: invalid parameter identifier or value.
+ * - DEAD_OBJECT: the effect engine has been deleted.
+ */
+ virtual status_t setParameter(effect_param_t *param);
+
+ /* Prepare a new parameter value that will be set by next call to
+ * setParameterCommit(). This method can be used to set multiple parameters
+ * in a synchronous manner or to avoid multiple binder calls for each
+ * parameter.
+ *
+ * Parameters:
+ * param: pointer to effect_param_t structure containing the parameter
+ * and its value (See audio_effect.h).
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation.
+ * - INVALID_OPERATION: the application does not have control of the effect engine.
+ * - NO_MEMORY: no more space available in shared memory used for deferred parameter
+ * setting.
+ */
+ virtual status_t setParameterDeferred(effect_param_t *param);
+
+ /* Commit all parameter values previously prepared by setParameterDeferred().
+ *
+ * Parameters:
+ * none
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation.
+ * - INVALID_OPERATION: No new parameter values ready for commit.
+ * - BAD_VALUE: invalid parameter identifier or value: there is no indication
+ * as to which of the parameters caused this error.
+ * - DEAD_OBJECT: the effect engine has been deleted.
+ */
+ virtual status_t setParameterCommit();
+
+ /* Gets a parameter value.
+ *
+ * Parameters:
+ * param: pointer to effect_param_t structure containing the parameter
+ * and the returned value (See audio_effect.h).
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation.
+ * - INVALID_OPERATION: the AudioEffect was not successfully initialized.
+ * - BAD_VALUE: invalid parameter identifier.
+ * - DEAD_OBJECT: the effect engine has been deleted.
+ */
+ virtual status_t getParameter(effect_param_t *param);
+
+ /* Sends a command and receives a response to/from effect engine.
+ * See audio_effect.h for details on effect command() function, valid command codes
+ * and formats.
+ */
+ virtual status_t command(uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *cmdData,
+ uint32_t *replySize,
+ void *replyData);
+
+
+ /*
+ * Utility functions.
+ */
+
+ /* Converts the string passed as first argument to the effect_uuid_t
+ * pointed to by second argument
+ */
+ static status_t stringToGuid(const char *str, effect_uuid_t *guid);
+ /* Converts the effect_uuid_t pointed to by first argument to the
+ * string passed as second argument
+ */
+ static status_t guidToString(const effect_uuid_t *guid, char *str, size_t maxLen);
+
+ // kMaxPreProcessing is a reasonable value for the maximum number of preprocessing effects
+ // that can be applied simultaneously.
+ static const uint32_t kMaxPreProcessing = 10;
+
+protected:
+ bool mEnabled; // enable state
+ audio_session_t mSessionId; // audio session ID
+ int32_t mPriority; // priority for effect control
+ status_t mStatus; // effect status
+ effect_callback_t mCbf; // callback function for status, control and
+ // parameter changes notifications
+ void* mUserData; // client context for callback function
+ effect_descriptor_t mDescriptor; // effect descriptor
+ int32_t mId; // system wide unique effect engine instance ID
+ Mutex mLock; // Mutex for mEnabled access
+
+ String16 mOpPackageName; // The package name used for app op checks.
+
+ // IEffectClient
+ virtual void controlStatusChanged(bool controlGranted);
+ virtual void enableStatusChanged(bool enabled);
+ virtual void commandExecuted(uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t replySize,
+ void *pReplyData);
+
+private:
+
+ // Implements the IEffectClient interface
+ class EffectClient :
+ public android::BnEffectClient, public android::IBinder::DeathRecipient
+ {
+ public:
+
+ EffectClient(AudioEffect *effect) : mEffect(effect){}
+
+ // IEffectClient
+ virtual void controlStatusChanged(bool controlGranted) {
+ sp<AudioEffect> effect = mEffect.promote();
+ if (effect != 0) {
+ effect->controlStatusChanged(controlGranted);
+ }
+ }
+ virtual void enableStatusChanged(bool enabled) {
+ sp<AudioEffect> effect = mEffect.promote();
+ if (effect != 0) {
+ effect->enableStatusChanged(enabled);
+ }
+ }
+ virtual void commandExecuted(uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t replySize,
+ void *pReplyData) {
+ sp<AudioEffect> effect = mEffect.promote();
+ if (effect != 0) {
+ effect->commandExecuted(
+ cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+ }
+ }
+
+ // IBinder::DeathRecipient
+ virtual void binderDied(const wp<IBinder>& /*who*/) {
+ sp<AudioEffect> effect = mEffect.promote();
+ if (effect != 0) {
+ effect->binderDied();
+ }
+ }
+
+ private:
+ wp<AudioEffect> mEffect;
+ };
+
+ void binderDied();
+
+ sp<IEffect> mIEffect; // IEffect binder interface
+ sp<EffectClient> mIEffectClient; // IEffectClient implementation
+ sp<IMemory> mCblkMemory; // shared memory for deferred parameter setting
+ effect_param_cblk_t* mCblk; // control block for deferred parameter setting
+ pid_t mClientPid;
+};
+
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOEFFECT_H
diff --git a/media/libaudioclient/include/AudioIoDescriptor.h b/media/libaudioclient/include/AudioIoDescriptor.h
new file mode 100644
index 0000000..fed86c9
--- /dev/null
+++ b/media/libaudioclient/include/AudioIoDescriptor.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2015 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_AUDIO_IO_DESCRIPTOR_H
+#define ANDROID_AUDIO_IO_DESCRIPTOR_H
+
+namespace android {
+
+enum audio_io_config_event {
+ AUDIO_OUTPUT_OPENED,
+ AUDIO_OUTPUT_CLOSED,
+ AUDIO_OUTPUT_CONFIG_CHANGED,
+ AUDIO_INPUT_OPENED,
+ AUDIO_INPUT_CLOSED,
+ AUDIO_INPUT_CONFIG_CHANGED,
+};
+
+// audio input/output descriptor used to cache output configurations in client process to avoid
+// frequent calls through IAudioFlinger
+class AudioIoDescriptor : public RefBase {
+public:
+ AudioIoDescriptor() :
+ mIoHandle(AUDIO_IO_HANDLE_NONE),
+ mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(AUDIO_CHANNEL_NONE),
+ mFrameCount(0), mFrameCountHAL(0), mLatency(0)
+ {
+ memset(&mPatch, 0, sizeof(struct audio_patch));
+ }
+
+ virtual ~AudioIoDescriptor() {}
+
+ audio_port_handle_t getDeviceId() {
+ if (mPatch.num_sources != 0 && mPatch.num_sinks != 0) {
+ if (mPatch.sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ // this is an output mix
+ // FIXME: the API only returns the first device in case of multiple device selection
+ return mPatch.sinks[0].id;
+ } else {
+ // this is an input mix
+ return mPatch.sources[0].id;
+ }
+ }
+ return AUDIO_PORT_HANDLE_NONE;
+ }
+
+ audio_io_handle_t mIoHandle;
+ struct audio_patch mPatch;
+ uint32_t mSamplingRate;
+ audio_format_t mFormat;
+ audio_channel_mask_t mChannelMask;
+ size_t mFrameCount;
+ size_t mFrameCountHAL;
+ uint32_t mLatency; // only valid for output
+};
+
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_IO_DESCRIPTOR_H*/
diff --git a/media/libaudioclient/include/AudioMixer.h b/media/libaudioclient/include/AudioMixer.h
new file mode 100644
index 0000000..87ada76
--- /dev/null
+++ b/media/libaudioclient/include/AudioMixer.h
@@ -0,0 +1,389 @@
+/*
+**
+** Copyright 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_AUDIO_MIXER_H
+#define ANDROID_AUDIO_MIXER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/BufferProviders.h>
+#include <media/nbaio/NBLog.h>
+#include <system/audio.h>
+#include <utils/Compat.h>
+#include <utils/threads.h>
+
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioMixer
+{
+public:
+ AudioMixer(size_t frameCount, uint32_t sampleRate,
+ uint32_t maxNumTracks = MAX_NUM_TRACKS);
+
+ /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
+
+
+ // This mixer has a hard-coded upper limit of 32 active track inputs.
+ // Adding support for > 32 tracks would require more than simply changing this value.
+ static const uint32_t MAX_NUM_TRACKS = 32;
+ // maximum number of channels supported by the mixer
+
+ // This mixer has a hard-coded upper limit of 8 channels for output.
+ static const uint32_t MAX_NUM_CHANNELS = 8;
+ static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
+ // maximum number of channels supported for the content
+ static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
+
+ static const uint16_t UNITY_GAIN_INT = 0x1000;
+ static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
+
+ enum { // names
+
+ // track names (MAX_NUM_TRACKS units)
+ TRACK0 = 0x1000,
+
+ // 0x2000 is unused
+
+ // setParameter targets
+ TRACK = 0x3000,
+ RESAMPLE = 0x3001,
+ RAMP_VOLUME = 0x3002, // ramp to new volume
+ VOLUME = 0x3003, // don't ramp
+ TIMESTRETCH = 0x3004,
+
+ // set Parameter names
+ // for target TRACK
+ CHANNEL_MASK = 0x4000,
+ FORMAT = 0x4001,
+ MAIN_BUFFER = 0x4002,
+ AUX_BUFFER = 0x4003,
+ DOWNMIX_TYPE = 0X4004,
+ MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+ // for target RESAMPLE
+ SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
+ // parameter 'value' is the new sample rate in Hz.
+ // Only creates a sample rate converter the first time that
+ // the track sample rate is different from the mix sample rate.
+ // If the new sample rate is the same as the mix sample rate,
+ // and a sample rate converter already exists,
+ // then the sample rate converter remains present but is a no-op.
+ RESET = 0x4101, // Reset sample rate converter without changing sample rate.
+ // This clears out the resampler's input buffer.
+ REMOVE = 0x4102, // Remove the sample rate converter on this track name;
+ // the track is restored to the mix sample rate.
+ // for target RAMP_VOLUME and VOLUME (8 channels max)
+ // FIXME use float for these 3 to improve the dynamic range
+ VOLUME0 = 0x4200,
+ VOLUME1 = 0x4201,
+ AUXLEVEL = 0x4210,
+ // for target TIMESTRETCH
+ PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name;
+ // parameter 'value' is a pointer to the new playback rate.
+ };
+
+
+ // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
+
+ // Allocate a track name. Returns new track name if successful, -1 on failure.
+ // The failure could be because of an invalid channelMask or format, or that
+ // the track capacity of the mixer is exceeded.
+ int getTrackName(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId);
+
+ // Free an allocated track by name
+ void deleteTrackName(int name);
+
+ // Enable or disable an allocated track by name
+ void enable(int name);
+ void disable(int name);
+
+ void setParameter(int name, int target, int param, void *value);
+
+ void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+ void process();
+
+ uint32_t trackNames() const { return mTrackNames; }
+
+ size_t getUnreleasedFrames(int name) const;
+
+ static inline bool isValidPcmTrackFormat(audio_format_t format) {
+ switch (format) {
+ case AUDIO_FORMAT_PCM_8_BIT:
+ case AUDIO_FORMAT_PCM_16_BIT:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return true;
+ default:
+ return false;
+ }
+ }
+
+private:
+
+ enum {
+ // FIXME this representation permits up to 8 channels
+ NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
+ };
+
+ enum {
+ NEEDS_CHANNEL_1 = 0x00000000, // mono
+ NEEDS_CHANNEL_2 = 0x00000001, // stereo
+
+ // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+ NEEDS_MUTE = 0x00000100,
+ NEEDS_RESAMPLE = 0x00001000,
+ NEEDS_AUX = 0x00010000,
+ };
+
+ struct state_t;
+ struct track_t;
+
+ typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
+ int32_t* aux);
+ static const int BLOCKSIZE = 16; // 4 cache lines
+
+ struct track_t {
+ uint32_t needs;
+
+ // TODO: Eventually remove legacy integer volume settings
+ union {
+ int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+ int32_t volumeRL;
+ };
+
+ int32_t prevVolume[MAX_NUM_VOLUMES];
+
+ // 16-byte boundary
+
+ int32_t volumeInc[MAX_NUM_VOLUMES];
+ int32_t auxInc;
+ int32_t prevAuxLevel;
+
+ // 16-byte boundary
+
+ int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+ uint16_t frameCount;
+
+ uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+ uint8_t unused_padding; // formerly format, was always 16
+ uint16_t enabled; // actually bool
+ audio_channel_mask_t channelMask;
+
+ // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
+ // for how the Track buffer provider is wrapped by another one when dowmixing is required
+ AudioBufferProvider* bufferProvider;
+
+ // 16-byte boundary
+
+ mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+ hook_t hook;
+ const void* in; // current location in buffer
+
+ // 16-byte boundary
+
+ AudioResampler* resampler;
+ uint32_t sampleRate;
+ int32_t* mainBuffer;
+ int32_t* auxBuffer;
+
+ // 16-byte boundary
+
+ /* Buffer providers are constructed to translate the track input data as needed.
+ *
+ * TODO: perhaps make a single PlaybackConverterProvider class to move
+ * all pre-mixer track buffer conversions outside the AudioMixer class.
+ *
+ * 1) mInputBufferProvider: The AudioTrack buffer provider.
+ * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
+ * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
+ * requires reformat. For example, it may convert floating point input to
+ * PCM_16_bit if that's required by the downmixer.
+ * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
+ * the number of channels required by the mixer sink.
+ * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+ * the downmixer requirements to the mixer engine input requirements.
+ * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
+ */
+ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
+ PassthruBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting.
+ PassthruBufferProvider* downmixerBufferProvider; // wrapper for channel conversion.
+ PassthruBufferProvider* mPostDownmixReformatBufferProvider;
+ PassthruBufferProvider* mTimestretchBufferProvider;
+
+ int32_t sessionId;
+
+ audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ audio_format_t mFormat; // input track format
+ audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ // each track must be converted to this format.
+ audio_format_t mDownmixRequiresFormat; // required downmixer format
+ // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+ // AUDIO_FORMAT_INVALID if no required format
+
+ float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
+ float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
+ float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment
+
+ float mAuxLevel; // floating point set aux level
+ float mPrevAuxLevel; // floating point prev aux level
+ float mAuxInc; // floating point aux increment
+
+ audio_channel_mask_t mMixerChannelMask;
+ uint32_t mMixerChannelCount;
+
+ AudioPlaybackRate mPlaybackRate;
+
+ bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+ bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+ bool doesResample() const { return resampler != NULL; }
+ void resetResampler() { if (resampler != NULL) resampler->reset(); }
+ void adjustVolumeRamp(bool aux, bool useFloat = false);
+ size_t getUnreleasedFrames() const { return resampler != NULL ?
+ resampler->getUnreleasedFrames() : 0; };
+
+ status_t prepareForDownmix();
+ void unprepareForDownmix();
+ status_t prepareForReformat();
+ void unprepareForReformat();
+ bool setPlaybackRate(const AudioPlaybackRate &playbackRate);
+ void reconfigureBufferProviders();
+ };
+
+ typedef void (*process_hook_t)(state_t* state);
+
+ // pad to 32-bytes to fill cache line
+ struct state_t {
+ uint32_t enabledTracks;
+ uint32_t needsChanged;
+ size_t frameCount;
+ process_hook_t hook; // one of process__*, never NULL
+ int32_t *outputTemp;
+ int32_t *resampleTemp;
+ NBLog::Writer* mLog;
+ int32_t reserved[1];
+ // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
+ track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
+ };
+
+ // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
+ uint32_t mTrackNames;
+
+ // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
+ // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
+ const uint32_t mConfiguredNames;
+
+ const uint32_t mSampleRate;
+
+ NBLog::Writer mDummyLog;
+public:
+ void setLog(NBLog::Writer* log);
+private:
+ state_t mState __attribute__((aligned(32)));
+
+ // Call after changing either the enabled status of a track, or parameters of an enabled track.
+ // OK to call more often than that, but unnecessary.
+ void invalidateState(uint32_t mask);
+
+ bool setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+ static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+ int32_t* aux);
+ static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+ int32_t* aux);
+ static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+ int32_t* aux);
+ static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+ int32_t* aux);
+ static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+ int32_t* aux);
+
+ static void process__validate(state_t* state);
+ static void process__nop(state_t* state);
+ static void process__genericNoResampling(state_t* state);
+ static void process__genericResampling(state_t* state);
+ static void process__OneTrack16BitsStereoNoResampling(state_t* state);
+
+ static pthread_once_t sOnceControl;
+ static void sInitRoutine();
+
+ /* multi-format volume mixing function (calls template functions
+ * in AudioMixerOps.h). The template parameters are as follows:
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27)
+ */
+ template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+ typename TO, typename TI, typename TA>
+ static void volumeMix(TO *out, size_t outFrames,
+ const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
+
+ // multi-format process hooks
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ static void process_NoResampleOneTrack(state_t* state);
+
+ // multi-format track hooks
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ static void track__Resample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ static void track__NoResample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+
+ static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+ // hook types
+ enum {
+ PROCESSTYPE_NORESAMPLEONETRACK,
+ };
+ enum {
+ TRACKTYPE_NOP,
+ TRACKTYPE_RESAMPLE,
+ TRACKTYPE_NORESAMPLE,
+ TRACKTYPE_NORESAMPLEMONO,
+ };
+
+ // functions for determining the proper process and track hooks.
+ static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+ static hook_t getTrackHook(int trackType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H
diff --git a/media/libaudioclient/include/AudioParameter.h b/media/libaudioclient/include/AudioParameter.h
new file mode 100644
index 0000000..1ace607
--- /dev/null
+++ b/media/libaudioclient/include/AudioParameter.h
@@ -0,0 +1,105 @@
+/*
+ * Copyright (C) 2008-2011 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_AUDIOPARAMETER_H_
+#define ANDROID_AUDIOPARAMETER_H_
+
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/String8.h>
+
+namespace android {
+
+class AudioParameter {
+
+public:
+ AudioParameter() {}
+ AudioParameter(const String8& keyValuePairs);
+ virtual ~AudioParameter();
+
+ // reserved parameter keys for changing standard parameters with setParameters() function.
+ // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input
+ // configuration changes and act accordingly.
+ // keyRouting: to change audio routing, value is an int in audio_devices_t
+ // keySamplingRate: to change sampling rate routing, value is an int
+ // keyFormat: to change audio format, value is an int in audio_format_t
+ // keyChannels: to change audio channel configuration, value is an int in audio_channels_t
+ // keyFrameCount: to change audio output frame count, value is an int
+ // keyInputSource: to change audio input source, value is an int in audio_source_t
+ // (defined in media/mediarecorder.h)
+ // keyScreenState: either "on" or "off"
+ static const char * const keyRouting;
+ static const char * const keySamplingRate;
+ static const char * const keyFormat;
+ static const char * const keyChannels;
+ static const char * const keyFrameCount;
+ static const char * const keyInputSource;
+ static const char * const keyScreenState;
+
+ // keyBtNrec: BT SCO Noise Reduction + Echo Cancellation parameters
+ // keyHwAvSync: get HW synchronization source identifier from a device
+ // keyMonoOutput: Enable mono audio playback
+ // keyStreamHwAvSync: set HW synchronization source identifier on a stream
+ static const char * const keyBtNrec;
+ static const char * const keyHwAvSync;
+ static const char * const keyMonoOutput;
+ static const char * const keyStreamHwAvSync;
+
+ // keyStreamConnect / Disconnect: value is an int in audio_devices_t
+ static const char * const keyStreamConnect;
+ static const char * const keyStreamDisconnect;
+
+ // For querying stream capabilities. All the returned values are lists.
+ // keyStreamSupportedFormats: audio_format_t
+ // keyStreamSupportedChannels: audio_channel_mask_t
+ // keyStreamSupportedSamplingRates: sampling rate values
+ static const char * const keyStreamSupportedFormats;
+ static const char * const keyStreamSupportedChannels;
+ static const char * const keyStreamSupportedSamplingRates;
+
+ static const char * const valueOn;
+ static const char * const valueOff;
+
+ static const char * const valueListSeparator;
+
+ String8 toString() const { return toStringImpl(true); }
+ String8 keysToString() const { return toStringImpl(false); }
+
+ status_t add(const String8& key, const String8& value);
+ status_t addInt(const String8& key, const int value);
+ status_t addKey(const String8& key);
+ status_t addFloat(const String8& key, const float value);
+
+ status_t remove(const String8& key);
+
+ status_t get(const String8& key, String8& value) const;
+ status_t getInt(const String8& key, int& value) const;
+ status_t getFloat(const String8& key, float& value) const;
+ status_t getAt(size_t index, String8& key) const;
+ status_t getAt(size_t index, String8& key, String8& value) const;
+
+ size_t size() const { return mParameters.size(); }
+
+private:
+ String8 mKeyValuePairs;
+ KeyedVector <String8, String8> mParameters;
+
+ String8 toStringImpl(bool useValues) const;
+};
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIOPARAMETER_H_*/
diff --git a/media/libaudioclient/include/AudioPolicy.h b/media/libaudioclient/include/AudioPolicy.h
new file mode 100644
index 0000000..8da0069
--- /dev/null
+++ b/media/libaudioclient/include/AudioPolicy.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2014 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_AUDIO_POLICY_H
+#define ANDROID_AUDIO_POLICY_H
+
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <binder/Parcel.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+// Keep in sync with AudioMix.java, AudioMixingRule.java, AudioPolicyConfig.java
+#define RULE_EXCLUSION_MASK 0x8000
+#define RULE_MATCH_ATTRIBUTE_USAGE 0x1
+#define RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET (0x1 << 1)
+#define RULE_MATCH_UID (0x1 << 2)
+#define RULE_EXCLUDE_ATTRIBUTE_USAGE (RULE_EXCLUSION_MASK|RULE_MATCH_ATTRIBUTE_USAGE)
+#define RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET \
+ (RULE_EXCLUSION_MASK|RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET)
+#define RULE_EXCLUDE_UID (RULE_EXCLUSION_MASK|RULE_MATCH_UID)
+
+#define MIX_TYPE_INVALID (-1)
+#define MIX_TYPE_PLAYERS 0
+#define MIX_TYPE_RECORDERS 1
+
+// definition of the different events that can be reported on a dynamic policy from
+// AudioSystem's implementation of the AudioPolicyClient interface
+// keep in sync with AudioSystem.java
+#define DYNAMIC_POLICY_EVENT_MIX_STATE_UPDATE 0
+
+#define MIX_STATE_DISABLED (-1)
+#define MIX_STATE_IDLE 0
+#define MIX_STATE_MIXING 1
+
+#define MIX_ROUTE_FLAG_RENDER 0x1
+#define MIX_ROUTE_FLAG_LOOP_BACK (0x1 << 1)
+#define MIX_ROUTE_FLAG_ALL (MIX_ROUTE_FLAG_RENDER | MIX_ROUTE_FLAG_LOOP_BACK)
+
+#define MAX_MIXES_PER_POLICY 10
+#define MAX_CRITERIA_PER_MIX 20
+
+class AudioMixMatchCriterion {
+public:
+ AudioMixMatchCriterion() {}
+ AudioMixMatchCriterion(audio_usage_t usage, audio_source_t source, uint32_t rule);
+
+ status_t readFromParcel(Parcel *parcel);
+ status_t writeToParcel(Parcel *parcel) const;
+
+ union {
+ audio_usage_t mUsage;
+ audio_source_t mSource;
+ uid_t mUid;
+ } mValue;
+ uint32_t mRule;
+};
+
+class AudioMix {
+public:
+ // flag on an AudioMix indicating the activity on this mix (IDLE, MIXING)
+ // must be reported through the AudioPolicyClient interface
+ static const uint32_t kCbFlagNotifyActivity = 0x1;
+
+ AudioMix() {}
+ AudioMix(Vector<AudioMixMatchCriterion> criteria, uint32_t mixType, audio_config_t format,
+ uint32_t routeFlags, String8 registrationId, uint32_t flags) :
+ mCriteria(criteria), mMixType(mixType), mFormat(format),
+ mRouteFlags(routeFlags), mDeviceAddress(registrationId), mCbFlags(flags){}
+
+ status_t readFromParcel(Parcel *parcel);
+ status_t writeToParcel(Parcel *parcel) const;
+
+ Vector<AudioMixMatchCriterion> mCriteria;
+ uint32_t mMixType;
+ audio_config_t mFormat;
+ uint32_t mRouteFlags;
+ audio_devices_t mDeviceType;
+ String8 mDeviceAddress;
+ uint32_t mCbFlags; // flags indicating which callbacks to use, see kCbFlag*
+};
+
+
+// definitions for audio recording configuration updates
+// which update type is reported
+#define RECORD_CONFIG_EVENT_NONE -1
+#define RECORD_CONFIG_EVENT_START 1
+#define RECORD_CONFIG_EVENT_STOP 0
+
+}; // namespace android
+
+#endif // ANDROID_AUDIO_POLICY_H
diff --git a/media/libaudioclient/include/AudioPolicyHelper.h b/media/libaudioclient/include/AudioPolicyHelper.h
new file mode 100644
index 0000000..04f6a20
--- /dev/null
+++ b/media/libaudioclient/include/AudioPolicyHelper.h
@@ -0,0 +1,117 @@
+/*
+ * Copyright (C) 2014 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 AUDIO_POLICY_HELPER_H_
+#define AUDIO_POLICY_HELPER_H_
+
+#include <system/audio.h>
+
+static audio_stream_type_t audio_attributes_to_stream_type(const audio_attributes_t *attr)
+{
+ // flags to stream type mapping
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return AUDIO_STREAM_ENFORCED_AUDIBLE;
+ }
+ if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+ return AUDIO_STREAM_BLUETOOTH_SCO;
+ }
+
+ // usage to stream type mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANT:
+ return AUDIO_STREAM_MUSIC;
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ return AUDIO_STREAM_ACCESSIBILITY;
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return AUDIO_STREAM_SYSTEM;
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return AUDIO_STREAM_VOICE_CALL;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return AUDIO_STREAM_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ return AUDIO_STREAM_ALARM;
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return AUDIO_STREAM_RING;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return AUDIO_STREAM_NOTIFICATION;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return AUDIO_STREAM_MUSIC;
+ }
+}
+
+static void stream_type_to_audio_attributes(audio_stream_type_t streamType,
+ audio_attributes_t *attr) {
+ memset(attr, 0, sizeof(audio_attributes_t));
+
+ switch (streamType) {
+ case AUDIO_STREAM_DEFAULT:
+ case AUDIO_STREAM_MUSIC:
+ attr->content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ attr->usage = AUDIO_USAGE_MEDIA;
+ break;
+ case AUDIO_STREAM_VOICE_CALL:
+ attr->content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ attr->usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+ break;
+ case AUDIO_STREAM_ENFORCED_AUDIBLE:
+ attr->flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED;
+ // intended fall through, attributes in common with STREAM_SYSTEM
+ case AUDIO_STREAM_SYSTEM:
+ attr->content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ attr->usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_RING:
+ attr->content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ attr->usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+ break;
+ case AUDIO_STREAM_ALARM:
+ attr->content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ attr->usage = AUDIO_USAGE_ALARM;
+ break;
+ case AUDIO_STREAM_NOTIFICATION:
+ attr->content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ attr->usage = AUDIO_USAGE_NOTIFICATION;
+ break;
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ attr->content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ attr->usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+ attr->flags |= AUDIO_FLAG_SCO;
+ break;
+ case AUDIO_STREAM_DTMF:
+ attr->content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ attr->usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+ break;
+ case AUDIO_STREAM_TTS:
+ attr->content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ attr->usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+ break;
+ default:
+ ALOGE("invalid stream type %d when converting to attributes", streamType);
+ }
+}
+
+#endif //AUDIO_POLICY_HELPER_H_
diff --git a/media/libaudioclient/include/AudioRecord.h b/media/libaudioclient/include/AudioRecord.h
new file mode 100644
index 0000000..1c8746f
--- /dev/null
+++ b/media/libaudioclient/include/AudioRecord.h
@@ -0,0 +1,659 @@
+/*
+ * Copyright (C) 2008 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_AUDIORECORD_H
+#define ANDROID_AUDIORECORD_H
+
+#include <cutils/sched_policy.h>
+#include <media/AudioSystem.h>
+#include <media/AudioTimestamp.h>
+#include <media/IAudioRecord.h>
+#include <media/Modulo.h>
+#include <utils/threads.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+struct audio_track_cblk_t;
+class AudioRecordClientProxy;
+
+// ----------------------------------------------------------------------------
+
+class AudioRecord : public RefBase
+{
+public:
+
+ /* Events used by AudioRecord callback function (callback_t).
+ * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
+ */
+ enum event_type {
+ EVENT_MORE_DATA = 0, // Request to read available data from buffer.
+ // If this event is delivered but the callback handler
+ // does not want to read the available data, the handler must
+ // explicitly ignore the event by setting frameCount to zero.
+ EVENT_OVERRUN = 1, // Buffer overrun occurred.
+ EVENT_MARKER = 2, // Record head is at the specified marker position
+ // (See setMarkerPosition()).
+ EVENT_NEW_POS = 3, // Record head is at a new position
+ // (See setPositionUpdatePeriod()).
+ EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and
+ // voluntary invalidation by mediaserver, or mediaserver crash.
+ };
+
+ /* Client should declare a Buffer and pass 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
+ // frames to be read
+ // 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
+ // bytes to be read, 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 AudioRecord::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 AudioRecord::Buffer struct. The callback must not read
+ * more bytes than indicated by 'size' field and update 'size' if
+ * fewer bytes are consumed.
+ * - EVENT_OVERRUN: unused.
+ * - 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_NEW_IAUDIORECORD: unused.
+ */
+
+ typedef void (*callback_t)(int event, void* user, void *info);
+
+ /* Returns the minimum frame count required for the successful creation of
+ * an AudioRecord 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,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask);
+
+ /* How data is transferred from AudioRecord
+ */
+ 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 read()
+ };
+
+ /* Constructs an uninitialized AudioRecord. No connection with
+ * AudioFlinger takes place. Use set() after this.
+ *
+ * Parameters:
+ *
+ * opPackageName: The package name used for app ops.
+ */
+ AudioRecord(const String16& opPackageName);
+
+ /* Creates an AudioRecord 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:
+ *
+ * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT).
+ * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate.
+ * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
+ * 16 bits per sample).
+ * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true.
+ * opPackageName: The package name used for app ops.
+ * 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 AudioRecord could
+ * be larger if the requested size is not compatible with current audio HAL
+ * latency. Zero means to use a default value.
+ * cbf: Callback function. If not null, this function is called periodically
+ * to consume 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 are ready in record track output buffer.
+ * sessionId: Not yet supported.
+ * transferType: How data is transferred from AudioRecord.
+ * flags: See comments on audio_input_flags_t in <system/audio.h>
+ * pAttributes: If not NULL, supersedes inputSource for use case selection.
+ * threadCanCallJava: Not present in parameter list, and so is fixed at false.
+ */
+
+ AudioRecord(audio_source_t inputSource,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ const String16& opPackageName,
+ size_t frameCount = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ uint32_t notificationFrames = 0,
+ audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
+ transfer_type transferType = TRANSFER_DEFAULT,
+ audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
+ uid_t uid = AUDIO_UID_INVALID,
+ pid_t pid = -1,
+ const audio_attributes_t* pAttributes = NULL);
+
+ /* Terminates the AudioRecord and unregisters it from AudioFlinger.
+ * Also destroys all resources associated with the AudioRecord.
+ */
+protected:
+ virtual ~AudioRecord();
+public:
+
+ /* Initialize an AudioRecord that was created using the AudioRecord() constructor.
+ * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters.
+ * set() is not multi-thread safe.
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful intialization
+ * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
+ * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
+ * - NO_INIT: audio server or audio hardware not initialized
+ * - PERMISSION_DENIED: recording is not allowed for the requesting process
+ * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
+ *
+ * Parameters not listed in the AudioRecord constructors above:
+ *
+ * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI.
+ */
+ status_t set(audio_source_t inputSource,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ uint32_t notificationFrames = 0,
+ bool threadCanCallJava = false,
+ audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
+ transfer_type transferType = TRANSFER_DEFAULT,
+ audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
+ uid_t uid = AUDIO_UID_INVALID,
+ pid_t pid = -1,
+ const audio_attributes_t* pAttributes = NULL);
+
+ /* Result of constructing the AudioRecord. This must be checked for successful initialization
+ * before using any AudioRecord API (except for set()), because using
+ * an uninitialized AudioRecord 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 AudioRecord buffer size, resampling if applicable,
+ * and audio hardware driver.
+ */
+ uint32_t latency() const { return mLatency; }
+
+ /* getters, see constructor and set() */
+
+ audio_format_t format() const { return mFormat; }
+ uint32_t channelCount() const { return mChannelCount; }
+ size_t frameCount() const { return mFrameCount; }
+ size_t frameSize() const { return mFrameSize; }
+ audio_source_t inputSource() const { return mAttributes.source; }
+
+ /* After it's created the track is not active. Call start() to
+ * make it active. If set, the callback will start being called.
+ * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until
+ * the specified event occurs on the specified trigger session.
+ */
+ status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+ audio_session_t triggerSession = AUDIO_SESSION_NONE);
+
+ /* Stop a track. The callback will cease being called. Note that obtainBuffer() still
+ * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
+ */
+ void stop();
+ bool stopped() const;
+
+ /* Return the sink sample rate for this record track in Hz.
+ * If specified as zero in constructor or set(), this will be the source sample rate.
+ * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock.
+ */
+ uint32_t getSampleRate() const { return mSampleRate; }
+
+ /* Sets marker position. When record 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 AudioRecord 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 AudioRecord 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 recorded,
+ * a callback with event type EVENT_NEW_POS is called.
+ * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
+ * callback.
+ * If the AudioRecord 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 AudioRecord has no callback installed.
+ */
+ status_t setPositionUpdatePeriod(uint32_t updatePeriod);
+ status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
+
+ /* Return the total number of frames recorded since recording started.
+ * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz.
+ * It is reset to zero by stop().
+ *
+ * Parameters:
+ *
+ * position: Address where to return record 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) const;
+
+ /* Return the record timestamp.
+ *
+ * Parameters:
+ * timestamp: A pointer to the timestamp to be filled.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - BAD_VALUE: timestamp is NULL
+ */
+ status_t getTimestamp(ExtendedTimestamp *timestamp);
+
+ /* Returns a handle on the audio input used by this AudioRecord.
+ *
+ * Parameters:
+ * none.
+ *
+ * Returned value:
+ * handle on audio hardware input
+ */
+// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp
+ audio_io_handle_t getInput() const __attribute__((__deprecated__))
+ { return getInputPrivate(); }
+private:
+ audio_io_handle_t getInputPrivate() const;
+public:
+
+ /* Returns the audio session ID associated with this AudioRecord.
+ *
+ * Parameters:
+ * none.
+ *
+ * Returned value:
+ * AudioRecord session ID.
+ *
+ * No lock needed because session ID doesn't change after first set().
+ */
+ audio_session_t getSessionId() const { return mSessionId; }
+
+ /* Public API for TRANSFER_OBTAIN mode.
+ * Obtains a buffer of up to "audioBuffer->frameCount" full frames.
+ * After draining these frames of data, the caller should release them with releaseBuffer().
+ * If the track buffer is not empty, obtainBuffer() returns as many contiguous
+ * full 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 empty and track is stopped, obtainBuffer() returns WOULD_BLOCK
+ * regardless of the value of waitCount.
+ * If the track buffer is empty 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 frames requested
+ * size ignored
+ * raw ignored
+ * After error return:
+ * frameCount 0
+ * size 0
+ * raw undefined
+ * After successful return:
+ * frameCount actual number of 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);
+
+ // Explicit Routing
+ /**
+ * TODO Document this method.
+ */
+ status_t setInputDevice(audio_port_handle_t deviceId);
+
+ /**
+ * TODO Document this method.
+ */
+ audio_port_handle_t getInputDevice();
+
+ /* Returns the ID of the audio device actually used by the input to which this AudioRecord
+ * is attached.
+ * A value of AUDIO_PORT_HANDLE_NONE indicates the AudioRecord is not attached to any input.
+ *
+ * Parameters:
+ * none.
+ */
+ audio_port_handle_t getRoutedDeviceId();
+
+ /* Add an AudioDeviceCallback. The caller will be notified when the audio device
+ * to which this AudioRecord 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);
+
+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 an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill.
+ *
+ * Buffer fields:
+ * frameCount currently ignored but recommend to set to actual number of frames consumed
+ * size actual number of bytes consumed, must be multiple of frameSize
+ * raw ignored
+ */
+ void releaseBuffer(const Buffer* audioBuffer);
+
+ /* As a convenience we provide a read() 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 read >= 0,
+ * or one of the following negative status codes:
+ * INVALID_OPERATION AudioRecord is configured for streaming mode
+ * BAD_VALUE size is invalid
+ * WOULD_BLOCK when obtainBuffer() returns same, or
+ * AudioRecord was stopped during the read
+ * or any other error code returned by IAudioRecord::start() or restoreRecord_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 read
+ * the full content of the buffer.
+ */
+ ssize_t read(void* buffer, size_t size, bool blocking = true);
+
+ /* Return the number of input frames lost in the audio driver since the last call of this
+ * function. Audio driver is expected to reset the value to 0 and restart counting upon
+ * returning the current value by this function call. Such loss typically occurs when the
+ * user space process is blocked longer than the capacity of audio driver buffers.
+ * Units: the number of input audio frames.
+ * FIXME The side-effect of resetting the counter may be incompatible with multi-client.
+ * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
+ */
+ uint32_t getInputFramesLost() const;
+
+ /* Get the flags */
+ audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; }
+
+private:
+ /* copying audio record objects is not allowed */
+ AudioRecord(const AudioRecord& other);
+ AudioRecord& operator = (const AudioRecord& other);
+
+ /* a small internal class to handle the callback */
+ class AudioRecordThread : public Thread
+ {
+ public:
+ AudioRecordThread(AudioRecord& 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 AudioRecord;
+ virtual bool threadLoop();
+ AudioRecord& mReceiver;
+ virtual ~AudioRecordThread();
+ 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 AudioRecordThread::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 openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName);
+
+ // FIXME enum is faster than strcmp() for parameter 'from'
+ status_t restoreRecord_l(const char *from);
+
+ sp<AudioRecordThread> mAudioRecordThread;
+ mutable Mutex mLock;
+
+ // Current client state: false = stopped, true = active. Protected by mLock. If more states
+ // are added, consider changing this to enum State { ... } mState as in AudioTrack.
+ bool mActive;
+
+ // for client callback handler
+ callback_t mCbf; // callback handler for events, or NULL
+ void* mUserData;
+
+ // for notification APIs
+ uint32_t mNotificationFramesReq; // requested number of frames between each
+ // notification callback
+ // as specified in constructor or set()
+ uint32_t mNotificationFramesAct; // actual number of frames between each
+ // notification callback
+ bool mRefreshRemaining; // processAudioBuffer() should refresh
+ // mRemainingFrames and mRetryOnPartialBuffer
+
+ // 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
+
+ status_t mStatus;
+
+ String16 mOpPackageName; // The package name used for app ops.
+
+ size_t mFrameCount; // corresponds to current IAudioRecord, value is
+ // reported back by AudioFlinger to the client
+ size_t mReqFrameCount; // frame count to request the first or next time
+ // a new IAudioRecord is needed, non-decreasing
+
+ int64_t mFramesRead; // total frames read. reset to zero after
+ // the start() following stop(). It is not
+ // changed after restoring the track.
+ int64_t mFramesReadServerOffset; // An offset to server frames read due to
+ // restoring AudioRecord, or stop/start.
+ // constant after constructor or set()
+ uint32_t mSampleRate;
+ audio_format_t mFormat;
+ uint32_t mChannelCount;
+ size_t mFrameSize; // app-level frame size == AudioFlinger frame size
+ uint32_t mLatency; // in ms
+ audio_channel_mask_t mChannelMask;
+
+ audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may
+ // be denied by client or server, such as
+ // AUDIO_INPUT_FLAG_FAST. mLock must be
+ // held to read or write those bits reliably.
+ audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const
+
+ audio_session_t mSessionId;
+ transfer_type mTransfer;
+
+ // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0
+ // provided the initial set() was successful
+ sp<IAudioRecord> mAudioRecord;
+ sp<IMemory> mCblkMemory;
+ audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ sp<IMemory> mBufferMemory;
+ audio_io_handle_t mInput; // returned by AudioSystem::getInput()
+
+ 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.
+ // 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<AudioRecordClientProxy> mProxy;
+
+ bool mInOverrun; // whether recorder is currently in overrun state
+
+private:
+ class DeathNotifier : public IBinder::DeathRecipient {
+ public:
+ DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { }
+ protected:
+ virtual void binderDied(const wp<IBinder>& who);
+ private:
+ const wp<AudioRecord> mAudioRecord;
+ };
+
+ sp<DeathNotifier> mDeathNotifier;
+ uint32_t mSequence; // incremented for each new IAudioRecord attempt
+ uid_t mClientUid;
+ pid_t mClientPid;
+ audio_attributes_t mAttributes;
+
+ // For Device Selection API
+ // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing.
+ audio_port_handle_t mSelectedDeviceId;
+ sp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
+ audio_port_handle_t mPortId; // unique ID allocated by audio policy
+
+};
+
+}; // namespace android
+
+#endif // ANDROID_AUDIORECORD_H
diff --git a/media/libaudioclient/include/AudioSystem.h b/media/libaudioclient/include/AudioSystem.h
new file mode 100644
index 0000000..853d318
--- /dev/null
+++ b/media/libaudioclient/include/AudioSystem.h
@@ -0,0 +1,475 @@
+/*
+ * Copyright (C) 2008 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_AUDIOSYSTEM_H_
+#define ANDROID_AUDIOSYSTEM_H_
+
+#include <sys/types.h>
+
+#include <media/AudioPolicy.h>
+#include <media/AudioIoDescriptor.h>
+#include <media/IAudioFlingerClient.h>
+#include <media/IAudioPolicyServiceClient.h>
+#include <system/audio.h>
+#include <system/audio_effect.h>
+#include <system/audio_policy.h>
+#include <utils/Errors.h>
+#include <utils/Mutex.h>
+
+namespace android {
+
+typedef void (*audio_error_callback)(status_t err);
+typedef void (*dynamic_policy_callback)(int event, String8 regId, int val);
+typedef void (*record_config_callback)(int event, audio_session_t session, int source,
+ const audio_config_base_t *clientConfig, const audio_config_base_t *deviceConfig,
+ audio_patch_handle_t patchHandle);
+
+class IAudioFlinger;
+class IAudioPolicyService;
+class String8;
+
+class AudioSystem
+{
+public:
+
+ // FIXME Declare in binder opcode order, similarly to IAudioFlinger.h and IAudioFlinger.cpp
+
+ /* These are static methods to control the system-wide AudioFlinger
+ * only privileged processes can have access to them
+ */
+
+ // mute/unmute microphone
+ static status_t muteMicrophone(bool state);
+ static status_t isMicrophoneMuted(bool *state);
+
+ // set/get master volume
+ static status_t setMasterVolume(float value);
+ static status_t getMasterVolume(float* volume);
+
+ // mute/unmute audio outputs
+ static status_t setMasterMute(bool mute);
+ static status_t getMasterMute(bool* mute);
+
+ // set/get stream volume on specified output
+ static status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output);
+ static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
+ audio_io_handle_t output);
+
+ // mute/unmute stream
+ static status_t setStreamMute(audio_stream_type_t stream, bool mute);
+ static status_t getStreamMute(audio_stream_type_t stream, bool* mute);
+
+ // set audio mode in audio hardware
+ static status_t setMode(audio_mode_t mode);
+
+ // returns true in *state if tracks are active on the specified stream or have been active
+ // in the past inPastMs milliseconds
+ static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
+ // returns true in *state if tracks are active for what qualifies as remote playback
+ // on the specified stream or have been active in the past inPastMs milliseconds. Remote
+ // playback isn't mutually exclusive with local playback.
+ static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
+ uint32_t inPastMs);
+ // returns true in *state if a recorder is currently recording with the specified source
+ static status_t isSourceActive(audio_source_t source, bool *state);
+
+ // set/get audio hardware parameters. The function accepts a list of parameters
+ // key value pairs in the form: key1=value1;key2=value2;...
+ // Some keys are reserved for standard parameters (See AudioParameter class).
+ // The versions with audio_io_handle_t are intended for internal media framework use only.
+ static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
+ static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
+ // The versions without audio_io_handle_t are intended for JNI.
+ static status_t setParameters(const String8& keyValuePairs);
+ static String8 getParameters(const String8& keys);
+
+ static void setErrorCallback(audio_error_callback cb);
+ static void setDynPolicyCallback(dynamic_policy_callback cb);
+ static void setRecordConfigCallback(record_config_callback);
+
+ // helper function to obtain AudioFlinger service handle
+ static const sp<IAudioFlinger> get_audio_flinger();
+
+ static float linearToLog(int volume);
+ static int logToLinear(float volume);
+
+ // Returned samplingRate and frameCount output values are guaranteed
+ // to be non-zero if status == NO_ERROR
+ // FIXME This API assumes a route, and so should be deprecated.
+ static status_t getOutputSamplingRate(uint32_t* samplingRate,
+ audio_stream_type_t stream);
+ // FIXME This API assumes a route, and so should be deprecated.
+ static status_t getOutputFrameCount(size_t* frameCount,
+ audio_stream_type_t stream);
+ // FIXME This API assumes a route, and so should be deprecated.
+ static status_t getOutputLatency(uint32_t* latency,
+ audio_stream_type_t stream);
+ // returns the audio HAL sample rate
+ static status_t getSamplingRate(audio_io_handle_t ioHandle,
+ uint32_t* samplingRate);
+ // For output threads with a fast mixer, returns the number of frames per normal mixer buffer.
+ // For output threads without a fast mixer, or for input, this is same as getFrameCountHAL().
+ static status_t getFrameCount(audio_io_handle_t ioHandle,
+ size_t* frameCount);
+ // returns the audio output latency in ms. Corresponds to
+ // audio_stream_out->get_latency()
+ static status_t getLatency(audio_io_handle_t output,
+ uint32_t* latency);
+
+ // return status NO_ERROR implies *buffSize > 0
+ // FIXME This API assumes a route, and so should deprecated.
+ static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask, size_t* buffSize);
+
+ static status_t setVoiceVolume(float volume);
+
+ // return the number of audio frames written by AudioFlinger to audio HAL and
+ // audio dsp to DAC since the specified output has exited standby.
+ // returned status (from utils/Errors.h) can be:
+ // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
+ // - INVALID_OPERATION: Not supported on current hardware platform
+ // - BAD_VALUE: invalid parameter
+ // NOTE: this feature is not supported on all hardware platforms and it is
+ // necessary to check returned status before using the returned values.
+ static status_t getRenderPosition(audio_io_handle_t output,
+ uint32_t *halFrames,
+ uint32_t *dspFrames);
+
+ // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
+ static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
+
+ // Allocate a new unique ID for use as an audio session ID or I/O handle.
+ // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead.
+ // FIXME If AudioFlinger were to ever exhaust the unique ID namespace,
+ // this method could fail by returning either a reserved ID like AUDIO_UNIQUE_ID_ALLOCATE
+ // or an unspecified existing unique ID.
+ static audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use);
+
+ static void acquireAudioSessionId(audio_session_t audioSession, pid_t pid);
+ static void releaseAudioSessionId(audio_session_t audioSession, pid_t pid);
+
+ // Get the HW synchronization source used for an audio session.
+ // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs
+ // or no HW sync source is used.
+ static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId);
+
+ // Indicate JAVA services are ready (scheduling, power management ...)
+ static status_t systemReady();
+
+ // Returns the number of frames per audio HAL buffer.
+ // Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input.
+ // See also getFrameCount().
+ static status_t getFrameCountHAL(audio_io_handle_t ioHandle,
+ size_t* frameCount);
+
+ // Events used to synchronize actions between audio sessions.
+ // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until
+ // playback is complete on another audio session.
+ // See definitions in MediaSyncEvent.java
+ enum sync_event_t {
+ SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event
+ SYNC_EVENT_NONE = 0,
+ SYNC_EVENT_PRESENTATION_COMPLETE,
+
+ //
+ // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ...
+ //
+ SYNC_EVENT_CNT,
+ };
+
+ // Timeout for synchronous record start. Prevents from blocking the record thread forever
+ // if the trigger event is not fired.
+ static const uint32_t kSyncRecordStartTimeOutMs = 30000;
+
+ //
+ // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
+ //
+ static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
+ const char *device_address, const char *device_name);
+ static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+ const char *device_address);
+ static status_t handleDeviceConfigChange(audio_devices_t device,
+ const char *device_address,
+ const char *device_name);
+ static status_t setPhoneState(audio_mode_t state);
+ static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
+ static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
+ // or release it with releaseOutput().
+ static audio_io_handle_t getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
+ static status_t getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uid_t uid,
+ const audio_config_t *config,
+ audio_output_flags_t flags,
+ audio_port_handle_t selectedDeviceId,
+ audio_port_handle_t *portId);
+ static status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ static status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ static void releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
+ // or release it with releaseInput().
+ static status_t getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ pid_t pid,
+ uid_t uid,
+ const audio_config_base_t *config,
+ audio_input_flags_t flags,
+ audio_port_handle_t selectedDeviceId,
+ audio_port_handle_t *portId);
+
+ static status_t startInput(audio_io_handle_t input,
+ audio_session_t session);
+ static status_t stopInput(audio_io_handle_t input,
+ audio_session_t session);
+ static void releaseInput(audio_io_handle_t input,
+ audio_session_t session);
+ static status_t initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax);
+ static status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ static status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
+
+ static uint32_t getStrategyForStream(audio_stream_type_t stream);
+ static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+ static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
+ static status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ audio_session_t session,
+ int id);
+ static status_t unregisterEffect(int id);
+ static status_t setEffectEnabled(int id, bool enabled);
+
+ // clear stream to output mapping cache (gStreamOutputMap)
+ // and output configuration cache (gOutputs)
+ static void clearAudioConfigCache();
+
+ static const sp<IAudioPolicyService> get_audio_policy_service();
+
+ // helpers for android.media.AudioManager.getProperty(), see description there for meaning
+ static uint32_t getPrimaryOutputSamplingRate();
+ static size_t getPrimaryOutputFrameCount();
+
+ static status_t setLowRamDevice(bool isLowRamDevice);
+
+ // Check if hw offload is possible for given format, stream type, sample rate,
+ // bit rate, duration, video and streaming or offload property is enabled
+ static bool isOffloadSupported(const audio_offload_info_t& info);
+
+ // check presence of audio flinger service.
+ // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise
+ static status_t checkAudioFlinger();
+
+ /* List available audio ports and their attributes */
+ static status_t listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation);
+
+ /* Get attributes for a given audio port */
+ static status_t getAudioPort(struct audio_port *port);
+
+ /* Create an audio patch between several source and sink ports */
+ static status_t createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle);
+
+ /* Release an audio patch */
+ static status_t releaseAudioPatch(audio_patch_handle_t handle);
+
+ /* List existing audio patches */
+ static status_t listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation);
+ /* Set audio port configuration */
+ static status_t setAudioPortConfig(const struct audio_port_config *config);
+
+
+ static status_t acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device);
+ static status_t releaseSoundTriggerSession(audio_session_t session);
+
+ static audio_mode_t getPhoneState();
+
+ static status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration);
+
+ static status_t startAudioSource(const struct audio_port_config *source,
+ const audio_attributes_t *attributes,
+ audio_patch_handle_t *handle);
+ static status_t stopAudioSource(audio_patch_handle_t handle);
+
+ static status_t setMasterMono(bool mono);
+ static status_t getMasterMono(bool *mono);
+
+ // ----------------------------------------------------------------------------
+
+ class AudioPortCallback : public RefBase
+ {
+ public:
+
+ AudioPortCallback() {}
+ virtual ~AudioPortCallback() {}
+
+ virtual void onAudioPortListUpdate() = 0;
+ virtual void onAudioPatchListUpdate() = 0;
+ virtual void onServiceDied() = 0;
+
+ };
+
+ static status_t addAudioPortCallback(const sp<AudioPortCallback>& callback);
+ static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callback);
+
+ class AudioDeviceCallback : public RefBase
+ {
+ public:
+
+ AudioDeviceCallback() {}
+ virtual ~AudioDeviceCallback() {}
+
+ virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo,
+ audio_port_handle_t deviceId) = 0;
+ };
+
+ static status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
+ audio_io_handle_t audioIo);
+ static status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
+ audio_io_handle_t audioIo);
+
+ static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
+
+private:
+
+ class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
+ {
+ public:
+ AudioFlingerClient() :
+ mInBuffSize(0), mInSamplingRate(0),
+ mInFormat(AUDIO_FORMAT_DEFAULT), mInChannelMask(AUDIO_CHANNEL_NONE) {
+ }
+
+ void clearIoCache();
+ status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask, size_t* buffSize);
+ sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle);
+
+ // DeathRecipient
+ virtual void binderDied(const wp<IBinder>& who);
+
+ // IAudioFlingerClient
+
+ // indicate a change in the configuration of an output or input: keeps the cached
+ // values for output/input parameters up-to-date in client process
+ virtual void ioConfigChanged(audio_io_config_event event,
+ const sp<AudioIoDescriptor>& ioDesc);
+
+
+ status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
+ audio_io_handle_t audioIo);
+ status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
+ audio_io_handle_t audioIo);
+
+ audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
+
+ private:
+ Mutex mLock;
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioIoDescriptor> > mIoDescriptors;
+ DefaultKeyedVector<audio_io_handle_t, Vector < sp<AudioDeviceCallback> > >
+ mAudioDeviceCallbacks;
+ // cached values for recording getInputBufferSize() queries
+ size_t mInBuffSize; // zero indicates cache is invalid
+ uint32_t mInSamplingRate;
+ audio_format_t mInFormat;
+ audio_channel_mask_t mInChannelMask;
+ sp<AudioIoDescriptor> getIoDescriptor_l(audio_io_handle_t ioHandle);
+ };
+
+ class AudioPolicyServiceClient: public IBinder::DeathRecipient,
+ public BnAudioPolicyServiceClient
+ {
+ public:
+ AudioPolicyServiceClient() {
+ }
+
+ int addAudioPortCallback(const sp<AudioPortCallback>& callback);
+ int removeAudioPortCallback(const sp<AudioPortCallback>& callback);
+
+ // DeathRecipient
+ virtual void binderDied(const wp<IBinder>& who);
+
+ // IAudioPolicyServiceClient
+ virtual void onAudioPortListUpdate();
+ virtual void onAudioPatchListUpdate();
+ virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state);
+ virtual void onRecordingConfigurationUpdate(int event, audio_session_t session,
+ audio_source_t source, const audio_config_base_t *clientConfig,
+ const audio_config_base_t *deviceConfig, audio_patch_handle_t patchHandle);
+
+ private:
+ Mutex mLock;
+ Vector <sp <AudioPortCallback> > mAudioPortCallbacks;
+ };
+
+ static const sp<AudioFlingerClient> getAudioFlingerClient();
+ static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle);
+
+ static sp<AudioFlingerClient> gAudioFlingerClient;
+ static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
+ friend class AudioFlingerClient;
+ friend class AudioPolicyServiceClient;
+
+ static Mutex gLock; // protects gAudioFlinger and gAudioErrorCallback,
+ static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient
+ static sp<IAudioFlinger> gAudioFlinger;
+ static audio_error_callback gAudioErrorCallback;
+ static dynamic_policy_callback gDynPolicyCallback;
+ static record_config_callback gRecordConfigCallback;
+
+ static size_t gInBuffSize;
+ // previous parameters for recording buffer size queries
+ static uint32_t gPrevInSamplingRate;
+ static audio_format_t gPrevInFormat;
+ static audio_channel_mask_t gPrevInChannelMask;
+
+ static sp<IAudioPolicyService> gAudioPolicyService;
+};
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIOSYSTEM_H_*/
diff --git a/media/libaudioclient/include/AudioTimestamp.h b/media/libaudioclient/include/AudioTimestamp.h
new file mode 100644
index 0000000..498de8e
--- /dev/null
+++ b/media/libaudioclient/include/AudioTimestamp.h
@@ -0,0 +1,156 @@
+/*
+ * Copyright (C) 2013 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_AUDIO_TIMESTAMP_H
+#define ANDROID_AUDIO_TIMESTAMP_H
+
+#include <string>
+#include <sstream>
+#include <time.h>
+
+namespace android {
+
+class AudioTimestamp {
+public:
+ AudioTimestamp() : mPosition(0) {
+ mTime.tv_sec = 0;
+ mTime.tv_nsec = 0;
+ }
+ // FIXME change type to match android.media.AudioTrack
+ uint32_t mPosition; // a frame position in AudioTrack::getPosition() units
+ struct timespec mTime; // corresponding CLOCK_MONOTONIC when frame is expected to present
+};
+
+struct alignas(8) /* bug 29096183, bug 29108507 */ ExtendedTimestamp {
+ enum Location {
+ LOCATION_INVALID = -1,
+ // Locations in the audio playback / record pipeline.
+ LOCATION_CLIENT, // timestamp of last read frame from client-server track buffer.
+ LOCATION_SERVER, // timestamp of newest frame from client-server track buffer.
+ LOCATION_KERNEL, // timestamp of newest frame in the kernel (alsa) buffer.
+
+ // Historical data: info when the kernel timestamp was OK (prior to the newest frame).
+ // This may be useful when the newest frame kernel timestamp is unavailable.
+ // Available for playback timestamps.
+ LOCATION_SERVER_LASTKERNELOK, // timestamp of server the prior time kernel timestamp OK.
+ LOCATION_KERNEL_LASTKERNELOK, // timestamp of kernel the prior time kernel timestamp OK.
+ LOCATION_MAX // for sizing arrays only
+ };
+
+ // This needs to be kept in sync with android.media.AudioTimestamp
+ enum Timebase {
+ TIMEBASE_MONOTONIC, // Clock monotonic offset (generally 0)
+ TIMEBASE_BOOTTIME,
+ TIMEBASE_MAX,
+ };
+
+ ExtendedTimestamp() {
+ clear();
+ }
+
+ // mPosition is expressed in frame units.
+ // It is generally nonnegative, though we keep this signed for
+ // to potentially express algorithmic latency at the start of the stream
+ // and to prevent unintentional unsigned integer underflow.
+ int64_t mPosition[LOCATION_MAX];
+
+ // mTimeNs is in nanoseconds for the default timebase, monotonic.
+ // If this value is -1, then both time and position are invalid.
+ // If this value is 0, then the time is not valid but the position is valid.
+ int64_t mTimeNs[LOCATION_MAX];
+
+ // mTimebaseOffset is the offset in ns from monotonic when the
+ // timestamp was taken. This may vary due to suspend time
+ // or NTP adjustment.
+ int64_t mTimebaseOffset[TIMEBASE_MAX];
+
+ // Playback only:
+ // mFlushed is number of flushed frames before entering the server mix;
+ // hence not included in mPosition. This is used for adjusting server positions
+ // information for frames "dropped".
+ // FIXME: This variable should be eliminated, with the offset added on the server side
+ // before sending to client, but differences in legacy position offset handling
+ // and new extended timestamps require this to be maintained as a separate quantity.
+ int64_t mFlushed;
+
+ // Call to reset the timestamp to the original (invalid) state
+ void clear() {
+ memset(mPosition, 0, sizeof(mPosition)); // actually not necessary if time is -1
+ for (int i = 0; i < LOCATION_MAX; ++i) {
+ mTimeNs[i] = -1;
+ }
+ memset(mTimebaseOffset, 0, sizeof(mTimebaseOffset));
+ mFlushed = 0;
+ }
+
+ // Returns the best timestamp as judged from the closest-to-hw stage in the
+ // pipeline with a valid timestamp. If the optional location parameter is non-null,
+ // it will be filled with the location where the time was obtained.
+ status_t getBestTimestamp(
+ int64_t *position, int64_t *time, int timebase, Location *location = nullptr) const {
+ if (position == nullptr || time == nullptr
+ || timebase < 0 || timebase >= TIMEBASE_MAX) {
+ return BAD_VALUE;
+ }
+ // look for the closest-to-hw stage in the pipeline with a valid timestamp.
+ // We omit LOCATION_CLIENT as we prefer at least LOCATION_SERVER based accuracy
+ // when getting the best timestamp.
+ for (int i = LOCATION_KERNEL; i >= LOCATION_SERVER; --i) {
+ if (mTimeNs[i] > 0) {
+ *position = mPosition[i];
+ *time = mTimeNs[i] + mTimebaseOffset[timebase];
+ if (location != nullptr) {
+ *location = (Location)i;
+ }
+ return OK;
+ }
+ }
+ return INVALID_OPERATION;
+ }
+
+ status_t getBestTimestamp(AudioTimestamp *timestamp, Location *location = nullptr) const {
+ if (timestamp == nullptr) {
+ return BAD_VALUE;
+ }
+ int64_t position, time;
+ if (getBestTimestamp(&position, &time, TIMEBASE_MONOTONIC, location) == OK) {
+ timestamp->mPosition = position;
+ timestamp->mTime.tv_sec = time / 1000000000;
+ timestamp->mTime.tv_nsec = time - timestamp->mTime.tv_sec * 1000000000LL;
+ return OK;
+ }
+ return INVALID_OPERATION;
+ }
+
+ // convert fields to a printable string
+ std::string toString() {
+ std::stringstream ss;
+
+ ss << "BOOTTIME offset " << mTimebaseOffset[TIMEBASE_BOOTTIME] << "\n";
+ for (int i = 0; i < LOCATION_MAX; ++i) {
+ ss << "ExtendedTimestamp[" << i << "] position: "
+ << mPosition[i] << " time: " << mTimeNs[i] << "\n";
+ }
+ return ss.str();
+ }
+ // TODO:
+ // Consider adding buffer status:
+ // size, available, algorithmic latency
+};
+
+} // namespace
+
+#endif // ANDROID_AUDIO_TIMESTAMP_H
diff --git a/media/libaudioclient/include/AudioTrack.h b/media/libaudioclient/include/AudioTrack.h
new file mode 100644
index 0000000..1996dbe
--- /dev/null
+++ b/media/libaudioclient/include/AudioTrack.h
@@ -0,0 +1,1154 @@
+/*
+ * 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; }
+
+ // TODO consider notificationFrames() if needed
+
+ /* 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
+
+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;
+
+ int32_t mVolumeShaperId;
+
+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
diff --git a/media/libaudioclient/include/IAudioFlinger.h b/media/libaudioclient/include/IAudioFlinger.h
new file mode 100644
index 0000000..8c5e61a
--- /dev/null
+++ b/media/libaudioclient/include/IAudioFlinger.h
@@ -0,0 +1,268 @@
+/*
+ * 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_IAUDIOFLINGER_H
+#define ANDROID_IAUDIOFLINGER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <media/IAudioTrack.h>
+#include <media/IAudioRecord.h>
+#include <media/IAudioFlingerClient.h>
+#include <system/audio.h>
+#include <system/audio_effect.h>
+#include <system/audio_policy.h>
+#include <media/IEffect.h>
+#include <media/IEffectClient.h>
+#include <utils/String8.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class IAudioFlinger : public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(AudioFlinger);
+
+
+ // invariant on exit for all APIs that return an sp<>:
+ // (return value != 0) == (*status == NO_ERROR)
+
+ /* create an audio track and registers it with AudioFlinger.
+ * return null if the track cannot be created.
+ */
+ virtual sp<IAudioTrack> createTrack(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t *pFrameCount,
+ audio_output_flags_t *flags,
+ const sp<IMemory>& sharedBuffer,
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
+ audio_io_handle_t output,
+ pid_t pid,
+ pid_t tid, // -1 means unused, otherwise must be valid non-0
+ audio_session_t *sessionId,
+ int clientUid,
+ status_t *status,
+ audio_port_handle_t portId) = 0;
+
+ virtual sp<IAudioRecord> openRecord(
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
+ audio_io_handle_t input,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ const String16& callingPackage,
+ size_t *pFrameCount,
+ audio_input_flags_t *flags,
+ pid_t pid,
+ pid_t tid, // -1 means unused, otherwise must be valid non-0
+ int clientUid,
+ audio_session_t *sessionId,
+ size_t *notificationFrames,
+ sp<IMemory>& cblk,
+ sp<IMemory>& buffers, // return value 0 means it follows cblk
+ status_t *status,
+ audio_port_handle_t portId) = 0;
+
+ // FIXME Surprisingly, format/latency don't work for input handles
+
+ /* query the audio hardware state. This state never changes,
+ * and therefore can be cached.
+ */
+ virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const = 0;
+
+ // reserved; formerly channelCount()
+
+ virtual audio_format_t format(audio_io_handle_t output) const = 0;
+ virtual size_t frameCount(audio_io_handle_t ioHandle) const = 0;
+
+ // return estimated latency in milliseconds
+ virtual uint32_t latency(audio_io_handle_t output) const = 0;
+
+ /* set/get the audio hardware state. This will probably be used by
+ * the preference panel, mostly.
+ */
+ virtual status_t setMasterVolume(float value) = 0;
+ virtual status_t setMasterMute(bool muted) = 0;
+
+ virtual float masterVolume() const = 0;
+ virtual bool masterMute() const = 0;
+
+ /* set/get stream type state. This will probably be used by
+ * the preference panel, mostly.
+ */
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output) = 0;
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted) = 0;
+
+ virtual float streamVolume(audio_stream_type_t stream,
+ audio_io_handle_t output) const = 0;
+ virtual bool streamMute(audio_stream_type_t stream) const = 0;
+
+ // set audio mode
+ virtual status_t setMode(audio_mode_t mode) = 0;
+
+ // mic mute/state
+ virtual status_t setMicMute(bool state) = 0;
+ virtual bool getMicMute() const = 0;
+
+ virtual status_t setParameters(audio_io_handle_t ioHandle,
+ const String8& keyValuePairs) = 0;
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys)
+ const = 0;
+
+ // Register an object to receive audio input/output change and track notifications.
+ // For a given calling pid, AudioFlinger disregards any registrations after the first.
+ // Thus the IAudioFlingerClient must be a singleton per process.
+ virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
+
+ // retrieve the audio recording buffer size
+ // FIXME This API assumes a route, and so should be deprecated.
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask) const = 0;
+
+ virtual status_t openOutput(audio_module_handle_t module,
+ audio_io_handle_t *output,
+ audio_config_t *config,
+ audio_devices_t *devices,
+ const String8& address,
+ uint32_t *latencyMs,
+ audio_output_flags_t flags) = 0;
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2) = 0;
+ virtual status_t closeOutput(audio_io_handle_t output) = 0;
+ virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+ virtual status_t restoreOutput(audio_io_handle_t output) = 0;
+
+ virtual status_t openInput(audio_module_handle_t module,
+ audio_io_handle_t *input,
+ audio_config_t *config,
+ audio_devices_t *device,
+ const String8& address,
+ audio_source_t source,
+ audio_input_flags_t flags) = 0;
+ virtual status_t closeInput(audio_io_handle_t input) = 0;
+
+ virtual status_t invalidateStream(audio_stream_type_t stream) = 0;
+
+ virtual status_t setVoiceVolume(float volume) = 0;
+
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const = 0;
+
+ virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
+
+ virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use) = 0;
+
+ virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid) = 0;
+ virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid) = 0;
+
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;
+
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const = 0;
+
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pEffectUUID,
+ effect_descriptor_t *pDescriptor) const = 0;
+
+ virtual sp<IEffect> createEffect(
+ effect_descriptor_t *pDesc,
+ const sp<IEffectClient>& client,
+ int32_t priority,
+ // AudioFlinger doesn't take over handle reference from client
+ audio_io_handle_t output,
+ audio_session_t sessionId,
+ const String16& callingPackage,
+ pid_t pid,
+ status_t *status,
+ int *id,
+ int *enabled) = 0;
+
+ virtual status_t moveEffects(audio_session_t session, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput) = 0;
+
+ virtual audio_module_handle_t loadHwModule(const char *name) = 0;
+
+ // helpers for android.media.AudioManager.getProperty(), see description there for meaning
+ // FIXME move these APIs to AudioPolicy to permit a more accurate implementation
+ // that looks on primary device for a stream with fast flag, primary flag, or first one.
+ virtual uint32_t getPrimaryOutputSamplingRate() = 0;
+ virtual size_t getPrimaryOutputFrameCount() = 0;
+
+ // Intended for AudioService to inform AudioFlinger of device's low RAM attribute,
+ // and should be called at most once. For a definition of what "low RAM" means, see
+ // android.app.ActivityManager.isLowRamDevice().
+ virtual status_t setLowRamDevice(bool isLowRamDevice) = 0;
+
+ /* List available audio ports and their attributes */
+ virtual status_t listAudioPorts(unsigned int *num_ports,
+ struct audio_port *ports) = 0;
+
+ /* Get attributes for a given audio port */
+ virtual status_t getAudioPort(struct audio_port *port) = 0;
+
+ /* Create an audio patch between several source and sink ports */
+ virtual status_t createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle) = 0;
+
+ /* Release an audio patch */
+ virtual status_t releaseAudioPatch(audio_patch_handle_t handle) = 0;
+
+ /* List existing audio patches */
+ virtual status_t listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches) = 0;
+ /* Set audio port configuration */
+ virtual status_t setAudioPortConfig(const struct audio_port_config *config) = 0;
+
+ /* Get the HW synchronization source used for an audio session */
+ virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) = 0;
+
+ /* Indicate JAVA services are ready (scheduling, power management ...) */
+ virtual status_t systemReady() = 0;
+
+ // Returns the number of frames per audio HAL buffer.
+ virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const = 0;
+};
+
+
+// ----------------------------------------------------------------------------
+
+class BnAudioFlinger : public BnInterface<IAudioFlinger>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_IAUDIOFLINGER_H
diff --git a/media/libaudioclient/include/IAudioFlingerClient.h b/media/libaudioclient/include/IAudioFlingerClient.h
new file mode 100644
index 0000000..0080bc9
--- /dev/null
+++ b/media/libaudioclient/include/IAudioFlingerClient.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2009 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_IAUDIOFLINGERCLIENT_H
+#define ANDROID_IAUDIOFLINGERCLIENT_H
+
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <utils/KeyedVector.h>
+#include <system/audio.h>
+#include <media/AudioIoDescriptor.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class IAudioFlingerClient : public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(AudioFlingerClient);
+
+ // Notifies a change of audio input/output configuration.
+ virtual void ioConfigChanged(audio_io_config_event event,
+ const sp<AudioIoDescriptor>& ioDesc) = 0;
+
+};
+
+
+// ----------------------------------------------------------------------------
+
+class BnAudioFlingerClient : public BnInterface<IAudioFlingerClient>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_IAUDIOFLINGERCLIENT_H
diff --git a/media/libaudioclient/include/IAudioPolicyService.h b/media/libaudioclient/include/IAudioPolicyService.h
new file mode 100644
index 0000000..d111fd2
--- /dev/null
+++ b/media/libaudioclient/include/IAudioPolicyService.h
@@ -0,0 +1,190 @@
+/*
+ * Copyright (C) 2009 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_IAUDIOPOLICYSERVICE_H
+#define ANDROID_IAUDIOPOLICYSERVICE_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <media/AudioSystem.h>
+#include <media/AudioPolicy.h>
+#include <media/IAudioPolicyServiceClient.h>
+
+#include <system/audio_policy.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class IAudioPolicyService : public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(AudioPolicyService);
+
+ //
+ // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
+ //
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name) = 0;
+ virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+ const char *device_address) = 0;
+ virtual status_t handleDeviceConfigChange(audio_devices_t device,
+ const char *device_address,
+ const char *device_name) = 0;
+ virtual status_t setPhoneState(audio_mode_t state) = 0;
+ virtual status_t setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config) = 0;
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
+ virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ audio_channel_mask_t channelMask = 0,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
+ virtual status_t getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uid_t uid,
+ const audio_config_t *config,
+ audio_output_flags_t flags,
+ audio_port_handle_t selectedDeviceId,
+ audio_port_handle_t *portId) = 0;
+ virtual status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session) = 0;
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session) = 0;
+ virtual void releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session) = 0;
+ virtual status_t getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ pid_t pid,
+ uid_t uid,
+ const audio_config_base_t *config,
+ audio_input_flags_t flags,
+ audio_port_handle_t selectedDeviceId,
+ audio_port_handle_t *portId) = 0;
+ virtual status_t startInput(audio_io_handle_t input,
+ audio_session_t session) = 0;
+ virtual status_t stopInput(audio_io_handle_t input,
+ audio_session_t session) = 0;
+ virtual void releaseInput(audio_io_handle_t input,
+ audio_session_t session) = 0;
+ virtual status_t initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax) = 0;
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device) = 0;
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device) = 0;
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream) = 0;
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream) = 0;
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc) = 0;
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ audio_session_t session,
+ int id) = 0;
+ virtual status_t unregisterEffect(int id) = 0;
+ virtual status_t setEffectEnabled(int id, bool enabled) = 0;
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const = 0;
+ virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0)
+ const = 0;
+ virtual bool isSourceActive(audio_source_t source) const = 0;
+ virtual status_t queryDefaultPreProcessing(audio_session_t audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count) = 0;
+ // Check if offload is possible for given format, stream type, sample rate,
+ // bit rate, duration, video and streaming or offload property is enabled
+ virtual bool isOffloadSupported(const audio_offload_info_t& info) = 0;
+
+ /* List available audio ports and their attributes */
+ virtual status_t listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation) = 0;
+
+ /* Get attributes for a given audio port */
+ virtual status_t getAudioPort(struct audio_port *port) = 0;
+
+ /* Create an audio patch between several source and sink ports */
+ virtual status_t createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle) = 0;
+
+ /* Release an audio patch */
+ virtual status_t releaseAudioPatch(audio_patch_handle_t handle) = 0;
+
+ /* List existing audio patches */
+ virtual status_t listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation) = 0;
+ /* Set audio port configuration */
+ virtual status_t setAudioPortConfig(const struct audio_port_config *config) = 0;
+
+ virtual void registerClient(const sp<IAudioPolicyServiceClient>& client) = 0;
+
+ virtual void setAudioPortCallbacksEnabled(bool enabled) = 0;
+
+ virtual status_t acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device) = 0;
+
+ virtual status_t releaseSoundTriggerSession(audio_session_t session) = 0;
+
+ virtual audio_mode_t getPhoneState() = 0;
+
+ virtual status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration) = 0;
+
+ virtual status_t startAudioSource(const struct audio_port_config *source,
+ const audio_attributes_t *attributes,
+ audio_patch_handle_t *handle) = 0;
+ virtual status_t stopAudioSource(audio_patch_handle_t handle) = 0;
+
+ virtual status_t setMasterMono(bool mono) = 0;
+ virtual status_t getMasterMono(bool *mono) = 0;
+};
+
+
+// ----------------------------------------------------------------------------
+
+class BnAudioPolicyService : public BnInterface<IAudioPolicyService>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_IAUDIOPOLICYSERVICE_H
diff --git a/media/libaudioclient/include/IAudioPolicyServiceClient.h b/media/libaudioclient/include/IAudioPolicyServiceClient.h
new file mode 100644
index 0000000..d94ad00
--- /dev/null
+++ b/media/libaudioclient/include/IAudioPolicyServiceClient.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (C) 2009 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_IAUDIOPOLICYSERVICECLIENT_H
+#define ANDROID_IAUDIOPOLICYSERVICECLIENT_H
+
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <system/audio.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class IAudioPolicyServiceClient : public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(AudioPolicyServiceClient);
+
+ // Notifies a change of audio port configuration.
+ virtual void onAudioPortListUpdate() = 0;
+ // Notifies a change of audio patch configuration.
+ virtual void onAudioPatchListUpdate() = 0;
+ // Notifies a change in the mixing state of a specific mix in a dynamic audio policy
+ virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state) = 0;
+ // Notifies a change of audio recording configuration
+ virtual void onRecordingConfigurationUpdate(int event, audio_session_t session,
+ audio_source_t source,
+ const audio_config_base_t *clientConfig,
+ const audio_config_base_t *deviceConfig,
+ audio_patch_handle_t patchHandle) = 0;
+};
+
+
+// ----------------------------------------------------------------------------
+
+class BnAudioPolicyServiceClient : public BnInterface<IAudioPolicyServiceClient>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_IAUDIOPOLICYSERVICECLIENT_H
diff --git a/media/libaudioclient/include/IAudioRecord.h b/media/libaudioclient/include/IAudioRecord.h
new file mode 100644
index 0000000..7768176
--- /dev/null
+++ b/media/libaudioclient/include/IAudioRecord.h
@@ -0,0 +1,66 @@
+/*
+ * 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 IAUDIORECORD_H_
+#define IAUDIORECORD_H_
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <binder/IMemory.h>
+#include <system/audio.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class IAudioRecord : public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(AudioRecord);
+
+ /* After it's created the track is not active. Call start() to
+ * make it active.
+ */
+ virtual status_t start(int /*AudioSystem::sync_event_t*/ event,
+ audio_session_t triggerSession) = 0;
+
+ /* Stop a track. If set, the callback will cease being called and
+ * obtainBuffer will return an error. Buffers that are already released
+ * will be processed, unless flush() is called.
+ */
+ virtual void stop() = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnAudioRecord : public BnInterface<IAudioRecord>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif /*IAUDIORECORD_H_*/
diff --git a/media/libaudioclient/include/IAudioTrack.h b/media/libaudioclient/include/IAudioTrack.h
new file mode 100644
index 0000000..27a62d6
--- /dev/null
+++ b/media/libaudioclient/include/IAudioTrack.h
@@ -0,0 +1,103 @@
+/*
+ * 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_IAUDIOTRACK_H
+#define ANDROID_IAUDIOTRACK_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <binder/IMemory.h>
+#include <utils/String8.h>
+#include <media/AudioTimestamp.h>
+#include <media/VolumeShaper.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class IAudioTrack : public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(AudioTrack);
+
+ /* Get this track's control block */
+ virtual sp<IMemory> getCblk() const = 0;
+
+ /* After it's created the track is not active. Call start() to
+ * make it active.
+ */
+ virtual status_t start() = 0;
+
+ /* Stop a track. If set, the callback will cease being called and
+ * obtainBuffer will return an error. Buffers that are already released
+ * will continue to be processed, unless/until flush() is called.
+ */
+ virtual void stop() = 0;
+
+ /* Flush a stopped or paused track. All pending/released buffers are discarded.
+ * This function has no effect if the track is not stopped or paused.
+ */
+ virtual void flush() = 0;
+
+ /* Pause a track. If set, the callback will cease being called and
+ * obtainBuffer will return an error. Buffers that are already released
+ * will continue to be processed, unless/until flush() is called.
+ */
+ virtual void pause() = 0;
+
+ /* Attach track auxiliary output to specified effect. Use effectId = 0
+ * to detach track from effect.
+ */
+ virtual status_t attachAuxEffect(int effectId) = 0;
+
+ /* Send parameters to the audio hardware */
+ virtual status_t setParameters(const String8& keyValuePairs) = 0;
+
+ /* Return NO_ERROR if timestamp is valid. timestamp is undefined otherwise. */
+ virtual status_t getTimestamp(AudioTimestamp& timestamp) = 0;
+
+ /* Signal the playback thread for a change in control block */
+ virtual void signal() = 0;
+
+ /* Sets the volume shaper */
+ virtual VolumeShaper::Status applyVolumeShaper(
+ const sp<VolumeShaper::Configuration>& configuration,
+ const sp<VolumeShaper::Operation>& operation) = 0;
+
+ /* gets the volume shaper state */
+ virtual sp<VolumeShaper::State> getVolumeShaperState(int id) = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnAudioTrack : public BnInterface<IAudioTrack>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_IAUDIOTRACK_H
diff --git a/media/libaudioclient/include/IEffect.h b/media/libaudioclient/include/IEffect.h
new file mode 100644
index 0000000..ff04869
--- /dev/null
+++ b/media/libaudioclient/include/IEffect.h
@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2010 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_IEFFECT_H
+#define ANDROID_IEFFECT_H
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+#include <binder/IMemory.h>
+
+namespace android {
+
+class IEffect: public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(Effect);
+
+ virtual status_t enable() = 0;
+
+ virtual status_t disable() = 0;
+
+ virtual status_t command(uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t *pReplySize,
+ void *pReplyData) = 0;
+
+ virtual void disconnect() = 0;
+
+ virtual sp<IMemory> getCblk() const = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnEffect: public BnInterface<IEffect>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_IEFFECT_H
diff --git a/media/libaudioclient/include/IEffectClient.h b/media/libaudioclient/include/IEffectClient.h
new file mode 100644
index 0000000..2f78c98
--- /dev/null
+++ b/media/libaudioclient/include/IEffectClient.h
@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2010 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_IEFFECTCLIENT_H
+#define ANDROID_IEFFECTCLIENT_H
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+#include <binder/IMemory.h>
+
+namespace android {
+
+class IEffectClient: public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(EffectClient);
+
+ virtual void controlStatusChanged(bool controlGranted) = 0;
+ virtual void enableStatusChanged(bool enabled) = 0;
+ virtual void commandExecuted(uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t replySize,
+ void *pReplyData) = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnEffectClient: public BnInterface<IEffectClient>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_IEFFECTCLIENT_H
diff --git a/media/libaudioclient/include/ToneGenerator.h b/media/libaudioclient/include/ToneGenerator.h
new file mode 100644
index 0000000..9fd5f61
--- /dev/null
+++ b/media/libaudioclient/include/ToneGenerator.h
@@ -0,0 +1,327 @@
+/*
+ * Copyright (C) 2008 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_TONEGENERATOR_H_
+#define ANDROID_TONEGENERATOR_H_
+
+#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
+#include <utils/Compat.h>
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class ToneGenerator {
+public:
+
+ // List of all available tones
+ // This enum must be kept consistant with constants in ToneGenerator JAVA class
+ enum tone_type {
+ // DTMF tones ITU-T Recommendation Q.23
+ TONE_DTMF_0 = 0, // 0 key: 1336Hz, 941Hz
+ TONE_DTMF_1, // 1 key: 1209Hz, 697Hz
+ TONE_DTMF_2, // 2 key: 1336Hz, 697Hz
+ TONE_DTMF_3, // 3 key: 1477Hz, 697Hz
+ TONE_DTMF_4, // 4 key: 1209Hz, 770Hz
+ TONE_DTMF_5, // 5 key: 1336Hz, 770Hz
+ TONE_DTMF_6, // 6 key: 1477Hz, 770Hz
+ TONE_DTMF_7, // 7 key: 1209Hz, 852Hz
+ TONE_DTMF_8, // 8 key: 1336Hz, 852Hz
+ TONE_DTMF_9, // 9 key: 1477Hz, 852Hz
+ TONE_DTMF_S, // * key: 1209Hz, 941Hz
+ TONE_DTMF_P, // # key: 1477Hz, 941Hz
+ TONE_DTMF_A, // A key: 1633Hz, 697Hz
+ TONE_DTMF_B, // B key: 1633Hz, 770Hz
+ TONE_DTMF_C, // C key: 1633Hz, 852Hz
+ TONE_DTMF_D, // D key: 1633Hz, 941Hz
+ // Call supervisory tones: 3GPP TS 22.001 (CEPT)
+ TONE_SUP_DIAL, // Dial tone: CEPT: 425Hz, continuous
+ FIRST_SUP_TONE = TONE_SUP_DIAL,
+ TONE_SUP_BUSY, // Busy tone, CEPT: 425Hz, 500ms ON, 500ms OFF...
+ TONE_SUP_CONGESTION, // Congestion tone CEPT, JAPAN: 425Hz, 200ms ON, 200ms OFF...
+ TONE_SUP_RADIO_ACK, // Radio path acknowlegment, CEPT, ANSI: 425Hz, 200ms ON
+ TONE_SUP_RADIO_NOTAVAIL, // Radio path not available: 425Hz, 200ms ON, 200 OFF 3 bursts
+ TONE_SUP_ERROR, // Error/Special info: 950Hz+1400Hz+1800Hz, 330ms ON, 1s OFF...
+ TONE_SUP_CALL_WAITING, // Call Waiting CEPT,JAPAN: 425Hz, 200ms ON, 600ms OFF, 200ms ON, 3s OFF...
+ TONE_SUP_RINGTONE, // Ring Tone CEPT, JAPAN: 425Hz, 1s ON, 4s OFF...
+ LAST_SUP_TONE = TONE_SUP_RINGTONE,
+ // Proprietary tones: 3GPP TS 31.111
+ TONE_PROP_BEEP, // General beep: 400Hz+1200Hz, 35ms ON
+ TONE_PROP_ACK, // Positive Acknowlgement: 1200Hz, 100ms ON, 100ms OFF 2 bursts
+ TONE_PROP_NACK, // Negative Acknowlgement: 300Hz+400Hz+500Hz, 400ms ON
+ TONE_PROP_PROMPT, // Prompt tone: 400Hz+1200Hz, 200ms ON
+ TONE_PROP_BEEP2, // General double beep: 400Hz+1200Hz, 35ms ON, 200ms OFF, 35ms on
+ // Additional call supervisory tones: specified by IS-95 only
+ TONE_SUP_INTERCEPT, // Intercept tone: alternating 440 Hz and 620 Hz tones, each on for 250 ms.
+ TONE_SUP_INTERCEPT_ABBREV, // Abbreviated intercept: intercept tone limited to 4 seconds
+ TONE_SUP_CONGESTION_ABBREV, // Abbreviated congestion: congestion tone limited to 4 seconds
+ TONE_SUP_CONFIRM, // Confirm tone: a 350 Hz tone added to a 440 Hz tone repeated 3 times in a 100 ms on, 100 ms off cycle.
+ TONE_SUP_PIP, // Pip tone: four bursts of 480 Hz tone (0.1 s on, 0.1 s off).
+
+ // CDMA Tones
+ TONE_CDMA_DIAL_TONE_LITE,
+ TONE_CDMA_NETWORK_USA_RINGBACK,
+ TONE_CDMA_INTERCEPT,
+ TONE_CDMA_ABBR_INTERCEPT,
+ TONE_CDMA_REORDER,
+ TONE_CDMA_ABBR_REORDER,
+ TONE_CDMA_NETWORK_BUSY,
+ TONE_CDMA_CONFIRM,
+ TONE_CDMA_ANSWER,
+ TONE_CDMA_NETWORK_CALLWAITING,
+ TONE_CDMA_PIP,
+
+ // ISDN
+ TONE_CDMA_CALL_SIGNAL_ISDN_NORMAL, // ISDN Alert Normal
+ TONE_CDMA_CALL_SIGNAL_ISDN_INTERGROUP, // ISDN Intergroup
+ TONE_CDMA_CALL_SIGNAL_ISDN_SP_PRI, // ISDN SP PRI
+ TONE_CDMA_CALL_SIGNAL_ISDN_PAT3, // ISDN Alert PAT3
+ TONE_CDMA_CALL_SIGNAL_ISDN_PING_RING, // ISDN Alert PING RING
+ TONE_CDMA_CALL_SIGNAL_ISDN_PAT5, // ISDN Alert PAT5
+ TONE_CDMA_CALL_SIGNAL_ISDN_PAT6, // ISDN Alert PAT6
+ TONE_CDMA_CALL_SIGNAL_ISDN_PAT7, // ISDN Alert PAT7
+ // ISDN end
+
+ // IS54
+ TONE_CDMA_HIGH_L, // IS54 High Pitch Long
+ TONE_CDMA_MED_L, // IS54 Med Pitch Long
+ TONE_CDMA_LOW_L, // IS54 Low Pitch Long
+ TONE_CDMA_HIGH_SS, // IS54 High Pitch Short Short
+ TONE_CDMA_MED_SS, // IS54 Medium Pitch Short Short
+ TONE_CDMA_LOW_SS, // IS54 Low Pitch Short Short
+ TONE_CDMA_HIGH_SSL, // IS54 High Pitch Short Short Long
+ TONE_CDMA_MED_SSL, // IS54 Medium Pitch Short Short Long
+ TONE_CDMA_LOW_SSL, // IS54 Low Pitch Short Short Long
+ TONE_CDMA_HIGH_SS_2, // IS54 High Pitch Short Short 2
+ TONE_CDMA_MED_SS_2, // IS54 Med Pitch Short Short 2
+ TONE_CDMA_LOW_SS_2, // IS54 Low Pitch Short Short 2
+ TONE_CDMA_HIGH_SLS, // IS54 High Pitch Short Long Short
+ TONE_CDMA_MED_SLS, // IS54 Med Pitch Short Long Short
+ TONE_CDMA_LOW_SLS, // IS54 Low Pitch Short Long Short
+ TONE_CDMA_HIGH_S_X4, // IS54 High Pitch Short Short Short Short
+ TONE_CDMA_MED_S_X4, // IS54 Med Pitch Short Short Short Short
+ TONE_CDMA_LOW_S_X4, // IS54 Low Pitch Short Short Short Short
+ TONE_CDMA_HIGH_PBX_L, // PBX High Pitch Long
+ TONE_CDMA_MED_PBX_L, // PBX Med Pitch Long
+ TONE_CDMA_LOW_PBX_L, // PBX Low Pitch Long
+ TONE_CDMA_HIGH_PBX_SS, // PBX High Short Short
+ TONE_CDMA_MED_PBX_SS, // PBX Med Short Short
+ TONE_CDMA_LOW_PBX_SS, // PBX Low Short Short
+ TONE_CDMA_HIGH_PBX_SSL, // PBX High Short Short Long
+ TONE_CDMA_MED_PBX_SSL, // PBX Med Short Short Long
+ TONE_CDMA_LOW_PBX_SSL, // PBX Low Short Short Long
+ TONE_CDMA_HIGH_PBX_SLS, // PBX High SLS
+ TONE_CDMA_MED_PBX_SLS, // PBX Med SLS
+ TONE_CDMA_LOW_PBX_SLS, // PBX Low SLS
+ TONE_CDMA_HIGH_PBX_S_X4, // PBX High SSSS
+ TONE_CDMA_MED_PBX_S_X4, // PBX Med SSSS
+ TONE_CDMA_LOW_PBX_S_X4, // PBX LOW SSSS
+ //IS54 end
+ // proprietary
+ TONE_CDMA_ALERT_NETWORK_LITE,
+ TONE_CDMA_ALERT_AUTOREDIAL_LITE,
+ TONE_CDMA_ONE_MIN_BEEP,
+ TONE_CDMA_KEYPAD_VOLUME_KEY_LITE,
+ TONE_CDMA_PRESSHOLDKEY_LITE,
+ TONE_CDMA_ALERT_INCALL_LITE,
+ TONE_CDMA_EMERGENCY_RINGBACK,
+ TONE_CDMA_ALERT_CALL_GUARD,
+ TONE_CDMA_SOFT_ERROR_LITE,
+ TONE_CDMA_CALLDROP_LITE,
+ // proprietary end
+ TONE_CDMA_NETWORK_BUSY_ONE_SHOT,
+ TONE_CDMA_ABBR_ALERT,
+ TONE_CDMA_SIGNAL_OFF,
+ //CDMA end
+ NUM_TONES,
+ NUM_SUP_TONES = LAST_SUP_TONE-FIRST_SUP_TONE+1
+ };
+
+ ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava = false);
+ ~ToneGenerator();
+
+ bool startTone(tone_type toneType, int durationMs = -1);
+ void stopTone();
+
+ bool isInited() { return (mState == TONE_IDLE)?false:true;}
+
+ // returns the audio session this ToneGenerator belongs to or 0 if an error occured.
+ int getSessionId() { return (mpAudioTrack == 0) ? 0 : mpAudioTrack->getSessionId(); }
+
+private:
+
+ enum tone_state {
+ TONE_IDLE, // ToneGenerator is being initialized or initialization failed
+ TONE_INIT, // ToneGenerator has been successfully initialized and is not playing
+ TONE_STARTING, // ToneGenerator is starting playing
+ TONE_PLAYING, // ToneGenerator is playing
+ TONE_STOPPING, // ToneGenerator is stoping
+ TONE_STOPPED, // ToneGenerator is stopped: the AudioTrack will be stopped
+ TONE_RESTARTING // A start request was received in active state (playing or stopping)
+ };
+
+
+ // Region specific tones.
+ // These supervisory tones are different depending on the region (USA/CANADA, JAPAN, rest of the world).
+ // When a tone in the range [FIRST_SUP_TONE, LAST_SUP_TONE] is requested, the region is determined
+ // from system property gsm.operator.iso-country and the proper tone descriptor is selected with the
+ // help of sToneMappingTable[]
+ enum regional_tone_type {
+ // ANSI supervisory tones
+ TONE_ANSI_DIAL = NUM_TONES, // Dial tone: a continuous 350 Hz + 440 Hz tone.
+ TONE_ANSI_BUSY, // Busy tone on: a 480 Hz + 620 Hz tone repeated in a 500 ms on, 500 ms off cycle.
+ TONE_ANSI_CONGESTION, // Network congestion (reorder) tone on: a 480 Hz + 620 Hz tone repeated in a 250 ms on, 250 ms off cycle.
+ TONE_ANSI_CALL_WAITING, // Call waiting tone on: 440 Hz, on for 300 ms, 9,7 s off followed by
+ // (440 Hz, on for 100 ms off for 100 ms, on for 100 ms, 9,7s off and repeated as necessary).
+ TONE_ANSI_RINGTONE, // Ring Tone: a 440 Hz + 480 Hz tone repeated in a 2 s on, 4 s off pattern.
+ // JAPAN Supervisory tones
+ TONE_JAPAN_DIAL, // Dial tone: 400Hz, continuous
+ TONE_JAPAN_BUSY, // Busy tone: 400Hz, 500ms ON, 500ms OFF...
+ TONE_JAPAN_RADIO_ACK, // Radio path acknowlegment: 400Hz, 1s ON, 2s OFF...
+ // GB Supervisory tones
+ TONE_GB_RINGTONE, // Ring Tone: A 400Hz + 450Hz tone repeated in a 0.4s on, 0.2s off, 0.4s on, 2.0s off pattern.
+ // AUSTRALIA Supervisory tones
+ TONE_AUSTRALIA_RINGTONE, // Ring tone: A 400Hz + 450Hz tone repeated in a 0.4s on, 0.2s off, 0.4s on, 2.0s off pattern.
+ TONE_AUSTRALIA_BUSY, // Busy tone: 425 Hz repeated in a 0.375s on, 0.375s off pattern.
+ TONE_AUSTRALIA_CALL_WAITING,// Call waiting tone: 425Hz tone repeated in a 0.2s on, 0.2s off, 0.2s on, 4.4s off pattern.
+ TONE_AUSTRALIA_CONGESTION, // Congestion tone: 425Hz tone repeated in a 0.375s on, 0.375s off pattern
+ NUM_ALTERNATE_TONES
+ };
+
+ enum region {
+ ANSI,
+ JAPAN,
+ GB,
+ AUSTRALIA,
+ CEPT,
+ NUM_REGIONS
+ };
+
+ static const unsigned char sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES];
+
+ static const unsigned int TONEGEN_MAX_WAVES = 3; // Maximun number of sine waves in a tone segment
+ static const unsigned int TONEGEN_MAX_SEGMENTS = 12; // Maximun number of segments in a tone descriptor
+ static const unsigned int TONEGEN_INF = 0xFFFFFFFF; // Represents infinite time duration
+ static const CONSTEXPR float TONEGEN_GAIN = 0.9; // Default gain passed to WaveGenerator().
+
+ // ToneDescriptor class contains all parameters needed to generate a tone:
+ // - The array waveFreq[]:
+ // 1 for static tone descriptors: contains the frequencies of all individual waves making the multi-tone.
+ // 2 for active tone descritors: contains the indexes of the WaveGenerator objects in mWaveGens
+ // The number of sine waves varies from 1 to TONEGEN_MAX_WAVES.
+ // The first null value indicates that no more waves are needed.
+ // - The array segments[] is used to generate the tone pulses. A segment is a period of time
+ // during which the tone is ON or OFF. Segments with even index (starting from 0)
+ // correspond to tone ON state and segments with odd index to OFF state.
+ // The data stored in segments[] is the duration of the corresponding period in ms.
+ // The first segment encountered with a 0 duration indicates that no more segment follows.
+ // - loopCnt - Number of times to repeat a sequence of seqments after playing this
+ // - loopIndx - The segment index to go back and play is loopcnt > 0
+ // - repeatCnt indicates the number of times the sequence described by segments[] array must be repeated.
+ // When the tone generator encounters the first 0 duration segment, it will compare repeatCnt to mCurCount.
+ // If mCurCount > repeatCnt, the tone is stopped automatically. Otherwise, tone sequence will be
+ // restarted from segment repeatSegment.
+ // - repeatSegment number of the first repeated segment when repeatCnt is not null
+
+ class ToneSegment {
+ public:
+ unsigned int duration;
+ unsigned short waveFreq[TONEGEN_MAX_WAVES+1];
+ unsigned short loopCnt;
+ unsigned short loopIndx;
+ };
+
+ class ToneDescriptor {
+ public:
+ ToneSegment segments[TONEGEN_MAX_SEGMENTS+1];
+ unsigned long repeatCnt;
+ unsigned long repeatSegment;
+ };
+
+ static const ToneDescriptor sToneDescriptors[];
+
+ bool mThreadCanCallJava;
+ unsigned int mTotalSmp; // Total number of audio samples played (gives current time)
+ unsigned int mNextSegSmp; // Position of next segment transition expressed in samples
+ // NOTE: because mTotalSmp, mNextSegSmp are stored on 32 bit, current design will operate properly
+ // only if tone duration is less than about 27 Hours(@44100Hz sampling rate). If this time is exceeded,
+ // no crash will occur but tone sequence will show a glitch.
+ unsigned int mMaxSmp; // Maximum number of audio samples played (maximun tone duration)
+ int mDurationMs; // Maximum tone duration in ms
+
+ unsigned short mCurSegment; // Current segment index in ToneDescriptor segments[]
+ unsigned short mCurCount; // Current sequence repeat count
+ volatile unsigned short mState; // ToneGenerator state (tone_state)
+ unsigned short mRegion;
+ const ToneDescriptor *mpToneDesc; // pointer to active tone descriptor
+ const ToneDescriptor *mpNewToneDesc; // pointer to next active tone descriptor
+
+ unsigned short mLoopCounter; // Current tone loopback count
+
+ uint32_t mSamplingRate; // AudioFlinger Sampling rate
+ sp<AudioTrack> mpAudioTrack; // Pointer to audio track used for playback
+ Mutex mLock; // Mutex to control concurent access to ToneGenerator object from audio callback and application API
+ Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond
+ Condition mWaitCbkCond; // condition enabling interface to wait for audio callback completion after a change is requested
+ float mVolume; // Volume applied to audio track
+ audio_stream_type_t mStreamType; // Audio stream used for output
+ unsigned int mProcessSize; // Size of audio blocks generated at a time by audioCallback() (in PCM frames).
+ struct timespec mStartTime; // tone start time: needed to guaranty actual tone duration
+
+ bool initAudioTrack();
+ static void audioCallback(int event, void* user, void *info);
+ bool prepareWave();
+ unsigned int numWaves(unsigned int segmentIdx);
+ void clearWaveGens();
+ tone_type getToneForRegion(tone_type toneType);
+
+ // WaveGenerator generates a single sine wave
+ class WaveGenerator {
+ public:
+ enum gen_command {
+ WAVEGEN_START, // Start/restart wave from phase 0
+ WAVEGEN_CONT, // Continue wave from current phase
+ WAVEGEN_STOP // Stop wave on zero crossing
+ };
+
+ WaveGenerator(unsigned short samplingRate, unsigned short frequency,
+ float volume);
+ ~WaveGenerator();
+
+ void getSamples(short *outBuffer, unsigned int count,
+ unsigned int command);
+
+ private:
+ static const short GEN_AMP = 32000; // amplitude of generator
+ static const short S_Q14 = 14; // shift for Q14
+ static const short S_Q15 = 15; // shift for Q15
+
+ short mA1_Q14; // Q14 coefficient
+ // delay line of full amplitude generator
+ long mS1, mS2; // delay line S2 oldest
+ short mS2_0; // saved value for reinitialisation
+ short mAmplitude_Q15; // Q15 amplitude
+ };
+
+ KeyedVector<unsigned short, WaveGenerator *> mWaveGens; // list of active wave generators.
+};
+
+}
+; // namespace android
+
+#endif /*ANDROID_TONEGENERATOR_H_*/