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Commit 5d447855 authored by Dima Zavin's avatar Dima Zavin Committed by Android (Google) Code Review
Browse files

Merge changes I22d9e018,Ib0701fcc,Ibc637918,I9eb7e002,I4adcec73 

* changes:
  audioflinger: don't do work in constructor, instead do it in onFirstRef
  audioflinger: enumerate all the possible audio interfaces
  audio/media: convert to using the audio HAL and new audio defs
  libmedia: move AudioParameter out of AudioSystem
  audioflinger: move legacy audio hw/policy out to libhardware_legacy
parents 18a3dc66 5a61d2f2

cmds/stagefright/audioloop.cpp

+4 −2
Original line number Diff line number Diff line
@@ -11,6 +11,8 @@ 
#include <media/stagefright/OMXClient.h>

#include <media/stagefright/OMXCodec.h>



#include <hardware/audio.h>



using namespace android;



int main() {
@@ -31,8 +33,8 @@ int main() { 
            AUDIO_SOURCE_DEFAULT,

            kSampleRate,

            kNumChannels == 1

                ? AudioSystem::CHANNEL_IN_MONO

                : AudioSystem::CHANNEL_IN_STEREO);

                ? AUDIO_CHANNEL_IN_MONO

                : AUDIO_CHANNEL_IN_STEREO);

#endif



    sp<MetaData> meta = new MetaData;

include/media/AudioParameter.h

0 → 100644
+72 −0
Original line number Diff line number Diff line 
/*

 * 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)

    static const char *keyRouting;

    static const char *keySamplingRate;

    static const char *keyFormat;

    static const char *keyChannels;

    static const char *keyFrameCount;

    static const char *keyInputSource;



    String8 toString();



    status_t add(const String8& key, const String8& value);

    status_t addInt(const String8& key, const int value);

    status_t addFloat(const String8& key, const float value);



    status_t remove(const String8& key);



    status_t get(const String8& key, String8& value);

    status_t getInt(const String8& key, int& value);

    status_t getFloat(const String8& key, float& value);

    status_t getAt(size_t index, String8& key, String8& value);



    size_t size() { return mParameters.size(); }



private:

    String8 mKeyValuePairs;

    KeyedVector <String8, String8> mParameters;

};



};  // namespace android



#endif  /*ANDROID_AUDIOPARAMETER_H_*/

include/media/AudioRecord.h

+8 −7
Original line number Diff line number Diff line
@@ -30,6 +30,7 @@ 
#include <binder/IMemory.h>

#include <utils/threads.h>



#include <hardware/audio.h>



namespace android {
@@ -127,9 +128,9 @@ public: 
     *

     * inputSource:        Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT).

     * sampleRate:         Track sampling rate in Hz.

     * format:             Audio format (e.g AudioSystem::PCM_16_BIT for signed

     * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed

     *                     16 bits per sample).

     * channels:           Channel mask: see AudioSystem::audio_channels.

     * channels:           Channel mask: see audio_channels_t.

     * frameCount:         Total size of track PCM buffer in frames. This defines the

     *                     latency of the track.

     * flags:              A bitmask of acoustic values from enum record_flags.  It enables
@@ -142,15 +143,15 @@ public: 
     */



     enum record_flags {

         RECORD_AGC_ENABLE = AudioSystem::AGC_ENABLE,

         RECORD_NS_ENABLE  = AudioSystem::NS_ENABLE,

         RECORD_IIR_ENABLE = AudioSystem::TX_IIR_ENABLE

         RECORD_AGC_ENABLE = AUDIO_IN_ACOUSTICS_AGC_ENABLE,

         RECORD_NS_ENABLE  = AUDIO_IN_ACOUSTICS_NS_ENABLE,

         RECORD_IIR_ENABLE = AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE,

     };



                        AudioRecord(int inputSource,

                                    uint32_t sampleRate = 0,

                                    int format          = 0,

                                    uint32_t channels = AudioSystem::CHANNEL_IN_MONO,

                                    uint32_t channels = AUDIO_CHANNEL_IN_MONO,

                                    int frameCount      = 0,

                                    uint32_t flags      = 0,

                                    callback_t cbf = 0,
@@ -176,7 +177,7 @@ public: 
            status_t    set(int inputSource     = 0,

                            uint32_t sampleRate = 0,

                            int format          = 0,

                            uint32_t channels = AudioSystem::CHANNEL_IN_MONO,

                            uint32_t channels = AUDIO_CHANNEL_IN_MONO,

                            int frameCount      = 0,

                            uint32_t flags      = 0,

                            callback_t cbf = 0,

include/media/AudioSystem.h

+29 −310
Original line number Diff line number Diff line
@@ -21,10 +21,15 @@ 
#include <utils/threads.h>

#include <media/IAudioFlinger.h>



#include <hardware/audio.h>

#include <hardware/audio_policy.h>



/* XXX: Should be include by all the users instead */

#include <media/AudioParameter.h>



namespace android {



typedef void (*audio_error_callback)(status_t err);

typedef int audio_io_handle_t;



class IAudioPolicyService;

class String8;
@@ -33,155 +38,6 @@ class AudioSystem 
{

public:



    // must match android/media/AudioSystem.java STREAM_* constants

    enum stream_type {

        DEFAULT          =-1,

        VOICE_CALL       = 0,

        SYSTEM           = 1,

        RING             = 2,

        MUSIC            = 3,

        ALARM            = 4,

        NOTIFICATION     = 5,

        BLUETOOTH_SCO    = 6,

        ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker

        DTMF             = 8,

        TTS              = 9,

        NUM_STREAM_TYPES

    };



    // Audio sub formats (see AudioSystem::audio_format).

    enum pcm_sub_format {

        PCM_SUB_16_BIT          = 0x1, // must be 1 for backward compatibility

        PCM_SUB_8_BIT           = 0x2, // must be 2 for backward compatibility

    };



    // FIXME These sub_format enums are currently unused



    // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify

    // bit rate, stereo mode, version...

    enum mp3_sub_format {

        //TODO

    };



    // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned,

    // encoding mode for recording...

    enum amr_sub_format {

        //TODO

    };



    // AAC sub format field definition: specify profile or bitrate for recording...

    enum aac_sub_format {

        //TODO

    };



    // VORBIS sub format field definition: specify quality for recording...

    enum vorbis_sub_format {

        //TODO

    };



    // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits).

    // The main format indicates the main codec type. The sub format field indicates options and parameters

    // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate

    // or profile. It can also be used for certain formats to give informations not present in the encoded

    // audio stream (e.g. octet alignement for AMR).

    enum audio_format {

        INVALID_FORMAT      = -1,

        FORMAT_DEFAULT      = 0,

        PCM                 = 0x00000000, // must be 0 for backward compatibility

        MP3                 = 0x01000000,

        AMR_NB              = 0x02000000,

        AMR_WB              = 0x03000000,

        AAC                 = 0x04000000,

        HE_AAC_V1           = 0x05000000,

        HE_AAC_V2           = 0x06000000,

        VORBIS              = 0x07000000,

        MAIN_FORMAT_MASK    = 0xFF000000,

        SUB_FORMAT_MASK     = 0x00FFFFFF,

        // Aliases

        PCM_16_BIT          = (PCM|PCM_SUB_16_BIT),

        PCM_8_BIT          = (PCM|PCM_SUB_8_BIT)

    };





    // Channel mask definitions must be kept in sync with values in /media/java/android/media/AudioFormat.java

    enum audio_channels {

        // output channels

        CHANNEL_OUT_FRONT_LEFT = 0x4,

        CHANNEL_OUT_FRONT_RIGHT = 0x8,

        CHANNEL_OUT_FRONT_CENTER = 0x10,

        CHANNEL_OUT_LOW_FREQUENCY = 0x20,

        CHANNEL_OUT_BACK_LEFT = 0x40,

        CHANNEL_OUT_BACK_RIGHT = 0x80,

        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,

        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,

        CHANNEL_OUT_BACK_CENTER = 0x400,

        CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT,

        CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT),

        CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),

        CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER),

        CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),

        CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |

                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),

        CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |

                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER),



        // input channels

        CHANNEL_IN_LEFT = 0x4,

        CHANNEL_IN_RIGHT = 0x8,

        CHANNEL_IN_FRONT = 0x10,

        CHANNEL_IN_BACK = 0x20,

        CHANNEL_IN_LEFT_PROCESSED = 0x40,

        CHANNEL_IN_RIGHT_PROCESSED = 0x80,

        CHANNEL_IN_FRONT_PROCESSED = 0x100,

        CHANNEL_IN_BACK_PROCESSED = 0x200,

        CHANNEL_IN_PRESSURE = 0x400,

        CHANNEL_IN_X_AXIS = 0x800,

        CHANNEL_IN_Y_AXIS = 0x1000,

        CHANNEL_IN_Z_AXIS = 0x2000,

        CHANNEL_IN_VOICE_UPLINK = 0x4000,

        CHANNEL_IN_VOICE_DNLINK = 0x8000,

        CHANNEL_IN_MONO = CHANNEL_IN_FRONT,

        CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT),

        CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK|

                CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED|

                CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS |

                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)

    };



    // must match android/media/AudioSystem.java MODE_* values

    enum audio_mode {

        MODE_INVALID = -2,

        MODE_CURRENT = -1,

        MODE_NORMAL = 0,

        MODE_RINGTONE,

        MODE_IN_CALL,

        MODE_IN_COMMUNICATION,

        NUM_MODES  // not a valid entry, denotes end-of-list

    };



    enum audio_in_acoustics {

        AGC_ENABLE    = 0x0001,

        AGC_DISABLE   = 0,

        NS_ENABLE     = 0x0002,

        NS_DISABLE    = 0,

        TX_IIR_ENABLE = 0x0004,

        TX_DISABLE    = 0

    };



    // special audio session values

    enum audio_sessions {

        SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream

                                   // (value must be less than 0)

        SESSION_OUTPUT_MIX = 0,    // session for effects applied to output mix. These effects can

                                   // be moved by audio policy manager to another output stream

                                   // (value must be 0)

    };



    /* These are static methods to control the system-wide AudioFlinger

     * only privileged processes can have access to them

     */
@@ -206,7 +62,7 @@ public: 
    static status_t setStreamMute(int stream, bool mute);

    static status_t getStreamMute(int stream, bool* mute);



    // set audio mode in audio hardware (see AudioSystem::audio_mode)

    // set audio mode in audio hardware (see audio_mode_t)

    static status_t setMode(int mode);



    // returns true in *state if tracks are active on the specified stream or has been active
@@ -227,9 +83,9 @@ public: 
    static float linearToLog(int volume);

    static int logToLinear(float volume);



    static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);

    static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);

    static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);

    static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT);

    static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT);

    static status_t getOutputLatency(uint32_t* latency, int stream = AUDIO_STREAM_DEFAULT);



    static bool routedToA2dpOutput(int streamType);
@@ -247,93 +103,11 @@ public: 
    // - 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(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);

    static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = AUDIO_STREAM_DEFAULT);



    static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);



    static int newAudioSessionId();

    //

    // AudioPolicyService interface

    //



    enum audio_devices {

        // output devices

        DEVICE_OUT_EARPIECE = 0x1,

        DEVICE_OUT_SPEAKER = 0x2,

        DEVICE_OUT_WIRED_HEADSET = 0x4,

        DEVICE_OUT_WIRED_HEADPHONE = 0x8,

        DEVICE_OUT_BLUETOOTH_SCO = 0x10,

        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,

        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,

        DEVICE_OUT_BLUETOOTH_A2DP = 0x80,

        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,

        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,

        DEVICE_OUT_AUX_DIGITAL = 0x400,

        DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800,

        DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000,

        DEVICE_OUT_DEFAULT = 0x8000,

        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |

                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |

                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |

                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL |

                DEVICE_OUT_ANLG_DOCK_HEADSET | DEVICE_OUT_DGTL_DOCK_HEADSET |

                DEVICE_OUT_DEFAULT),

        DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |

                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),



        // input devices

        DEVICE_IN_COMMUNICATION = 0x10000,

        DEVICE_IN_AMBIENT = 0x20000,

        DEVICE_IN_BUILTIN_MIC = 0x40000,

        DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,

        DEVICE_IN_WIRED_HEADSET = 0x100000,

        DEVICE_IN_AUX_DIGITAL = 0x200000,

        DEVICE_IN_VOICE_CALL = 0x400000,

        DEVICE_IN_BACK_MIC = 0x800000,

        DEVICE_IN_DEFAULT = 0x80000000,



        DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |

                DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |

                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)

    };



    // device connection states used for setDeviceConnectionState()

    enum device_connection_state {

        DEVICE_STATE_UNAVAILABLE,

        DEVICE_STATE_AVAILABLE,

        NUM_DEVICE_STATES

    };



    // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)

    enum output_flags {

        OUTPUT_FLAG_INDIRECT = 0x0,

        OUTPUT_FLAG_DIRECT = 0x1

    };



    // device categories used for setForceUse()

    enum forced_config {

        FORCE_NONE,

        FORCE_SPEAKER,

        FORCE_HEADPHONES,

        FORCE_BT_SCO,

        FORCE_BT_A2DP,

        FORCE_WIRED_ACCESSORY,

        FORCE_BT_CAR_DOCK,

        FORCE_BT_DESK_DOCK,

        FORCE_ANALOG_DOCK,

        FORCE_DIGITAL_DOCK,

        NUM_FORCE_CONFIG,

        FORCE_DEFAULT = FORCE_NONE

    };



    // usages used for setForceUse(), must match AudioSystem.java

    enum force_use {

        FOR_COMMUNICATION,

        FOR_MEDIA,

        FOR_RECORD,

        FOR_DOCK,

        NUM_FORCE_USE

    };



    // types of io configuration change events received with ioConfigChanged()

    enum io_config_event {
@@ -364,40 +138,40 @@ public: 
    //

    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)

    //

    static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address);

    static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address);

    static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address);

    static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address);

    static status_t setPhoneState(int state);

    static status_t setRingerMode(uint32_t mode, uint32_t mask);

    static status_t setForceUse(force_use usage, forced_config config);

    static forced_config getForceUse(force_use usage);

    static audio_io_handle_t getOutput(stream_type stream,

    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);

    static audio_io_handle_t getOutput(audio_stream_type_t stream,

                                        uint32_t samplingRate = 0,

                                        uint32_t format = FORMAT_DEFAULT,

                                        uint32_t channels = CHANNEL_OUT_STEREO,

                                        output_flags flags = OUTPUT_FLAG_INDIRECT);

                                        uint32_t format = AUDIO_FORMAT_DEFAULT,

                                        uint32_t channels = AUDIO_CHANNEL_OUT_STEREO,

                                        audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);

    static status_t startOutput(audio_io_handle_t output,

                                AudioSystem::stream_type stream,

                                audio_stream_type_t stream,

                                int session = 0);

    static status_t stopOutput(audio_io_handle_t output,

                               AudioSystem::stream_type stream,

                               audio_stream_type_t stream,

                               int session = 0);

    static void releaseOutput(audio_io_handle_t output);

    static audio_io_handle_t getInput(int inputSource,

                                    uint32_t samplingRate = 0,

                                    uint32_t format = FORMAT_DEFAULT,

                                    uint32_t channels = CHANNEL_IN_MONO,

                                    audio_in_acoustics acoustics = (audio_in_acoustics)0);

                                    uint32_t format = AUDIO_FORMAT_DEFAULT,

                                    uint32_t channels = AUDIO_CHANNEL_IN_MONO,

                                    audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0);

    static status_t startInput(audio_io_handle_t input);

    static status_t stopInput(audio_io_handle_t input);

    static void releaseInput(audio_io_handle_t input);

    static status_t initStreamVolume(stream_type stream,

    static status_t initStreamVolume(audio_stream_type_t stream,

                                      int indexMin,

                                      int indexMax);

    static status_t setStreamVolumeIndex(stream_type stream, int index);

    static status_t getStreamVolumeIndex(stream_type stream, int *index);

    static status_t setStreamVolumeIndex(audio_stream_type_t stream, int index);

    static status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index);



    static uint32_t getStrategyForStream(stream_type stream);

    static uint32_t getDevicesForStream(stream_type stream);

    static uint32_t getStrategyForStream(audio_stream_type_t stream);

    static uint32_t getDevicesForStream(audio_stream_type_t stream);



    static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);

    static status_t registerEffect(effect_descriptor_t *desc,
@@ -411,17 +185,6 @@ public: 


    // ----------------------------------------------------------------------------



    static uint32_t popCount(uint32_t u);

    static bool isOutputDevice(audio_devices device);

    static bool isInputDevice(audio_devices device);

    static bool isA2dpDevice(audio_devices device);

    static bool isBluetoothScoDevice(audio_devices device);

    static bool isLowVisibility(stream_type stream);

    static bool isOutputChannel(uint32_t channel);

    static bool isInputChannel(uint32_t channel);

    static bool isValidFormat(uint32_t format);

    static bool isLinearPCM(uint32_t format);



private:



    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
@@ -473,50 +236,6 @@ private: 
    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;

};



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 AudioSystem::audio_devices

    //  keySamplingRate: to change sampling rate routing, value is an int

    //  keyFormat: to change audio format, value is an int in AudioSystem::audio_format

    //  keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels

    //  keyFrameCount: to change audio output frame count, value is an int

    //  keyInputSource: to change audio input source, value is an int in audio_source

    //     (defined in media/mediarecorder.h)

    static const char *keyRouting;

    static const char *keySamplingRate;

    static const char *keyFormat;

    static const char *keyChannels;

    static const char *keyFrameCount;

    static const char *keyInputSource;



    String8 toString();



    status_t add(const String8& key, const String8& value);

    status_t addInt(const String8& key, const int value);

    status_t addFloat(const String8& key, const float value);



    status_t remove(const String8& key);



    status_t get(const String8& key, String8& value);

    status_t getInt(const String8& key, int& value);

    status_t getFloat(const String8& key, float& value);

    status_t getAt(size_t index, String8& key, String8& value);



    size_t size() { return mParameters.size(); }



private:

    String8 mKeyValuePairs;

    KeyedVector <String8, String8> mParameters;

};



};  // namespace android



#endif  /*ANDROID_AUDIOSYSTEM_H_*/

include/media/AudioTrack.h

+3 −4
Original line number Diff line number Diff line
@@ -30,7 +30,6 @@ 
#include <binder/IMemory.h>

#include <utils/threads.h>





namespace android {



// ----------------------------------------------------------------------------
@@ -126,11 +125,11 @@ public: 
     * Parameters:

     *

     * streamType:         Select the type of audio stream this track is attached to

     *                     (e.g. AudioSystem::MUSIC).

     *                     (e.g. AUDIO_STREAM_MUSIC).

     * sampleRate:         Track sampling rate in Hz.

     * format:             Audio format (e.g AudioSystem::PCM_16_BIT for signed

     * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed

     *                     16 bits per sample).

     * channels:           Channel mask: see AudioSystem::audio_channels.

     * channels:           Channel mask: see audio_channels_t.

     * frameCount:         Total size of track PCM buffer in frames. This defines the

     *                     latency of the track.

     * flags:              Reserved for future use.