libaudio: code cleanup & RIL interface update

Anton Rogozin <ant.rogozin@samsung.com>:
 Unused code removed, dynamic loading of RIL lib, alsa lib resampler turning on

UK KIM <w0806.kim@samsung.com>:
1. deleted unneeded standby_l() and set_wakelock func() in each stream class.
2. use StreamOPS:close() to close stream.
3. reference new libaudio.

Eric Laurent <elaurent@google.com>:
Some more cleanup
Fixed clicks when playing call ended tone

Change-Id: Ieea1319262576b2f6680c675957643eacbab9c11

1 files changed, 68 insertions, 156 deletions

diff --git a/libaudio/AudioHardwareALSA.h b/libaudio/AudioHardwareALSA.h
index 0390beb..f65aa68 100755
--- a/libaudio/AudioHardwareALSA.h
+++ b/libaudio/AudioHardwareALSA.h
@@ -24,73 +24,9 @@
 
 #include <hardware_legacy/AudioHardwareBase.h>
 
-#if defined SEC_IPC
-#include <hardware/hardware.h>
-
 // sangsu fix : headers for IPC
 #include "secril-client.h"
 
-// sangsu fix : defines for IPC
-#define OEM_FUNCTION_ID_SOUND 0x08 // sound Main Cmd
-
-//sangsu fix : sound sub command for IPC
-#define OEM_SOUND_SET_VOLUME_CTRL 0x03
-#define OEM_SOUND_GET_VOLUME_CTRL 0x04
-#define OEM_SOUND_SET_AUDIO_PATH_CTRL 0x05
-#define OEM_SOUND_GET_AUDIO_PATH_CTRL 0x06 
-
-//sangsu fix : audio path for IPC
-#define OEM_SOUND_AUDIO_PATH_HANDSET     0x01 
-#define OEM_SOUND_AUDIO_PATH_HEADSET     0x02 
-#define OEM_SOUND_AUDIO_PATH_HANDFREE   0x03 
-#define OEM_SOUND_AUDIO_PATH_BLUETOOTH 0x04
-#define OEM_SOUND_AUDIO_PATH_STREOBT      0x05
-#define OEM_SOUND_AUDIO_PATH_SPEAKER      0x06 
-#define OEM_SOUND_AUDIO_PATH_HEADSET35 0x07 
-#define OEM_SOUND_AUDIO_PATH_BT_NSEC_OFF 0x08
-
-// sangsu fix : volume level for IPC
-#define OEM_SOUND_VOLUME_LEVEL_MUTE  0x00 
-#define OEM_SOUND_VOLUME_LEVEL1           0x01
-#define OEM_SOUND_VOLUME_LEVEL2           0x02
-#define OEM_SOUND_VOLUME_LEVEL3           0x03
-#define OEM_SOUND_VOLUME_LEVEL4           0x04
-#define OEM_SOUND_VOLUME_LEVEL5           0x05
-#define OEM_SOUND_VOLUME_LEVEL6           0x06
-#define OEM_SOUND_VOLUME_LEVEL7           0x07
-#define OEM_SOUND_VOLUME_LEVEL8           0x08
-
-// For synchronizing I2S clocking
-#if defined SYNCHRONIZE_CP
-#define OEM_SOUND_SET_CLOCK_CTRL	0x0A
-#define OEM_SOUND_CLOCK_START		0x01
-#define OEM_SOUND_CLOCK_STOP		0x00
-#endif
-
-// For VT
-#if defined VIDEO_TELEPHONY
-#define OEM_SOUND_VIDEO_CALL_STOP	0x00
-#define OEM_SOUND_VIDEO_CALL_START	0x01
-#define OEM_SOUND_SET_VIDEO_CALL_CTRL	0x07
-#endif
-
-// sangsu fix : volume type for IPC
-#define OEM_SOUND_TYPE_VOICE           0x01 // Receiver(0x00) + Voice(0x01)
-#define OEM_SOUND_TYPE_KEYTONE      0x02 // Receiver(0x00) + Key tone (0x02)
-#define OEM_SOUND_TYPE_BELL             0x03 // Receiver(0x00) + Bell (0x03)
-#define OEM_SOUND_TYPE_MESSAGE      0x04 // Receiver(0x00) + Message(0x04)
-#define OEM_SOUND_TYPE_ALARM          0x05 // Receiver(0x00) + Alarm (0x05)
-#define OEM_SOUND_TYPE_SPEAKER       0x11 // SpeakerPhone (0x10) + Voice(0x01)
-#define OEM_SOUND_TYPE_HFKVOICE     0x21 // HFK   (0x20) + Voice(0x01)
-#define OEM_SOUND_TYPE_HFKKEY         0x22 // HFK   (0x20) + Key tone (0x02)
-#define OEM_SOUND_TYPE_HFKBELL        0x23 // HFK   (0x20) + Bell (0x03)
-#define OEM_SOUND_TYPE_HFKMSG         0x24 // HFK   (0x20) + Message(0x04)
-#define OEM_SOUND_TYPE_HFKALARM    0x25 // HFK   (0x20) + Alarm (0x05)
-#define OEM_SOUND_TYPE_HFKPDA         0x26 // HFK   (0x20) + PDA miscellaneous sound (0x06)
-#define OEM_SOUND_TYPE_HEADSET       0x31 // Headset (0x30) + Voice(0x01)
-#define OEM_SOUND_TYPE_BTVOICE        0x41 // BT(0x40) + Voice(0x01)
-#endif
-
 #ifndef ALSA_DEFAULT_SAMPLE_RATE
 #define ALSA_DEFAULT_SAMPLE_RATE 44100 // in Hz
 #endif
@@ -112,7 +48,6 @@
 namespace android
 {
 
-
     class AudioHardwareALSA;
 
     // ----------------------------------------------------------------------------
@@ -181,13 +116,13 @@ namespace android
             status_t                sampleRate(uint32_t rate);
             virtual size_t          bufferSize() const;
             virtual int             format() const;
-                    int 		    getAndroidFormat(snd_pcm_format_t format);
+                    int             getAndroidFormat(snd_pcm_format_t format);
 
             virtual uint32_t        channels() const;
                     int             channelCount() const;
                     status_t        channelCount(int channelCount);
                     uint32_t        getAndroidChannels(int channelCount) const;
-                  
+
             status_t                open(int mode, uint32_t device);
             void                    close();
             status_t                setSoftwareParams();
@@ -209,7 +144,6 @@ namespace android
             snd_pcm_t              *mHandle;
             snd_pcm_hw_params_t    *mHardwareParams;
             snd_pcm_sw_params_t    *mSoftwareParams;
-            int                     mMode;
             uint32_t                mDevice;
 
             StreamDefaults         *mDefaults;
@@ -224,9 +158,9 @@ namespace android
             virtual                ~AudioStreamOutALSA();
 
 
-            status_t		     set(int *format,
-                         uint32_t *channelCount,
-                         uint32_t *sampleRate){
+            status_t                set(int *format,
+                                        uint32_t *channelCount,
+                                        uint32_t *sampleRate){
                 return ALSAStreamOps::set(format, channelCount, sampleRate);
             }
 
@@ -239,7 +173,7 @@ namespace android
                 return ALSAStreamOps::bufferSize();
             }
 
-            virtual uint32_t             channels() const
+            virtual uint32_t        channels() const
             {
                 return ALSAStreamOps::channels();
             }
@@ -254,18 +188,16 @@ namespace android
             virtual ssize_t         write(const void *buffer, size_t bytes);
             virtual status_t        dump(int fd, const Vector<String16>& args);
             virtual status_t        setDevice(int mode, uint32_t newDevice, uint32_t audio_mode);
-	    virtual status_t    setVolume(float left, float right); //Tushar: New arch
+            virtual status_t        setVolume(float left, float right); //Tushar: New arch
 
             status_t                setVolume(float volume);
 
             status_t                standby();
-            bool                    isStandby();
-            void                    setWakeLock();
 
-			virtual status_t    setParameters(const String8& keyValuePairs);
-			virtual String8     getParameters(const String8& keys);
-	
-			virtual status_t    getRenderPosition(uint32_t *dspFrames);
+            virtual status_t        setParameters(const String8& keyValuePairs);
+            virtual String8         getParameters(const String8& keys);
+
+            virtual status_t        getRenderPosition(uint32_t *dspFrames);
 
 
         private:
@@ -279,9 +211,9 @@ namespace android
                                     AudioStreamInALSA(AudioHardwareALSA *parent);
             virtual                ~AudioStreamInALSA();
 
-            status_t		     set(int *format,
-                         uint32_t *channelCount,
-                         uint32_t *sampleRate){
+            status_t                set(int *format,
+                                        uint32_t *channelCount,
+                                        uint32_t *sampleRate) {
                 return ALSAStreamOps::set(format, channelCount, sampleRate);
             }
 
@@ -294,7 +226,7 @@ namespace android
                 return ALSAStreamOps::bufferSize();
             }
 
-            virtual uint32_t             channels() const
+            virtual uint32_t        channels() const
             {
                 return ALSAStreamOps::channels();
             }
@@ -311,37 +243,18 @@ namespace android
             virtual status_t        setGain(float gain);
 
             virtual status_t        standby();
-                    status_t        standby_l();
-                    void            setWakeLock();
 
-	    virtual status_t    setParameters(const String8& keyValuePairs);
-	    virtual String8     getParameters(const String8& keys);
+            virtual status_t        setParameters(const String8& keyValuePairs);
+            virtual String8         getParameters(const String8& keys);
 
-		virtual unsigned int  getInputFramesLost() const { return 0; }
+            virtual unsigned int    getInputFramesLost() const { return 0; }
 
         private:
-            AudioHardwareALSA *mParent;
-            bool               mPowerLock;
-            int16_t            mBuffer[2 * PERIOD_SZ_CAPTURE];
+            AudioHardwareALSA      *mParent;
+            bool                    mPowerLock;
+            int16_t                 mBuffer[2 * PERIOD_SZ_CAPTURE];
     };
 
-#if defined SEC_IPC
-    //TODO..implementation has to be done
-    class AudioHardwareIPC
-    {
-	public:
-					AudioHardwareIPC();
-		virtual		~AudioHardwareIPC(); 	
-		status_t		transmitVolumeIPC(uint32_t type, float volume);
-		status_t		transmitAudioPathIPC(uint32_t path);
-#if defined SYNCHRONIZE_CP
-		status_t		transmitClock_IPC(uint32_t condition);
-#endif
-	private:
-		HRilClient		mClient;
-		char			data[100];	
-    };
-#endif
     class AudioHardwareALSA : public AudioHardwareBase
     {
         public:
@@ -373,34 +286,32 @@ namespace android
             // mic mute
             virtual status_t        setMicMute(bool state);
             virtual status_t        getMicMute(bool* state);
-		virtual size_t getInputBufferSize(
-			uint32_t sampleRate,
-			int format,
-			int channelCount);
-#if defined TURN_ON_DEVICE_ONLY_USE
-                virtual int             setMicStatus(int on);   // To deliver status of input stream(activated or not). If it's activated, doesn't turn off codec.
-#endif
- 	   /** This method creates and opens the audio hardware output stream */
-	    virtual AudioStreamOut* openOutputStream(
-                                uint32_t devices,
-                                int *format=0,
-                                uint32_t *channels=0,
-                                uint32_t *sampleRate=0,
-                                status_t *status=0);
-	    virtual    void        closeOutputStream(AudioStreamOut* out);
-
-	    /** This method creates and opens the audio hardware input stream */
-	    virtual AudioStreamIn* openInputStream(
-                                uint32_t devices,
-                                int *format,
-                                uint32_t *channels,
-                                uint32_t *sampleRate,
-                                status_t *status,
-                                AudioSystem::audio_in_acoustics acoustics);
-	    virtual    void        closeInputStream(AudioStreamIn* in);
-
-	    static      uint32_t    checkInputSampleRate(uint32_t sampleRate);
-        static const uint32_t inputSamplingRates[];
+            virtual size_t          getInputBufferSize(uint32_t sampleRate,
+                                                       int format,
+                                                       int channelCount);
+            virtual int             setMicStatus(int on);   // To deliver status of input stream(activated or not). If it's activated, doesn't turn off codec.
+
+            /** This method creates and opens the audio hardware output stream */
+            virtual AudioStreamOut* openOutputStream(uint32_t devices,
+                                                     int *format = 0,
+                                                     uint32_t *channels = 0,
+                                                     uint32_t *sampleRate = 0,
+                                                     status_t *status = 0);
+            virtual void            closeOutputStream(AudioStreamOut* out);
+
+            /** This method creates and opens the audio hardware input stream */
+            virtual AudioStreamIn*  openInputStream(uint32_t devices,
+                                                   int *format,
+                                                   uint32_t *channels,
+                                                   uint32_t *sampleRate,
+                                                   status_t *status,
+                                                   AudioSystem::audio_in_acoustics acoustics);
+            virtual void            closeInputStream(AudioStreamIn* in);
+
+            static uint32_t         checkInputSampleRate(uint32_t sampleRate);
+            static const uint32_t   inputSamplingRates[];
+
+                   int              mode() { return mMode; }
 
         protected:
             /**
@@ -409,34 +320,35 @@ namespace android
              * doRouting when required. If the device has any special requirements these
              * methods can be overriden.
              */
-            virtual status_t    doRouting(uint32_t device);
+            virtual status_t        doRouting(uint32_t device);
 
-            virtual status_t    dump(int fd, const Vector<String16>& args);
+            virtual status_t        dump(int fd, const Vector<String16>& args);
 
             friend class AudioStreamOutALSA;
             friend class AudioStreamInALSA;
 
-            ALSAMixer          *mMixer;
-            AudioStreamOutALSA *mOutput;
-            AudioStreamInALSA  *mInput;
-#if defined SEC_IPC
-	    AudioHardwareIPC   *mIPC; //for IPC	    
-	    uint32_t        mRoutes[AudioSystem::NUM_MODES];
-#endif
+            ALSAMixer              *mMixer;
+            AudioStreamOutALSA     *mOutput;
+            AudioStreamInALSA      *mInput;
 
+            uint32_t                mRoutes[AudioSystem::NUM_MODES];
         private:
-            Mutex               mLock;
-#if defined TURN_ON_DEVICE_ONLY_USE
-                bool            mActivatedInputDevice;
-#endif
+            Mutex                   mLock;
+            bool                    mActivatedInputDevice;
+
+            void                   *mSecRilLibHandle;
+            HRilClient              mRilClient;
+            HRilClient (*openClientRILD)  (void);
+            int        (*disconnectRILD)  (HRilClient);
+            int        (*closeClientRILD) (HRilClient);
+            int        (*isConnectedRILD) (HRilClient);
+            int        (*connectRILD)     (HRilClient);
+            int        (*setCallVolume)   (HRilClient, SoundType, int);
+            int        (*setCallAudioPath)(HRilClient, AudioPath);
+
+            void                    loadRILD(void);
+            status_t                connectRILDIfRequired(void);
     };
 
-    // ----------------------------------------------------------------------------
-
-#if defined SEC_IPC
-// sangsu fix : global functions for IPC
-static int onRawReqComplete(HRilClient client, const void *data, size_t datalen);
-static int onUnsol(HRilClient client, const void *data, size_t datalen);
-#endif
 };        // namespace android
 #endif    // ANDROID_AUDIO_HARDWARE_ALSA_H