Replicant

== Build system ==

 * root your phone

 * Backup your current system!!!

 * Check out the Dream Android repository following that howto: [http://source.android.com/documentation/building-for-dream],but be carefull...DO NOT put the files extracted by extract-files.sh in the build system...it would put proprietary files in your build,that you can't even redistribute,but keep wifi firmware for pushing it later if you can stand firmwares(/system/etc/wifi/Fw1251r1c.bin )

Android use a custom Makefile based buildsystem

 * Add the patches that can be find here [http://groups.fsf.org/wiki/Freest_hardware_comparisons/replicant]

 * Add this patch for sound:

{{{

diff --git a/include/hardware_legacy/AudioHardwareInterface.h b/include/hardware_legacy/AudioHardwareInterface.h

index 0aa3a95..371ee48 100644

--- a/include/hardware_legacy/AudioHardwareInterface.h

+++ b/include/hardware_legacy/AudioHardwareInterface.h

 -211,8 +211,7 @@ public:

                                 int format,

                                 int channelCount,

                                 uint32_t sampleRate,

-                                status_t *status,

-                                AudioSystem::audio_in_acoustics acoustics) = 0;

+                                status_t *status) = 0;

     /**This method dumps the state of the audio hardware */

     virtual status_t dumpState(int fd, const Vector<String16>& args) = 0;

diff --git a/libs/audioflinger/A2dpAudioInterface.cpp b/libs/audioflinger/A2dpAudioInterface.cpp

index b6d5078..15ce1dd 100644

--- a/libs/audioflinger/A2dpAudioInterface.cpp

+++ b/libs/audioflinger/A2dpAudioInterface.cpp

 -71,8 +71,7 @@ AudioStreamOut* A2dpAudioInterface::openOutputStream(

 }

 AudioStreamIn* A2dpAudioInterface::openInputStream(

-        int format, int channelCount, uint32_t sampleRate, status_t *status,

-        AudioSystem::audio_in_acoustics acoustics)

+        int format, int channelCount, uint32_t sampleRate, status_t *status)

 {

     if (status)

         *status = -1;

diff --git a/libs/audioflinger/A2dpAudioInterface.h b/libs/audioflinger/A2dpAudioInterface.h

index 7901a8c..e67f05f 100644

--- a/libs/audioflinger/A2dpAudioInterface.h

+++ b/libs/audioflinger/A2dpAudioInterface.h

 -58,8 +58,7 @@ public:

                                 int format,

                                 int channelCount,

                                 uint32_t sampleRate,

-                                status_t *status,

-                                AudioSystem::audio_in_acoustics acoustics);

+                                status_t *status);

 protected:

     virtual status_t    doRouting();

diff --git a/libs/audioflinger/AudioDumpInterface.h b/libs/audioflinger/AudioDumpInterface.h

index 9a94102..42204d6 100644

--- a/libs/audioflinger/AudioDumpInterface.h

+++ b/libs/audioflinger/AudioDumpInterface.h

 -78,9 +78,8 @@ public:

     virtual status_t    setParameter(const char* key, const char* value)

                             {return mFinalInterface->setParameter(key, value);}

-    virtual AudioStreamIn* openInputStream( int format, int channelCount, uint32_t sampleRate, status_t *status,

-                                            AudioSystem::audio_in_acoustics acoustics)

-                            {return mFinalInterface->openInputStream( format, channelCount, sampleRate, status, acoustics);}

+    virtual AudioStreamIn* openInputStream( int format, int channelCount, uint32_t sampleRate, status_t *status)

+                            {return mFinalInterface->openInputStream( format, channelCount, sampleRate, status);}

     virtual status_t    dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); }

diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp

index 43df7dd..f376b67 100644

--- a/libs/audioflinger/AudioFlinger.cpp

+++ b/libs/audioflinger/AudioFlinger.cpp

 -2435,8 +2435,7 @@ bool AudioFlinger::AudioRecordThread::threadLoop()

                     input = mAudioHardware->openInputStream(mRecordTrack->format(), 

                                     mRecordTrack->channelCount(), 

                                     mRecordTrack->sampleRate(), 

-                                    &mStartStatus,

-                                    (AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16));

+                                    &mStartStatus);

                     if (input != 0) {

                         inBufferSize = input->bufferSize();

                         inFrameCount = inBufferSize/input->frameSize();                        

diff --git a/libs/audioflinger/AudioHardwareGeneric.cpp b/libs/audioflinger/AudioHardwareGeneric.cpp

index 62beada..e455186 100644

--- a/libs/audioflinger/AudioHardwareGeneric.cpp

+++ b/libs/audioflinger/AudioHardwareGeneric.cpp

 -93,8 +93,7 @@ void AudioHardwareGeneric::closeOutputStream(AudioStreamOutGeneric* out) {

 }

 AudioStreamIn* AudioHardwareGeneric::openInputStream(

-        int format, int channelCount, uint32_t sampleRate, status_t *status,

-        AudioSystem::audio_in_acoustics acoustics)

+        int format, int channelCount, uint32_t sampleRate, status_t *status)

 {

     AutoMutex lock(mLock);

 -108,7 +107,7 @@ AudioStreamIn* AudioHardwareGeneric::openInputStream(

     // create new output stream

     AudioStreamInGeneric* in = new AudioStreamInGeneric();

-    status_t lStatus = in->set(this, mFd, format, channelCount, sampleRate, acoustics);

+    status_t lStatus = in->set(this, mFd, format, channelCount, sampleRate);

     if (status) {

         *status = lStatus;

     }

 -247,8 +246,7 @@ status_t AudioStreamInGeneric::set(

         int fd,

         int format,

         int channels,

-        uint32_t rate,

-        AudioSystem::audio_in_acoustics acoustics)

+        uint32_t rate)

 {

     // FIXME: remove logging

     LOGD("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, format, channels, rate);

diff --git a/libs/audioflinger/AudioHardwareGeneric.h b/libs/audioflinger/AudioHardwareGeneric.h

index c949aa1..64e0396 100644

--- a/libs/audioflinger/AudioHardwareGeneric.h

+++ b/libs/audioflinger/AudioHardwareGeneric.h

 -69,8 +69,7 @@ public:

             int mFd,

             int format,

             int channelCount,

-            uint32_t sampleRate,

-            AudioSystem::audio_in_acoustics acoustics);

+            uint32_t sampleRate);

     uint32_t    sampleRate() const { return 8000; }

     virtual size_t      bufferSize() const { return 320; }

 -115,8 +114,7 @@ public:

             int format,

             int channelCount,

             uint32_t sampleRate,

-            status_t *status,

-            AudioSystem::audio_in_acoustics acoustics);

+            status_t *status);

             void            closeOutputStream(AudioStreamOutGeneric* out);

             void            closeInputStream(AudioStreamInGeneric* in);

diff --git a/libs/audioflinger/AudioHardwareStub.cpp b/libs/audioflinger/AudioHardwareStub.cpp

index b13cb1c..683af04 100644

--- a/libs/audioflinger/AudioHardwareStub.cpp

+++ b/libs/audioflinger/AudioHardwareStub.cpp

 -57,10 +57,10 @@ AudioStreamOut* AudioHardwareStub::openOutputStream(

 AudioStreamIn* AudioHardwareStub::openInputStream(

         int format, int channelCount, uint32_t sampleRate,

-        status_t *status, AudioSystem::audio_in_acoustics acoustics)

+        status_t *status)

 {

     AudioStreamInStub* in = new AudioStreamInStub();

-    status_t lStatus = in->set(format, channelCount, sampleRate, acoustics);

+    status_t lStatus = in->set(format, channelCount, sampleRate);

     if (status) {

         *status = lStatus;

     }

 -143,8 +143,7 @@ status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)

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

-status_t AudioStreamInStub::set(int format, int channels, uint32_t rate,

-				AudioSystem::audio_in_acoustics acoustics)

+status_t AudioStreamInStub::set(int format, int channels, uint32_t rate)

 {

     if ((format == AudioSystem::PCM_16_BIT) &&

             (channels == channelCount()) &&

diff --git a/libs/audioflinger/AudioHardwareStub.h b/libs/audioflinger/AudioHardwareStub.h

index d406424..24736ed 100644

--- a/libs/audioflinger/AudioHardwareStub.h

+++ b/libs/audioflinger/AudioHardwareStub.h

 -43,7 +43,7 @@ public:

 class AudioStreamInStub : public AudioStreamIn {

 public:

-    virtual status_t    set(int format, int channelCount, uint32_t sampleRate, AudioSystem::audio_in_acoustics acoustics);

+    virtual status_t    set(int format, int channelCount, uint32_t sampleRate);

     virtual uint32_t    sampleRate() const { return 8000; }

     virtual size_t      bufferSize() const { return 320; }

     virtual int         channelCount() const { return 1; }

 -81,8 +81,7 @@ public:

                                 int format,

                                 int channelCount,

                                 uint32_t sampleRate,

-                                status_t *status,

-				AudioSystem::audio_in_acoustics acoustics);

+                                status_t *status);

 protected:

     virtual status_t    doRouting() { return NO_ERROR; }

diff --git a/libaudio/AudioHardware.cpp b/libaudio/AudioHardware.cpp

index 0ece3f7..88dd46f 100644

--- a/libaudio/AudioHardware.cpp

+++ b/libaudio/AudioHardware.cpp

 -36,12 +36,51 @@

 #define LOG_SND_RPC 0  // Set to 1 to log sound RPC's

 namespace android {

-static int audpre_index, tx_iir_index;

-static void * acoustic;

 const uint32_t AudioHardware::inputSamplingRates[] = {

         8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000

 };

+

+

+

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

+static int snd_get_endpoint( int cnt,msm_snd_endpoint * ept)

+{

+  int fd;

+  int status;

+  fd = open("/dev/msm_snd",O_RDWR);

+  if (fd < 0)

+    {

+      perror("Cannot open msm_snd device");

+      close(fd);

+      return -1;

+    }

+  status = ioctl(fd,SND_GET_ENDPOINT, ept);

+  close(fd);      

+  return status;

+}

+

+static int snd_get_num()

+{

+  int fd;

+  int status;

+  int mNumSndEndpoints;

+  fd = open("/dev/msm_snd",O_RDWR);

+  if (fd < 0)

+  {

+    perror("Cannot open msm_snd device");

+    return -1;

+  }

+

+  if(ioctl(fd,SND_GET_NUM_ENDPOINTS,&mNumSndEndpoints)<0 )

+  {

+    perror("get number of endpoints error");

+    close(fd);

+    return -1;

+  }

+  close(fd);

+  return mNumSndEndpoints;

+

+}

 AudioHardware::AudioHardware() :

     mInit(false), mMicMute(true), mBluetoothNrec(true), mBluetoothId(0),

 -54,44 +93,22 @@ AudioHardware::AudioHardware() :

     SND_DEVICE_HEADSET(-1),

     SND_DEVICE_HEADSET_AND_SPEAKER(-1)

 {

-

-    int (*snd_get_num)();

-    int (*snd_get_endpoint)(int, msm_snd_endpoint *);

-    int (*set_acoustic_parameters)();

+    LOGE("inside AudioHardware::AudioHardware()");

+    //int snd_get_num();

+    //int snd_get_endpoint(int cnt,msm_snd_endpoint *ept);

+    //int (*snd_get_num)();

+    //int (*snd_get_endpoint)(int, msm_snd_endpoint *);

     struct msm_snd_endpoint *ept;

-

-    acoustic = ::dlopen("/system/lib/libhtc_acoustic.so", RTLD_NOW);

-    if (acoustic == NULL ) {

-        LOGE("Could not open libhtc_acoustic.so");

-        return;

-    }

-

-    set_acoustic_parameters = (int (*)(void))::dlsym(acoustic, "set_acoustic_parameters");

-    if ((*set_acoustic_parameters) == 0 ) {

-        LOGE("Could not open set_acoustic_parameters()");

-        return;

-    }

-

-    int rc = set_acoustic_parameters();

-    if (rc < 0) {

-        LOGE("Could not set acoustic parameters to share memory: %d", rc);

-//        return;

-    }

-

-    snd_get_num = (int (*)(void))::dlsym(acoustic, "snd_get_num_endpoints");

-    if ((*snd_get_num) == 0 ) {

-        LOGE("Could not open snd_get_num()");

-//        return;

-    }

-

+    LOGE("before");

     mNumSndEndpoints = snd_get_num();

+    LOGE("after");

     LOGD("mNumSndEndpoints = %d", mNumSndEndpoints);

     mSndEndpoints = new msm_snd_endpoint[mNumSndEndpoints];

     mInit = true;

     LOGV("constructed %d SND endpoints)", mNumSndEndpoints);

-    ept = mSndEndpoints;

-    snd_get_endpoint = (int (*)(int, msm_snd_endpoint *))::dlsym(acoustic, "snd_get_endpoint");

+    ept = mSndEndpoints; //LOOK AT mSndEndpoints type...

+    

     if ((*snd_get_endpoint) == 0 ) {

         LOGE("Could not open snd_get_endpoint()");

         return;

 -121,7 +138,6 @@ AudioHardware::~AudioHardware()

     delete mInput;

     delete mOutput;

     delete [] mSndEndpoints;

-    ::dlclose(acoustic);

     mInit = false;

 }

 -168,8 +184,7 @@ void AudioHardware::closeOutputStream(AudioStreamOutMSM72xx* out) {

 }

 AudioStreamIn* AudioHardware::openInputStream(

-        int format, int channelCount, uint32_t sampleRate, status_t *status,

-        AudioSystem::audio_in_acoustics acoustic_flags)

+        int format, int channelCount, uint32_t sampleRate, status_t *status)

 {

     mLock.lock();

     // input stream already open?

 -182,7 +197,7 @@ AudioStreamIn* AudioHardware::openInputStream(

     }

     AudioStreamInMSM72xx* in = new AudioStreamInMSM72xx();

-    status_t lStatus = in->set(this, format, channelCount, sampleRate, acoustic_flags);

+    status_t lStatus = in->set(this, format, channelCount, sampleRate);

     if (status) {

         *status = lStatus;

     }

 -449,6 +464,42 @@ static int count_bits(uint32_t vector)

     return bits;

 }

+static int msm72xx_enable_audpp (uint16_t enable_mask)

+{

+  int fd;

+

+//  if (!audpp_filter_inited)

+//    return -1;

+

+  fd = open ("/dev/msm_pcm_ctl", O_RDWR);

+  if (fd < 0)

+    {

+      perror ("Cannot open audio device");

+      return -1;

+    }

+

+  if (enable_mask & ADRC_ENABLE)

+    enable_mask &= ~ADRC_ENABLE;

+  if (enable_mask & EQ_ENABLE)

+    enable_mask &= ~EQ_ENABLE;

+  if (enable_mask & RX_IIR_ENABLE)

+    enable_mask &= ~RX_IIR_ENABLE;

+

+  printf ("msm72xx_enable_audpp: 0x%04x", enable_mask);

+  if (ioctl (fd, AUDIO_ENABLE_AUDPP, &enable_mask) < 0)

+    {

+      perror ("enable audpp error");

+      close (fd);

+      return -1;

+    }

+

+  close (fd);

+  return 0;

+}

+

+

+

+

 status_t AudioHardware::doRouting()

 {

     Mutex::Autolock lock(mLock);

 -460,8 +511,7 @@ status_t AudioHardware::doRouting()

                  " picking closest possible route...", routes);

         }

     }

-    int (*msm72xx_enable_audpp)(int);

-    msm72xx_enable_audpp = (int (*)(int))::dlsym(acoustic, "msm72xx_enable_audpp");

+

     status_t ret = NO_ERROR;

     if (routes & AudioSystem::ROUTE_BLUETOOTH_SCO) {

         LOGI("Routing audio to Bluetooth PCM\n");

 -700,14 +750,12 @@ bool AudioHardware::AudioStreamOutMSM72xx::checkStandby()

 AudioHardware::AudioStreamInMSM72xx::AudioStreamInMSM72xx() :

     mHardware(0), mFd(-1), mState(AUDIO_INPUT_CLOSED), mRetryCount(0),

     mFormat(AUDIO_HW_IN_FORMAT), mChannelCount(AUDIO_HW_IN_CHANNELS), 

-    mSampleRate(AUDIO_HW_IN_SAMPLERATE), mBufferSize(AUDIO_HW_IN_BUFFERSIZE),

-    mAcoustics((AudioSystem::audio_in_acoustics)0)

+    mSampleRate(AUDIO_HW_IN_SAMPLERATE), mBufferSize(AUDIO_HW_IN_BUFFERSIZE)

 {

 }

 status_t AudioHardware::AudioStreamInMSM72xx::set(

-        AudioHardware* hw, int format, int channelCount, uint32_t sampleRate,

-        AudioSystem::audio_in_acoustics acoustic_flags)

+        AudioHardware* hw, int format, int channelCount, uint32_t sampleRate)

 {

     LOGV("AudioStreamInMSM72xx::set(%d, %d, %u)", format, channelCount, sampleRate);

     if (mFd >= 0) {

 -763,26 +811,7 @@ status_t AudioHardware::AudioStreamInMSM72xx::set(

     mHardware = hw;

     mHardware->setMicMute_nosync(false);

     mState = AUDIO_INPUT_OPENED;

-    audpre_index = calculate_audpre_table_index(sampleRate);

-    tx_iir_index = (audpre_index * 2) + (hw->checkOutputStandby() ? 0 : 1);

-    LOGD("audpre_index = %d, tx_iir_index = %d\n", audpre_index, tx_iir_index);

-

-    /**

-     * If audio-preprocessing failed, we should not block record.

-     */

-    int (*msm72xx_set_audpre_params)(int, int);

-    msm72xx_set_audpre_params = (int (*)(int, int))::dlsym(acoustic, "msm72xx_set_audpre_params");

-    status = msm72xx_set_audpre_params(audpre_index, tx_iir_index);

-    if (status < 0)

-        LOGE("Cannot set audpre parameters");

-

-    int (*msm72xx_enable_audpre)(int, int, int);

-    msm72xx_enable_audpre = (int (*)(int, int, int))::dlsym(acoustic, "msm72xx_enable_audpre");

-    mAcoustics = acoustic_flags;

-    status = msm72xx_enable_audpre((int)acoustic_flags, audpre_index, tx_iir_index);

-    if (status < 0)

-        LOGE("Cannot enable audpre");

-

+    

     return NO_ERROR;

 Error:

 -811,7 +840,7 @@ ssize_t AudioHardware::AudioStreamInMSM72xx::read( void* buffer, ssize_t bytes)

     uint8_t* p = static_cast<uint8_t*>(buffer);

     if (mState < AUDIO_INPUT_OPENED) {

-        if (set(mHardware, mFormat, mChannelCount, mSampleRate, mAcoustics) != NO_ERROR) {

+        if (set(mHardware, mFormat, mChannelCount, mSampleRate) != NO_ERROR) {

             return -1;

         }

     }

diff --git a/libaudio/AudioHardware.h b/libaudio/AudioHardware.h

index 047bad3..2111e16 100644

--- a/libaudio/AudioHardware.h

+++ b/libaudio/AudioHardware.h

 -128,8 +128,8 @@ public:

                                 int format,

                                 int channelCount,

                                 uint32_t sampleRate,

-                                status_t *status,

-                                AudioSystem::audio_in_acoustics acoustics);

+                                status_t *status

+					   );

                void        closeOutputStream(AudioStreamOutMSM72xx* out);

                void        closeInputStream(AudioStreamInMSM72xx* in);

 -190,8 +190,8 @@ private:

                 status_t    set(AudioHardware* mHardware,

                                 int format,

                                 int channelCount,

-                                uint32_t sampleRate,

-                                AudioSystem::audio_in_acoustics acoustics);

+                                uint32_t sampleRate

+				);

         virtual size_t      bufferSize() const { return mBufferSize; }

         virtual int         channelCount() const { return mChannelCount; }

         virtual int         format() const { return mFormat; }

 -210,7 +210,7 @@ private:

                 int         mChannelCount;

                 uint32_t    mSampleRate;

                 size_t      mBufferSize;

-                AudioSystem::audio_in_acoustics mAcoustics;

+

     };

             static const uint32_t inputSamplingRates[];

}}}

Build process:

Let's say that our build directory is ~/replicant

 * Add that file:

{{{

TARGET_PRODUCT := htc_dream

}}}

 To ~/replicant/buildspec.mk 

 * Then type

{{{

 cd ~/replicant

 export ANDROID_JAVA_HOME=$JAVA_HOME

 make

}}}

* The files to flash are in ~/replicant/out/target/product/dream,flash them and then clear the cache

* boot and push the wifi firmware if you want it

== Status ==

 * web browser crashes

 * ril seem to work(I made some calls and received sms)

 *  sound on the application cpu(the baseband cpu has sound too) now works