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Audio VOIP call Issue with TI Gingerbread 2.3.4

prateek arora
Intellectual 415 points
Other Parts Discussed in Thread: TLV320AIC3254

hi ,

 I am working on  DM37xx EVM , kernel 2.6.32 and TI Gingerbread 2.3.4 .

 and use TLV320AIC3254 audio codec.

 we are facing a problem in SIP call .

 when we initiate a SIP call then services (  Audio flinger , media server )   has been died when ALSA  try to play RINGTONE.

following are the logs that comes during SIP call.

D/AudioHardwareInterface(  988): setMode(RINGTONE)
D/TIALSAModule(  988): route called for devices 00000000 in mode 1...
E/TIALSAModule(  988): Why are we routing to a device that isn't supported by this object?!?!?!?!
D/TIALSAModule(  988): open called for devices 00000000 in mode 1...
E/TIALSAModule(  988): Unable to set buffer size to 8192:  Invalid argument
I/TIALSAModule(  988): Initialized ALSA PLAYBACK device default
W/AudioPolicyManagerALSA(  988): AudioPolicyManagerALSA::getStrategyForStream() Stub Called...
W/AudioPolicyManagerALSA(  988):  redirecting AudioPolicyManagerALSA::getStrategy()
D/TIALSAModule(  988): open called for devices 00000000 in mode 0...
E/TIALSAModule(  988): Failed to Initialize any ALSA PLAYBACK device: Unknown error: -16


W/AudioSystem( 1283): AudioFlinger server died!
W/IMediaDeathNotifier( 1283): media server died
W/MediaMetadataRetriever( 1259): MediaMetadataRetriever server died!
W/AudioSystem( 1282): AudioFlinger server died!
W/IMediaDeathNotifier( 1282): media server died
I/ServiceManager(  982): service 'media.audio_flinger' died
E/MediaPlayer( 1282): error (100, 0)
W/IMediaDeathNotifier( 1116): media server died
E/MediaPlayer( 1282): Error (100,0)
I/ServiceManager(  982): service 'media.player' died
I/ServiceManager(  982): service 'media.camera' died
I/ServiceManager(  982): service 'media.audio_policy' died
W/AudioSystem( 1282): AudioPolicyService server died!
W/AudioSystem( 1116): AudioFlinger server died!
W/AudioSystem( 1116): AudioPolicyService server died!

 

but when we received a call then after 3-4 sec , media server has been started and call is working fine .

but in this scenario we are not able to hear RINGTONE and also getting 3-5 sec delay after receiving a  call. 

Logs:

D/TIALSAModule( 1940): TI ALSA module opened
D/TIALSAModule( 1940): Initializing devices for TI ALSA module
I/CameraService( 1940): CameraService started (pid=1940)
D/AudioHardwareALSA( 1940): openOutputStream called for devices: 0x00000002
D/TIALSAModule( 1940): open called for devices 00000002 in mode 0...
D/StatusBarService( 1257): DISABLE_EXPAND: yes
D/StatusBarService( 1257): animateCollapse(): mExpanded=false mExpandedVisible=false mExpanded=false mAnimating=false mAnimY=0.0 mAnimVel=0.0
I/ServiceManager( 1282): Waiting for service media.audio_policy...
E/ALSAControl( 1940): Control 'HeadsetR Mixer AudioR1' cannot get element info: -2
E/ALSAControl( 1940): Control 'HeadsetL Mixer AudioL1' cannot get element info: -2
E/ALSAControl( 1940): Control 'Headset Playback Volume' cannot get element info: -2
I/TIALSAModule( 1940): Initialized ALSA PLAYBACK device default
I/AudioFlinger( 1940): AudioFlinger's thread 0x496c8 ready to run
D/TIALSAModule( 1940): route called for devices 00000002 in mode 0...
E/AudioHardwareALSA( 1940): CALLING STANDBY
E/AudioService( 1116): Media server started.
E/AudioHardwareALSA( 1940): setParameters INVALID OPERATION
D/AudioHardwareInterface( 1940): setMode(RINGTONE)
D/TIALSAModule( 1940): route called for devices 00000000 in mode 1...
E/TIALSAModule( 1940): Why are we routing to a device that isn't supported by this object?!?!?!?!
D/TIALSAModule( 1940): open called for devices 00000000 in mode 1...
E/TIALSAModule( 1940): Unable to set buffer size to 8192:  Invalid argument
I/TIALSAModule( 1940): Initialized ALSA PLAYBACK device default
E/AudioHardwareALSA( 1940): setParameters INVALID OPERATION
W/AudioPolicyManagerALSA( 1940): AudioPolicyManagerALSA::getStrategyForStream() Stub Called...
W/AudioPolicyManagerALSA( 1940):  redirecting AudioPolicyManagerALSA::getStrategy()

 

kindly suggest any solution as soon as possible because .VOIP call is critical for us

 

 

  • 0
    Prodigy 40 points

    We are facing some Audio VOIP call Issue with TI Gingerbread 2.3.4.Please find the attached logs taken in different scenarios.

     

    1.In ringing mode, unable to hear ringtone and voice delay was noticeable after receiving the call. The log taken under this scenario is attached(initialcall_delay.log). We have done some temporary fix to resolve this is issue.

     

    Changes:3554.initialcall_delay.log

     

    /hardware/alsa_sound/AudioHardwareALSA.cpp

     

    status_t AudioHardwareALSA::setMode(int mode) {

        status_t status = NO_ERROR;

     

    // temporary fix

        if( mode ==1)

         {// (mode==RINGTONE)

             mode = 0;

         }

     

        if (mode != mMode) {

            status = AudioHardwareBase::setMode(mode);

     

            if (status == NO_ERROR) {

                // take care of mode change.

                for(ALSAHandleList::iterator it = mDeviceList.begin();

                    it != mDeviceList.end(); ++it) {

                    status = mALSADevice->route(&(*it), it->curDev, mode);

                    if (status != NO_ERROR)

                        break;

                }

            }

        }

     

        return status;

    }

     

    2. When we switch the call from Hold to Unhold and vise versa 70% of the times media server and libasound crashes. Please find the attached call_hold_unhold.log.

     

    3. In case of VOIP call using Bluetooth the voice is not coming. Please find the attached bt_issue.log

    4.When keypad tone is enable and the user dial some number media server crashes

     

    Please analyze the attached logs and help us to resolve these issues.

     

     

    Thanks,

    Rinki0027.call_hold_unhold.log8816.bt_issue.log

  • 0 in reply to Rinki Kumari
    Prodigy 40 points

    We tried by changing the buffer size to 2048 in ti/omap3/modules/alsa/alsa_module.cpp and still mediaserver and lasound libraries are crashing. Please find the attached logs taken and provide your inputs.

     

    Changes done by HCL

     

    hardware/ti/omap3/modules/alsa/alsa_module.cpp

     

    #ifndef ALSA_DEFAULT_SAMPLE_RATE

    #define ALSA_DEFAULT_SAMPLE_RATE 8000 // in Hz #endif

     

    static alsa_handle_t _defaults[] = {

        {

            module      : 0,

            devices     : OMAP3_OUT_SCO,

            curDev      : 0,

            curMode     : 0,

            handle      : 0,

            format      : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT

            channels    : 2,

            //sampleRate  : DEFAULT_SAMPLE_RATE, // should ideally be 8000

            sampleRate  : 8000,

            latency     : 200000, // Desired Delay in usec

            bufferSize  : 2048, // Desired Number of samples

                    mmap            :0,

            modPrivate  : (void *)&setScoControls,

        },

     

    {

            module      : 0,

            devices     : OMAP3_OUT_DEFAULT,

            curDev      : 0,

            curMode     : 0,

            handle      : 0,3716.bt_call_incomming.txt

            format      : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT

            channels    : 2,

            sampleRate  : DEFAULT_SAMPLE_RATE,

            latency     : 200000, // Desired Delay in usec

            bufferSize  : 2048, // Desired Number of samples

                    mmap            :0,

            modPrivate  : (void *)&setDefaultControls,

        },

     

    Thanks,

    Rinki

    1307.hold_unhold.txt5074.incomming_call.txt2605.normaloutgoing_call.txt