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VoIPVoice.c
资源名称：mmi.rar　[点击查看]
上传用户：lqx1163
上传日期：2014-08-13
资源大小：9183k
文件大小：34k
源码类别：

MTK

开发平台：

C/C++

VoIPVoice.c：源码内容
  1. /*****************************************************************************
  2. *  Copyright Statement:
  3. *  --------------------
  4. *  This software is protected by Copyright and the information contained
  5. *  herein is confidential. The software may not be copied and the information
  6. *  contained herein may not be used or disclosed except with the written
  7. *  permission of MediaTek Inc. (C) 2005
  8. *
  9. *  BY OPENING THIS FILE, BUYER HEREBY UNEQUIVOCALLY ACKNOWLEDGES AND AGREES
  10. *  THAT THE SOFTWARE/FIRMWARE AND ITS DOCUMENTATIONS ("MEDIATEK SOFTWARE")
  11. *  RECEIVED FROM MEDIATEK AND/OR ITS REPRESENTATIVES ARE PROVIDED TO BUYER ON
  12. *  AN "AS-IS" BASIS ONLY. MEDIATEK EXPRESSLY DISCLAIMS ANY AND ALL WARRANTIES,
  13. *  EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE IMPLIED WARRANTIES OF
  14. *  MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE OR NONINFRINGEMENT.
  15. *  NEITHER DOES MEDIATEK PROVIDE ANY WARRANTY WHATSOEVER WITH RESPECT TO THE
  16. *  SOFTWARE OF ANY THIRD PARTY WHICH MAY BE USED BY, INCORPORATED IN, OR
  17. *  SUPPLIED WITH THE MEDIATEK SOFTWARE, AND BUYER AGREES TO LOOK ONLY TO SUCH
  18. *  THIRD PARTY FOR ANY WARRANTY CLAIM RELATING THERETO. MEDIATEK SHALL ALSO
  19. *  NOT BE RESPONSIBLE FOR ANY MEDIATEK SOFTWARE RELEASES MADE TO BUYER'S
  20. *  SPECIFICATION OR TO CONFORM TO A PARTICULAR STANDARD OR OPEN FORUM.
  21. *
  22. *  BUYER'S SOLE AND EXCLUSIVE REMEDY AND MEDIATEK'S ENTIRE AND CUMULATIVE
  23. *  LIABILITY WITH RESPECT TO THE MEDIATEK SOFTWARE RELEASED HEREUNDER WILL BE,
  24. *  AT MEDIATEK'S OPTION, TO REVISE OR REPLACE THE MEDIATEK SOFTWARE AT ISSUE,
  25. *  OR REFUND ANY SOFTWARE LICENSE FEES OR SERVICE CHARGE PAID BY BUYER TO
  26. *  MEDIATEK FOR SUCH MEDIATEK SOFTWARE AT ISSUE. 
  27. *
  28. *  THE TRANSACTION CONTEMPLATED HEREUNDER SHALL BE CONSTRUED IN ACCORDANCE
  29. *  WITH THE LAWS OF THE STATE OF CALIFORNIA, USA, EXCLUDING ITS CONFLICT OF
  30. *  LAWS PRINCIPLES.  ANY DISPUTES, CONTROVERSIES OR CLAIMS ARISING THEREOF AND
  31. *  RELATED THERETO SHALL BE SETTLED BY ARBITRATION IN SAN FRANCISCO, CA, UNDER
  32. *  THE RULES OF THE INTERNATIONAL CHAMBER OF COMMERCE (ICC).
  33. *
  34. *****************************************************************************/
  35. /*****************************************************************************
  36.  *
  37.  * Filename:
  38.  * ---------
  39.  *  VoIPVoice.c
  40.  *
  41.  * Project:
  42.  * --------
  43.  *  MAUI
  44.  *
  45.  * Description:
  46.  * ------------
  47.  *  Coding Template header file
  48.  *
  49.  * Author:
  50.  * -------
  51.  * -------
  52.  *
  53.  *============================================================================
  54.  *             HISTORY
  55.  * Below this line, this part is controlled by PVCS VM. DO NOT MODIFY!!
  56.  *------------------------------------------------------------------------------
  57.  * removed!
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  110.  *
  111.  *------------------------------------------------------------------------------
  112.  * Upper this line, this part is controlled by PVCS VM. DO NOT MODIFY!!
  113.  *============================================================================
  114.  ****************************************************************************/
  115. #ifndef __MTK_TARGET__
  116. #include <windows.h>
  117. #endif 
  118. #include "MMI_features.h"
  120. #ifdef __MMI_VOIP__
  122. #include "stdC.h"
  123. #include "L4Dr1.h"
  124. #include "DebugInitDef.h"
  125. #include "mmi_trc.h"            /* debug info */
  126. #include "GlobalMenuItems.h"
  127. #include "GlobalScrEnum.h"
  128. #include "CustMenuRes.h"
  129. #include "CustDataRes.h"
  130. #include "ProtocolEvents.h"
  131. #include "CommonScreens.h"
  132. #include "SettingProfile.h"
  133. #include "EventsGprot.h"
  134. #include "wgui_categories_popup.h"
  135. #include "wgui_categories_inputs.h"
  136. #include "wgui_categories_util.h"
  137. #include "NVRAMEnum.h"
  138. #include "NVRAMProt.h"
  139. #include "NVRAMType.h"
  140. #include "custom_nvram_editor_data_item.h"
  141. #include "custom_data_account.h"
  143. #include "AlarmFrameWorkProt.h"
  144. #include "CallManagementGprot.h"
  145. #include "gpioInc.h"
  146. #include "mdi_datatype.h"
  147. #include "mdi_audio.h"
  148. #include "ProfileGprots.h"
  149. #include "wgui_status_icons.h"
  150. #include "app2soc_struct.h"
  151. #include "soc_api.h"
  152. #include "med_struct.h"
  153. #include "rtp_api.h"
  154. #include "device.h"         /* for call_status_req enum */
  156. #include "VoIPDef.h"
  157. #include "VoIPGProt.h"
  158. #include "VoIPProt.h"
  161. /*****************************************************************************
  162.  * FUNCTION
  163.  *  mmi_voip_switch_session
  164.  * DESCRIPTION
  165.  *  Start critical session based on the current state. For bluetooth SCO link,
  166.  *  session is started when outgoing call is dialing or incoming call is ringing;
  167.  *  session is ended when the last call is disconnected.
  168.  *  For MDI sound control, session is started when outgoing call is dialing or
  169.  *  incoming call is answering; session is ended when the last call is disconnected or 
  170.  *  only one incoming call exists, because session should stop first in order to play ringtone.
  171.  * PARAMETERS
  172.  *  currState       [IN]        Current state to switch session
  173.  * RETURNS
  174.  *  void
  175.  *****************************************************************************/
  176. void mmi_voip_switch_session(mmi_voip_call_state_enum currState)
  177. {
  178.     /*----------------------------------------------------------------*/
  179.     /* Local Variables                                                */
  180.     /*----------------------------------------------------------------*/
  182.     /*----------------------------------------------------------------*/
  183.     /* Code Body                                                      */
  184.     /*----------------------------------------------------------------*/
  185.     switch (currState)
  186.     {
  187.         case MMI_VOIP_OUTGOING_STATE:
  188.             if (g_voip_cntx_p->call_list_info.numTotal == 1)
  189.             {
  190.                 AlmDisableExpiryHandler();
  191.             #ifdef __MMI_BT_PROFILE__
  192.                 mmi_profiles_bt_call_start_callback();
  193.             #endif
  194.                 mmi_voip_call_status_req(UEM_CALL_SETUP); /* trigger uem to send headset key gpio detection */
  196.                 if (g_voip_cntx_p->call_list_info.inSession == FALSE)
  197.                 {
  198.                     g_voip_cntx_p->call_list_info.inSession = TRUE;
  199.                     mdi_audio_speech_session_start(MDI_AUDIO_SPEECH_APP_ID_VOIP); /* set mdi speech mode on */
  200.                     /* mmi_sndrec_auto_record_switch(MDI_AUDIO_SPEECH_APP_ID_VOIP, MMI_TRUE); */
  201.                 }
  202.             }
  203.             break;
  204.         case MMI_VOIP_INCOMING_STATE:
  205.             if (g_voip_cntx_p->call_list_info.numTotal == 1)
  206.             {
  207.                 AlmDisableExpiryHandler();
  208.             #ifdef __MMI_BT_PROFILE__
  209.                 mmi_profiles_bt_call_start_callback();
  210.             #endif
  211.                 mmi_voip_call_status_req(UEM_CALL_SETUP); /* trigger uem to send headset key gpio detection */
  212.             }
  213.             break;
  214.         case MMI_VOIP_ACTIVE_STATE:
  215.             if (g_voip_cntx_p->call_list_info.inSession == FALSE)
  216.             {
  217.                 g_voip_cntx_p->call_list_info.inSession = TRUE;
  218.                 mdi_audio_speech_session_start(MDI_AUDIO_SPEECH_APP_ID_VOIP); /* set mdi speech mode on */
  219.                 /* mmi_sndrec_auto_record_switch(MDI_AUDIO_SPEECH_APP_ID_VOIP, MMI_TRUE); */
  220.             }
  221.             break;
  222.         case MMI_VOIP_IDLE_STATE:
  223.             /* incoming call needs to play ringtone, so mdi session should stop first. 
  224.                when accepting call, mdi session should not stop */
  225.             if ((g_voip_cntx_p->call_list_info.numTotal == 1) && 
  226.                 (mmi_voip_get_incoming_call_id() != -1) && 
  227.                 (g_voip_cntx_p->call_list_info.processCId == -1))
  228.             {
  229.                 if (g_voip_cntx_p->call_list_info.inSession == TRUE)
  230.                 {
  231.                     /* mmi_sndrec_auto_record_switch(MDI_AUDIO_SPEECH_APP_ID_VOIP, MMI_FALSE); */
  232.                     mdi_audio_speech_session_stop(MDI_AUDIO_SPEECH_APP_ID_VOIP); /* set mdi speech mode off */
  233.                     g_voip_cntx_p->call_list_info.inSession = FALSE;
  234.                 }
  235.             }
  236.             else if (g_voip_cntx_p->call_list_info.numTotal == 0)
  237.             {
  238.                 if (g_voip_cntx_p->call_misc_info.isMute == TRUE)
  239.                 {
  240.                     g_voip_cntx_p->call_misc_info.isMute = FALSE;
  241.                     MuteOffMicrophone();
  242.                     HideStatusIcon(STATUS_ICON_MUTE);
  243.                     UpdateStatusIcons();
  244.                 }
  245.                 if (g_voip_cntx_p->call_misc_info.isLoud == TRUE)
  246.                 {
  247.                     g_voip_cntx_p->call_misc_info.isLoud = FALSE;
  248.                     DisbleHandsFree();
  249.                     SetLoudSpeaker(FALSE);
  250.                 }
  251.                 /* reset common structure to prevent misuse by next call */
  252.                 memset(&g_voip_cntx_p->call_misc_info, 0, sizeof(mmi_voip_call_misc_struct));
  253.                 g_voip_cntx_p->call_misc_info.isDtmf = TRUE;
  255.                 if (g_voip_cntx_p->call_list_info.inSession == TRUE)
  256.                 {
  257.                     /* mmi_sndrec_auto_record_switch(MDI_AUDIO_SPEECH_APP_ID_VOIP, MMI_FALSE); */
  258.                     mdi_audio_speech_session_stop(MDI_AUDIO_SPEECH_APP_ID_VOIP); /* set mdi speech mode off */
  259.                     g_voip_cntx_p->call_list_info.inSession = FALSE;
  260.                 }
  262.                 mmi_voip_call_status_req(UEM_CALL_DISCONNECT); /* trigger uem to send headset key gpio detection */
  263.             #ifdef __MMI_BT_PROFILE__
  264.                 /* do not turn off sco link in case gsm needs to turn on again */
  265.                 if (GetTotalCallCount() == 0)
  266.                 {
  267.                     PRINT_INFORMATION(("n[mmi_voip_switch_session] SCO Endsn"));
  268.                     mmi_profiles_bt_call_end_callback();
  269.                 }
  270.                 else
  271.                 {
  272.                     PRINT_INFORMATION(("n[mmi_voip_switch_session] SCO Not Endsn"));
  273.                 }
  274.             #endif
  275.                 mdi_audio_resume_background_play(); /* resume audio */
  276.                 AlmEnableExpiryHandler();
  277.             }
  278.             break;
  279.         default:
  280.             MMI_ASSERT(0);
  281.             break;
  282.     }
  283. }
  286. /*****************************************************************************
  287.  * FUNCTION
  288.  *  mmi_voip_create_rtp
  289.  * DESCRIPTION
  290.  *  Create RTP.
  291.  * PARAMETERS
  292.  *  sdp     [IN]        Sdp information passed by voip cc
  293.  * RETURNS
  294.  *  rtp handle that is given by voip media
  295.  *****************************************************************************/
  296. S32 mmi_voip_create_rtp(voip_sdp_struct *sdp)
  297. {
  298.     /*----------------------------------------------------------------*/
  299.     /* Local Variables                                                */
  300.     /*----------------------------------------------------------------*/
  301.     S32 rtpHandle = 0, profIndex = 0, i = 0;
  302.     U16 sdpCodec = 0;
  304.     /*----------------------------------------------------------------*/
  305.     /* Code Body                                                      */
  306.     /*----------------------------------------------------------------*/
  307.     profIndex = g_voip_cntx_p->prof_setting_info.actprofIndex;
  309.     if (sdp->num_codec == 0)
  310.     {
  311.         return -1;
  312.     }
  314.     rtpHandle = rtp_create(
  315.                     g_voip_cntx_p->prof_setting_info.saved_prof[profIndex].comm_info.dataAcct,
  316.                     sdp->local_addr,
  317.                     sdp->remote_addr,
  318.                     sdp->remote_rtp_port,
  319.                     sdp->remote_rtcp_port,
  320.                     sdp->local_rtp_port,
  321.                     sdp->local_rtcp_port,
  322.                     SOC_QOS_VOICE);
  323.     if (rtpHandle < 0)
  324.     {
  325.         PRINT_INFORMATION(("n[mmi_voip_create_rtp] Create RTP Fail, RTP Handle: %dn", rtpHandle));
  326.         return -1;
  327.     }
  329.     rtp_set_payload_type(rtpHandle, sdp->local_payload_type[0]);
  330.     if (sdp->sdp_status & VOIP_SDP_STATUS_IS_DTMF)
  331.     {
  332.         rtp_set_dtmf_payload_type(rtpHandle, sdp->dtmf_payload_type);
  333.     }
  334.     voip_set_ptime(rtpHandle, sdp->ptime);
  335.     voip_set_maxptime(rtpHandle, sdp->maxptime);
  337.     sdpCodec = sdp->codec[0];
  338.     /* g7231 annexa */
  339.     if ((sdp->codec[0] == VOIP_CODEC_G7231) && (sdp->g7231_annexa != VOIP_ANNEX_NONE))
  340.     {
  341.         switch (sdp->g7231_annexa)
  342.         {
  343.             case VOIP_ANNEX_NO:
  344.                 sdpCodec |= VOIP_CODEC_G7231_ANNEXA_PRESENT;
  345.                 break;
  346.             case VOIP_ANNEX_YES:
  347.                 sdpCodec |= VOIP_CODEC_G7231_ANNEXA_PRESENT;
  348.                 sdpCodec |= VOIP_CODEC_G7231_ANNEXA;
  349.                 break;
  350.             default:
  351.                 MMI_ASSERT(0); /* g7231_annexa should either VOIP_ANNEX_NO or VOIP_ANNEX_YES */
  352.         }
  353.     }
  355.     /* g729 annexb */
  356.     if ((sdp->codec[0] == VOIP_CODEC_G729) && (sdp->g729_annexb != VOIP_ANNEX_NONE))
  357.     {
  358.         switch (sdp->g729_annexb)
  359.         {
  360.             case VOIP_ANNEX_NO:
  361.                 sdpCodec |= VOIP_CODEC_G729_ANNEXB_PRESENT;
  362.                 break;
  363.             case VOIP_ANNEX_YES:
  364.                 sdpCodec |= VOIP_CODEC_G729_ANNEXB_PRESENT;
  365.                 sdpCodec |= VOIP_CODEC_G729_ANNEXB;
  366.                 break;
  367.             default:
  368.                 MMI_ASSERT(0); /* g729_annexb should either VOIP_ANNEX_NO or VOIP_ANNEX_YES */
  369.         }
  370.     }
  371.     
  372.     /* comfort noise */
  373.     if (g_voip_cntx_p->call_setting_info.saved_setting.comfortNoise == 0)
  374.     {
  375.         for (i = 0; i < VOIP_MAX_NUM_CODEC; i++)
  376.         {
  377.             if (sdp->codec[i] == VOIP_CODEC_CN)
  378.             {
  379.                 sdpCodec |= VOIP_CODEC_CN;
  380.                 break;
  381.             }
  382.         }
  383.     }
  384.     rtp_voip_init(rtpHandle, sdpCodec, sdp->modeset[0]);
  385.     PRINT_INFORMATION(("n[mmi_voip_create_rtp] Use to Lock Resource...Start"));
  386.     mdi_audio_speech_encode_start((S8)rtpHandle);
  387.     mdi_audio_speech_encode_stop((S8)rtpHandle);
  388.     mdi_audio_speech_decode_start((S8)rtpHandle);
  389.     mdi_audio_speech_decode_stop((S8)rtpHandle);
  390.     PRINT_INFORMATION(("n[mmi_voip_create_rtp] Use to Lock Resource...End"));
  391.     return rtpHandle;
  392. }
  395. /*****************************************************************************
  396.  * FUNCTION
  397.  *  mmi_voip_close_rtp
  398.  * DESCRIPTION
  399.  *  Close RTP.
  400.  * PARAMETERS
  401.  *  rtpHandle       [IN]        Intended to close rtp handle
  402.  * RETURNS
  403.  *  void
  404.  *****************************************************************************/
  405. void mmi_voip_close_rtp(S32 rtpHandle)
  406. {
  407.     /*----------------------------------------------------------------*/
  408.     /* Local Variables                                                */
  409.     /*----------------------------------------------------------------*/
  411.     /*----------------------------------------------------------------*/
  412.     /* Code Body                                                      */
  413.     /*----------------------------------------------------------------*/
  414.     switch (voip_curr_voip_state((S8) rtpHandle))
  415.     {
  416.         case VOIP_ENCODE:
  417.             mdi_audio_speech_encode_stop((S8) rtpHandle);
  418.             break;
  419.         case VOIP_DECODE:
  420.             mdi_audio_speech_decode_stop((S8) rtpHandle);
  421.             break;
  422.         case VOIP_ENCODE_DECODE:
  423.             mdi_audio_speech_encode_stop((S8) rtpHandle);
  424.             mdi_audio_speech_decode_stop((S8) rtpHandle);
  425.             break;
  426.         default:
  427.             break;
  428.     }
  429.     rtp_close((S8) rtpHandle);
  430. }
  433. /*****************************************************************************
  434.  * FUNCTION
  435.  *  mmi_voip_switch_rtp
  436.  * DESCRIPTION
  437.  *  Switch RTP based on the current RTP direction in the call point of view.
  438.  * PARAMETERS
  439.  *  isSuspend       [IN]        Suspend all audio paths before action
  440.  *  callId          [IN]        Call id
  441.  *  dialogId        [IN]        Dialog id
  442.  * RETURNS
  443.  *  void
  444.  *****************************************************************************/
  445. void mmi_voip_switch_rtp(BOOL isSuspend, S32 callId, S32 dialogId)
  446. {
  447.     /*----------------------------------------------------------------*/
  448.     /* Local Variables                                                */
  449.     /*----------------------------------------------------------------*/
  450.     S32 callIndex = 0, dialogIndex = 0;
  451.     S32 i = 0, j = 0, rtpHandle = 0, rtpDirection = 0, numDialog = 0;
  452.     BOOL isMixer = FALSE;
  454.     /*----------------------------------------------------------------*/
  455.     /* Code Body                                                      */
  456.     /*----------------------------------------------------------------*/
  457.     if (callId != -1)
  458.     {
  459.         callIndex = mmi_voip_get_call_index(callId);
  460.         if (dialogId != -1)
  461.         {
  462.             dialogIndex = mmi_voip_get_dialog_index(callIndex, dialogId);
  463.             rtpHandle = g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[dialogIndex].rtpHandle;
  464.             if (rtpHandle != -1)
  465.             {
  466.                 numDialog = g_voip_cntx_p->call_list_info.call_info[callIndex].numDialog;
  467.                 rtpDirection =
  468.                     (isSuspend == TRUE) ? (VOIP_RTP_DIRECTION_INACTIVE) : (g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[dialogIndex].sdp_info.direction);
  469.                 if (numDialog == VOIP_MAX_NUM_DIALOG)
  470.                 {
  471.                     isMixer = g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[dialogIndex].isMixer;
  472.                     if ((rtpDirection == VOIP_RTP_DIRECTION_SENDRECV) && (isMixer == FALSE))
  473.                     {
  474.                         mmi_voip_control_rtp(rtpHandle, rtpDirection);
  475.                         mdi_audio_speech_mixer_add((S8) rtpHandle);
  476.                         g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[dialogIndex].isMixer = TRUE;
  477.                     }
  478.                     else if ((rtpDirection != VOIP_RTP_DIRECTION_SENDRECV) && (isMixer == TRUE))
  479.                     {
  480.                         mdi_audio_speech_mixer_remove((S8) rtpHandle);
  481.                         g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[dialogIndex].isMixer = FALSE;
  482.                         mmi_voip_control_rtp(rtpHandle, rtpDirection);
  483.                     }
  484.                     else
  485.                     {
  486.                         mmi_voip_control_rtp(rtpHandle, rtpDirection);  /* remain the same */
  487.                     }
  488.                 }
  489.                 else    /* not conference call */
  490.                 {
  491.                     g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[dialogIndex].isMixer = FALSE;
  492.                     mmi_voip_control_rtp(rtpHandle, rtpDirection);
  493.                 }
  494.             }
  495.         }
  496.         else    /* dialogId == -1 */
  497.         {
  498.             for (i = 0; i < g_voip_cntx_p->call_list_info.call_info[callIndex].numDialog; i++)
  499.             {
  500.                 rtpHandle = g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].rtpHandle;
  501.                 if (rtpHandle != -1)
  502.                 {
  503.                     numDialog = g_voip_cntx_p->call_list_info.call_info[callIndex].numDialog;
  504.                     rtpDirection =
  505.                         (isSuspend == TRUE) ? (VOIP_RTP_DIRECTION_INACTIVE) : (g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].sdp_info.direction);
  506.                     if (numDialog == VOIP_MAX_NUM_DIALOG)
  507.                     {
  508.                         isMixer = g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].isMixer;
  509.                         if ((rtpDirection == VOIP_RTP_DIRECTION_SENDRECV) && (isMixer == FALSE))
  510.                         {
  511.                             mmi_voip_control_rtp(rtpHandle, rtpDirection);
  512.                             mdi_audio_speech_mixer_add((S8) rtpHandle);
  513.                             g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].isMixer = TRUE;
  514.                         }
  515.                         else if ((rtpDirection != VOIP_RTP_DIRECTION_SENDRECV) && (isMixer == TRUE))
  516.                         {
  517.                             mdi_audio_speech_mixer_remove((S8) rtpHandle);
  518.                             g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].isMixer = FALSE;
  519.                             mmi_voip_control_rtp(rtpHandle, rtpDirection);
  520.                         }
  521.                         else
  522.                         {
  523.                             mmi_voip_control_rtp(rtpHandle, rtpDirection);      /* remain the same */
  524.                         }
  525.                     }
  526.                     else    /* not conference call */
  527.                     {
  528.                         g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].isMixer = FALSE;
  529.                         mmi_voip_control_rtp(rtpHandle, rtpDirection);
  530.                     }
  531.                 }
  532.             }
  533.         }
  534.     }
  535.     else    /* callId == -1 */
  536.     {
  537.         for (i = 0; i < g_voip_cntx_p->call_list_info.numTotal; i++)
  538.         {
  539.             for (j = 0; j < g_voip_cntx_p->call_list_info.call_info[i].numDialog; j++)
  540.             {
  541.                 rtpHandle = g_voip_cntx_p->call_list_info.call_info[i].dialog_info[j].rtpHandle;
  542.                 if (rtpHandle != -1)
  543.                 {
  544.                     numDialog = g_voip_cntx_p->call_list_info.call_info[i].numDialog;
  545.                     rtpDirection =
  546.                         (isSuspend == TRUE) ? (VOIP_RTP_DIRECTION_INACTIVE) : (g_voip_cntx_p->call_list_info.call_info[i].dialog_info[j].sdp_info.direction);
  547.                     if (numDialog == VOIP_MAX_NUM_DIALOG)
  548.                     {
  549.                         isMixer = g_voip_cntx_p->call_list_info.call_info[i].dialog_info[j].isMixer;
  550.                         if ((rtpDirection == VOIP_RTP_DIRECTION_SENDRECV) && (isMixer == FALSE))
  551.                         {
  552.                             mmi_voip_control_rtp(rtpHandle, rtpDirection);
  553.                             mdi_audio_speech_mixer_add((S8) rtpHandle);
  554.                             g_voip_cntx_p->call_list_info.call_info[i].dialog_info[j].isMixer = TRUE;
  555.                         }
  556.                         else if ((rtpDirection != VOIP_RTP_DIRECTION_SENDRECV) && (isMixer == TRUE))
  557.                         {
  558.                             mdi_audio_speech_mixer_remove((S8) rtpHandle);
  559.                             g_voip_cntx_p->call_list_info.call_info[i].dialog_info[j].isMixer = FALSE;
  560.                             mmi_voip_control_rtp(rtpHandle, rtpDirection);
  561.                         }
  562.                         else
  563.                         {
  564.                             mmi_voip_control_rtp(rtpHandle, rtpDirection);      /* remain the same */
  565.                         }
  566.                     }
  567.                     else    /* not conference call */
  568.                     {
  569.                         g_voip_cntx_p->call_list_info.call_info[i].dialog_info[j].isMixer = FALSE;
  570.                         mmi_voip_control_rtp(rtpHandle, rtpDirection);
  571.                     }
  572.                 }
  573.             }
  574.         }
  575.     }
  576. }
  579. /*****************************************************************************
  580.  * FUNCTION
  581.  *  mmi_voip_control_rtp
  582.  * DESCRIPTION
  583.  *  Switch RTP based on the current RTP direction in the RTP point of view.
  584.  * PARAMETERS
  585.  *  rtpHandle           [IN]        Rtp handle
  586.  *  rtpDirection        [IN]        Rtp direction
  587.  * RETURNS
  588.  *  void
  589.  *****************************************************************************/
  590. void mmi_voip_control_rtp(S32 rtpHandle, S32 rtpDirection)
  591. {
  592.     /*----------------------------------------------------------------*/
  593.     /* Local Variables                                                */
  594.     /*----------------------------------------------------------------*/
  596.     /*----------------------------------------------------------------*/
  597.     /* Code Body                                                      */
  598.     /*----------------------------------------------------------------*/
  599.     switch (rtpDirection)
  600.     {
  601.         case VOIP_RTP_DIRECTION_SENDRECV:                   /*  send and receive */
  602.             switch (voip_curr_voip_state((S8) rtpHandle))   /* ensure not to set encoding / decoding twice */
  603.             {
  604.                 case VOIP_IDLE:
  605.                     mdi_audio_speech_encode_start((S8) rtpHandle);
  606.                     mdi_audio_speech_decode_start((S8) rtpHandle);
  607.                     break;
  608.                 case VOIP_ENCODE:
  609.                     mdi_audio_speech_decode_start((S8) rtpHandle);
  610.                     break;
  611.                 case VOIP_DECODE:
  612.                     mdi_audio_speech_encode_start((S8) rtpHandle);
  613.                     break;
  614.                 case VOIP_ENCODE_DECODE:
  615.                     break;
  616.                 default:
  617.                     MMI_ASSERT(0);
  618.                     break;
  619.             }
  620.             break;
  621.         #if 0   /* implement send only as inactive */
  622. /* under construction !*/
  623. /* under construction !*/
  624. /* under construction !*/
  625. /* under construction !*/
  626. /* under construction !*/
  627. /* under construction !*/
  628. /* under construction !*/
  629. /* under construction !*/
  630. /* under construction !*/
  631. /* under construction !*/
  632. /* under construction !*/
  633. /* under construction !*/
  634. /* under construction !*/
  635. /* under construction !*/
  636. /* under construction !*/
  637. /* under construction !*/
  638. /* under construction !*/
  639. /* under construction !*/
  640. /* under construction !*/
  641. /* under construction !*/
  642.         #endif /* 0 */ 
  643.         case VOIP_RTP_DIRECTION_RECVONLY:                   /*  receive only */
  644.             switch (voip_curr_voip_state((S8) rtpHandle))   /* ensure not to set encoding / decoding twice */
  645.             {
  646.                 case VOIP_IDLE:
  647.                     mdi_audio_speech_decode_start((S8) rtpHandle);
  648.                     break;
  649.                 case VOIP_ENCODE:
  650.                     mdi_audio_speech_encode_stop((S8) rtpHandle);
  651.                     mdi_audio_speech_decode_start((S8) rtpHandle);
  652.                     break;
  653.                 case VOIP_DECODE:
  654.                     break;
  655.                 case VOIP_ENCODE_DECODE:
  656.                     mdi_audio_speech_encode_stop((S8) rtpHandle);
  657.                     break;
  658.                 default:
  659.                     MMI_ASSERT(0);
  660.                     break;
  661.             }
  662.             break;
  663.         case VOIP_RTP_DIRECTION_SENDONLY:                   /*  send only */
  664.         case VOIP_RTP_DIRECTION_INACTIVE:                   /*  rtp session suspends */
  665.             switch (voip_curr_voip_state((S8) rtpHandle))   /* ensure not to set encoding / decoding twice */
  666.             {
  667.                 case VOIP_IDLE:
  668.                     break;
  669.                 case VOIP_ENCODE:
  670.                     mdi_audio_speech_encode_stop((S8) rtpHandle);
  671.                     break;
  672.                 case VOIP_DECODE:
  673.                     mdi_audio_speech_decode_stop((S8) rtpHandle);
  674.                     break;
  675.                 case VOIP_ENCODE_DECODE:
  676.                     mdi_audio_speech_encode_stop((S8) rtpHandle);
  677.                     mdi_audio_speech_decode_stop((S8) rtpHandle);
  678.                     break;
  679.                 default:
  680.                     MMI_ASSERT(0);
  681.                     break;
  682.             }
  683.             break;
  684.         default:
  685.             MMI_ASSERT(0);
  686.     }
  687. }
  690. /*****************************************************************************
  691.  * FUNCTION
  692.  *  mmi_voip_remove_mixer_before_close
  693.  * DESCRIPTION
  694.  *  Remove mixer if any before closeing RTP.
  695.  * PARAMETERS
  696.  *  callIndex       [IN]        Call index
  697.  * RETURNS
  698.  *  void
  699.  *****************************************************************************/
  700. void mmi_voip_remove_mixer_before_close(S32 callIndex)
  701. {
  702.     /*----------------------------------------------------------------*/
  703.     /* Local Variables                                                */
  704.     /*----------------------------------------------------------------*/
  705.     S32 i = 0;
  707.     /*----------------------------------------------------------------*/
  708.     /* Code Body                                                      */
  709.     /*----------------------------------------------------------------*/
  710.     for (i = 0; i < VOIP_MAX_NUM_DIALOG; i++)
  711.     {
  712.         if (g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].isMixer == TRUE)
  713.         {
  714.             mdi_audio_speech_mixer_remove((S8) g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].rtpHandle);
  715.             g_voip_cntx_p->call_list_info.call_info[callIndex].dialog_info[i].isMixer = FALSE;
  716.         }
  717.     }
  718. }
  721. /*****************************************************************************
  722.  * FUNCTION
  723.  *  mmi_voip_send_dtmf_start
  724.  * DESCRIPTION
  725.  *  Send DTMF sound to remote side.
  726.  * PARAMETERS
  727.  *  void
  728.  * RETURNS
  729.  *  void
  730.  *****************************************************************************/
  731. void mmi_voip_send_dtmf_start(void)
  732. {
  733.     /*----------------------------------------------------------------*/
  734.     /* Local Variables                                                */
  735.     /*----------------------------------------------------------------*/
  736.     S32 i = 0, currhiliteTab = 0, numDialog = 0, rtpHandle[VOIP_MAX_NUM_DIALOG];
  737.     U16 keyCode = 0, keyType = 0;
  738.     BOOL outbandDtmf = FALSE;
  740.     /*----------------------------------------------------------------*/
  741.     /* Code Body                                                      */
  742.     /*----------------------------------------------------------------*/
  743.     currhiliteTab = g_voip_cntx_p->call_misc_info.currhiliteTab;
  744.     numDialog = g_voip_cntx_p->call_list_info.call_info[currhiliteTab].numDialog;
  746.     for (i = 0; i < numDialog; i++)
  747.     {
  748.         /* only send dtmf to the dialog that encoding is started */
  749.         if (g_voip_cntx_p->call_list_info.call_info[currhiliteTab].dialog_info[i].sdp_info.direction == VOIP_RTP_DIRECTION_SENDRECV)
  750.         {
  751.             rtpHandle[i] = g_voip_cntx_p->call_list_info.call_info[currhiliteTab].dialog_info[i].rtpHandle;
  752.             outbandDtmf = (g_voip_cntx_p->call_list_info.call_info[currhiliteTab].dialog_info[i].sdp_info.sdp_status & VOIP_SDP_STATUS_IS_DTMF);
  753.         }
  754.         else
  755.         {
  756.             rtpHandle[i] = -1;
  757.         }
  758.     }
  759.     if (numDialog == VOIP_MAX_NUM_DIALOG)
  760.     {
  761.         /* always send inband dtmf for conference call in case one dialog is inband and the other is outband */
  762.         outbandDtmf = FALSE;
  763.     }    
  765.     PRINT_INFORMATION(("n[mmi_voip_send_dtmf_start] Local DTMF: %d, Remote DTMF: %dn", g_voip_cntx_p->call_setting_info.saved_setting.dtmf, outbandDtmf));
  766.     GetkeyInfo(&keyCode, &keyType);
  767.     if (mmi_voip_validate_dtmf(keyCode))
  768.     {
  769.         if (g_voip_cntx_p->call_setting_info.saved_setting.dtmf == VOIP_DTMF_TYPE_NONE)
  770.         {
  771.             /* do nothing */
  772.         }
  773.         else if ((g_voip_cntx_p->call_setting_info.saved_setting.dtmf == VOIP_DTMF_IN_BAND) || 
  774.                  (outbandDtmf == FALSE)) /* remote doesn't support outband dtmf */
  775.         {
  776.             if (g_voip_cntx_p->call_misc_info.isMute == FALSE)
  777.             {
  778.                 MuteOnMicrophone(); /* temporarily turn off microphone */
  779.             }
  780.             for (i = 0; i < numDialog; i++)
  781.             {
  782.                 if (rtpHandle[i] != -1)
  783.                 {
  784.                     mdi_audio_speech_dtmf_start(
  785.                         rtpHandle[i],
  786.                         mmi_voip_get_dtmf_keycode_enum(&keyCode),
  787.                         RTP_DTMF_TYPE_INBAND);
  788.                 }
  789.             }
  790.         }
  791.         /* send outband dtmf only if both local and remote support outband dtmf */
  792.         else if ((g_voip_cntx_p->call_setting_info.saved_setting.dtmf == VOIP_DTMF_OUT_OF_BAND) && 
  793.                  (outbandDtmf == TRUE))
  794.         {
  795.             if (g_voip_cntx_p->call_misc_info.isMute == FALSE)
  796.             {
  797.                 MuteOnMicrophone(); /* temporarily turn off microphone */
  798.             }
  799.             for (i = 0; i < numDialog; i++)
  800.             {
  801.                 if (rtpHandle[i] != -1)
  802.                 {
  803.                     mdi_audio_speech_dtmf_start(
  804.                         rtpHandle[i],
  805.                         mmi_voip_get_dtmf_keycode_enum(&keyCode),
  806.                         RTP_DTMF_TYPE_RFC2833);
  807.                 }
  808.             }
  809.         }
  810.     }
  811.     wgui_execute_key_handler(keyCode, keyType);
  812. }
  815. /*****************************************************************************
  816.  * FUNCTION
  817.  *  mmi_voip_send_dtmf_stop
  818.  * DESCRIPTION
  819.  *  Stop DTMF sound to remote side.
  820.  * PARAMETERS
  821.  *  void
  822.  * RETURNS
  823.  *  void
  824.  *****************************************************************************/
  825. void mmi_voip_send_dtmf_stop(void)
  826. {
  827.     /*----------------------------------------------------------------*/
  828.     /* Local Variables                                                */
  829.     /*----------------------------------------------------------------*/
  830.     S32 i = 0, currhiliteTab = 0, numDialog = 0, rtpHandle[VOIP_MAX_NUM_DIALOG];
  832.     /*----------------------------------------------------------------*/
  833.     /* Code Body                                                      */
  834.     /*----------------------------------------------------------------*/
  835.     currhiliteTab = g_voip_cntx_p->call_misc_info.currhiliteTab;
  836.     numDialog = g_voip_cntx_p->call_list_info.call_info[currhiliteTab].numDialog;
  838.     for (i = 0; i < numDialog; i++)
  839.     {
  840.         /* only send dtmf to the dialog that encoding is started */
  841.         if (g_voip_cntx_p->call_list_info.call_info[currhiliteTab].dialog_info[i].sdp_info.direction == VOIP_RTP_DIRECTION_SENDRECV)
  842.         {
  843.             rtpHandle[i] = g_voip_cntx_p->call_list_info.call_info[currhiliteTab].dialog_info[i].rtpHandle;
  844.         }
  845.         else
  846.         {
  847.             rtpHandle[i] = -1;
  848.         }
  849.     } 
  851.     if (g_voip_cntx_p->call_setting_info.saved_setting.dtmf != VOIP_DTMF_TYPE_NONE)
  852.     {
  853.         if (g_voip_cntx_p->call_misc_info.isMute == FALSE)
  854.         {
  855.             MuteOffMicrophone();
  856.         }
  857.         for (i = 0; i < numDialog; i++)
  858.         {
  859.             if (rtpHandle[i] != -1)
  860.             {
  861.                 mdi_audio_speech_dtmf_stop(rtpHandle[i]);
  862.             }
  863.         }
  864.     }
  865.     else    /* g_voip_cntx_p->call_setting_info.saved_setting.dtmf == VOIP_DTMF_TYPE_NONE */
  866.     {
  867.         /* do nothing */
  868.     }
  869. }
  870. #endif /* __MMI_VOIP__ */