audio/msm7630: Fix for 16kHz VoIP call failure issueics-release

-VoIP call with 16kHz is not working
-VoIP call failure because of directoutput thread is created for
 VoIP calls 8kHz and 16kHz sample rates, but call failed because
 target does not have 16kHz PCM support.
-Inherited needsDirectOutput() and added a condition.
 so that direct out put thread is created only for 8kHz.
 Modified MVS frame structure to use 8x55 modem compatible
 MVS buffer size of 320 bytes.

CRs-Fixed: 362068
Change-Id: Id845e9adf0380e66f0b260fa7cd6d1f6227166da

3 files changed, 27 insertions, 2 deletions

diff --git a/audio/msm7x30/AudioHardware.cpp b/audio/msm7x30/AudioHardware.cpp
index 29c70316..4ee712d7 100644
--- a/audio/msm7x30/AudioHardware.cpp
+++ b/audio/msm7x30/AudioHardware.cpp
@@ -944,7 +944,7 @@ AudioStreamIn* AudioHardware::openInputStream(
 
 #ifdef WITH_QCOM_VOIP_OVER_MVS
     AudioStreamIn *in;
-    if(devices == AudioSystem::DEVICE_IN_COMMUNICATION) {
+    if((devices == AudioSystem::DEVICE_IN_COMMUNICATION)&& (*sampleRate == 8000)) {
         LOGE("Create Audio stream Voip \n");
         AudioStreamInVoip* inVoip = new AudioStreamInVoip();
         status_t lStatus = NO_ERROR;
@@ -3237,7 +3237,7 @@ ssize_t AudioHardware::AudioStreamOutDirect::write(const void* buffer, size_t by
             mStandby = false;
         }
     }
-    struct msm_audio_mvs_frame audio_mvs_frame;
+    struct q5v2_msm_audio_mvs_frame audio_mvs_frame;
     audio_mvs_frame.frame_type = 0;
     while (count) {
         audio_mvs_frame.len = mBufferSize;
diff --git a/audio/msm7x30/AudioPolicyManager.cpp b/audio/msm7x30/AudioPolicyManager.cpp
index 6a6c0e1e..aed44e78 100644
--- a/audio/msm7x30/AudioPolicyManager.cpp
+++ b/audio/msm7x30/AudioPolicyManager.cpp
@@ -780,7 +780,24 @@ status_t AudioPolicyManager::stopInput(audio_io_handle_t input)
     return NO_ERROR;
 }
 
+#ifdef WITH_QCOM_VOIP_OVER_MVS
+bool AudioPolicyManager::needsDirectOuput(AudioSystem::stream_type stream,
+                                          uint32_t samplingRate,
+                                          uint32_t format,
+                                          uint32_t channels,
+                                          AudioSystem::output_flags flags,
+                                          uint32_t device)
+{
+   LOGV("AudioPolicyManager::needsDirectOuput stream = %d mPhoneState = %d,"
+        "channels=%d,samplingRate=%d",stream, mPhoneState,channels,samplingRate);
+
+   return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+          (format !=0 && !AudioSystem::isLinearPCM(format)) ||
+          ((stream == AudioSystem::VOICE_CALL) && (channels == AudioSystem::CHANNEL_OUT_MONO)
+          && (samplingRate == 8000 )));
 
+}
+#endif
 // ----------------------------------------------------------------------------
 // AudioPolicyManager
 // ----------------------------------------------------------------------------
diff --git a/audio/msm7x30/AudioPolicyManager.h b/audio/msm7x30/AudioPolicyManager.h
index 0161e2c3..443b0ce7 100644
--- a/audio/msm7x30/AudioPolicyManager.h
+++ b/audio/msm7x30/AudioPolicyManager.h
@@ -73,6 +73,14 @@ public:
                                     int session = 0);
         virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
         status_t startInput(audio_io_handle_t input);
+#ifdef WITH_QCOM_VOIP_OVER_MVS
+        virtual bool needsDirectOuput(AudioSystem::stream_type stream,
+                                      uint32_t samplingRate,
+                                      uint32_t format,
+                                      uint32_t channels,
+                                      AudioSystem::output_flags flags,
+                                      uint32_t device);
+#endif
 
 protected:
         // true is current platform implements a back microphone