Snap for 8426163 from 7f5ca1faeccbd81eda47e36261d1fed1577ce327 to mainline-tzdata2-releaseandroid-mainline-12.0.0_r112aml_tz2_305400500aml_tz2_305400300aml_tz2_305400100aml_tz2_304500300aml_tz2_303900110aml_tz2_303900102aml_tz2_303800002aml_tz2_303800001aml_tz2_303200001android12-mainline-tzdata2-releaseaml_tz2_305400100

Change-Id: I29c1230bdf2c32fac2da050918fa57c9f902735f

20 files changed, 85 insertions, 714 deletions

diff --git a/Android.bp b/Android.bp
index 8cc4146..0f8f3a4 100644
--- a/Android.bp
+++ b/Android.bp
@@ -1,18 +1 @@
-package {
-    default_applicable_licenses: ["external_sonivox_license"],
-}
-
-// Added automatically by a large-scale-change
-// See: http://go/android-license-faq
-license {
-    name: "external_sonivox_license",
-    visibility: [":__subpackages__"],
-    license_kinds: [
-        "SPDX-license-identifier-Apache-2.0",
-    ],
-    license_text: [
-        "NOTICE",
-    ],
-}
-
 subdirs = ["arm-wt-22k"]
diff --git a/METADATA b/METADATA
deleted file mode 100644
index d97975c..0000000
--- a/METADATA
+++ /dev/null
@@ -1,3 +0,0 @@
-third_party {
-  license_type: NOTICE
-}
diff --git a/OWNERS b/OWNERS
index 8333c2f..5c49fbc 100644
--- a/OWNERS
+++ b/OWNERS
@@ -1,3 +1,4 @@
-# owners for external/sonivox
-include platform/frameworks/av:/media/janitors/codec_OWNERS
-essick@google.com
+# Default code reviewers picked from top 3 or more developers.
+# Please update this list if you find better candidates.
+marcone@google.com
+wjia@google.com
diff --git a/arm-wt-22k/Android.bp b/arm-wt-22k/Android.bp
index c05443b..bf45b0d 100644
--- a/arm-wt-22k/Android.bp
+++ b/arm-wt-22k/Android.bp
@@ -1,20 +1,3 @@
-package {
-    default_applicable_licenses: ["external_sonivox_arm-wt-22k_license"],
-}
-
-// Added automatically by a large-scale-change
-// See: http://go/android-license-faq
-license {
-    name: "external_sonivox_arm-wt-22k_license",
-    visibility: [":__subpackages__"],
-    license_kinds: [
-        "SPDX-license-identifier-Apache-2.0",
-    ],
-    license_text: [
-        "NOTICE",
-    ],
-}
-
 cc_defaults {
     name: "libsonivox-defaults",
     srcs: [
@@ -87,14 +70,6 @@ cc_defaults {
         "liblog",
     ],
 
-    host_supported: true,
-
-    target: {
-        darwin: {
-            enabled: false,
-        },
-    },
-
     arch: {
         arm: {
             instruction_set: "arm",
@@ -111,15 +86,22 @@ cc_defaults {
                 // In order to use #include instead of .include
                 "-xassembler-with-cpp",
 
-                "-DSAMPLE_RATE_22050=1",
-                "-DSTEREO_OUTPUT=1",
-                "-DFILTER_ENABLED=1",
-                "-DSAMPLES_16_BIT=1",
+                "-Wa,--defsym,SAMPLE_RATE_22050=1",
+                "-Wa,--defsym,STEREO_OUTPUT=1",
+                "-Wa,--defsym,FILTER_ENABLED=1",
+                "-Wa,--defsym,SAMPLES_16_BIT=1",
             ],
 
             cflags: [
                 "-DNATIVE_EAS_KERNEL",
             ],
+
+            // .s files not ported for Clang assembler yet.
+            clang_asflags: ["-no-integrated-as"],
+        },
+        arm64: {
+            // .s files not ported for Clang assembler yet.
+            clang_asflags: ["-no-integrated-as"],
         },
     },
     sanitize: {
@@ -153,3 +135,5 @@ cc_library {
         "-DJET_INTERFACE",
     ],
 }
+
+
diff --git a/arm-wt-22k/lib_src/ARM-E_filter_gnu.s b/arm-wt-22k/lib_src/ARM-E_filter_gnu.s
index c4ffd55..859d9a4 100644
--- a/arm-wt-22k/lib_src/ARM-E_filter_gnu.s
+++ b/arm-wt-22k/lib_src/ARM-E_filter_gnu.s
@@ -55,6 +55,7 @@ tmp2	.req	r10
 @RestoreRegs	RLIST	{r4-r10, pc}

 

 

+	.func	WT_VoiceFilter

 WT_VoiceFilter:

 

 	STMFD	sp!, {r4-r10, lr}

@@ -111,7 +112,7 @@ FilterLoop:
 

 	MOV		z1, tmp1, ASR #14				@ shift result to low word

 	

-	LDRSHGT	tmp0, [pBuffer, #NEXT_OUTPUT_PCM]	@ fetch next sample

+	LDRGTSH	tmp0, [pBuffer, #NEXT_OUTPUT_PCM]	@ fetch next sample

 

 	STRH	z1, [pBuffer], #NEXT_OUTPUT_PCM	@ write back to buffer

 

@@ -128,5 +129,6 @@ FilterLoop:
 	LDMFD	sp!,{r4-r10, lr}

 	BX		lr

 

+	.endfunc

 	.end

 

diff --git a/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s b/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s
index 59ab0fd..2529e93 100644
--- a/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s
+++ b/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s
@@ -56,6 +56,7 @@ phaseFracMask	.req	r11
 @SaveRegs	RLIST	{r4-r11,lr}

 @RestoreRegs	RLIST	{r4-r11,pc}

 

+	.func	WT_Interpolate

 WT_Interpolate:

 

 	STMFD	sp!,{r4-r11,lr}

@@ -80,15 +81,13 @@ InterpolationLoop:
 	SUBS	tmp0, pPhaseAccum, pLoopEnd		@ check for loop end

 	ADDGE	pPhaseAccum, pLoopStart, tmp0	@ loop back to start

 

-	#ifdef	SAMPLES_8_BIT

+	.ifdef	SAMPLES_8_BIT

 	LDRSB	tmp0, [pPhaseAccum]				@ tmp0 = x0

 	LDRSB	tmp1, [pPhaseAccum, #1]			@ tmp1 = x1

-	#elif	SAMPLES_16_BIT

+	.else

 	LDRSH	tmp0, [pPhaseAccum]				@ tmp0 = x0

 	LDRSH	tmp1, [pPhaseAccum, #2]			@ tmp1 = x1

-	#else

-	#error Must define one of SAMPLES_8_BIT or SAMPLES_16_BIT.

-	#endif

+	.endif

 

 	ADD		tmp2, phaseIncrement, phaseFrac	@ increment pointer here to avoid pipeline stall

 

@@ -102,13 +101,11 @@ InterpolationLoop:
 @ saturation operation should take in the filter before scaling back to

 @ 16 bits or the signal path should be increased to 18 bits or more.

 

-	#ifdef	SAMPLES_8_BIT

+	.ifdef	SAMPLES_8_BIT

 	MOV		tmp0, tmp0, LSL #6							@ boost 8-bit signal by 36dB

-	#elif	SAMPLES_16_BIT

+	.else

 	MOV		tmp0, tmp0, ASR #2							@ reduce 16-bit signal by 12dB

-	#else

-	#error Must define one of SAMPLES_8_BIT or SAMPLES_16_BIT.

-	#endif

+	.endif															

 	

 	ADD		tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6)	@ tmp1 = tmp0 + (tmp1 >> (15-6))

 															@	   = x0 + f * (x1 - x0) == interpolated result

@@ -129,5 +126,6 @@ InterpolationLoop:
 	LDMFD	sp!,{r4-r11,lr}

 	BX		lr

 

+	.endfunc

 	.end

 	

diff --git a/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s b/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s
index baa6f7a..55a0ba7 100644
--- a/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s
+++ b/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s
@@ -54,6 +54,7 @@ tmp2	.req	r9
 @SaveRegs	RLIST	{r4-r9,lr}

 @RestoreRegs	RLIST	{r4-r9,pc}

 

+	.func	WT_InterpolateNoLoop

 WT_InterpolateNoLoop:

 

 	STMFD	sp!, {r4-r9,lr}

@@ -72,15 +73,13 @@ WT_InterpolateNoLoop:
 

 InterpolationLoop:

 

-	#ifdef	SAMPLES_8_BIT

+	.ifdef	SAMPLES_8_BIT

 	LDRSB	tmp0, [pPhaseAccum]				@ tmp0 = x0

 	LDRSB	tmp1, [pPhaseAccum, #1]			@ tmp1 = x1

-	#elif	SAMPLES_16_BIT

+	.else

 	LDRSH	tmp0, [pPhaseAccum]				@ tmp0 = x0

 	LDRSH	tmp1, [pPhaseAccum, #2]			@ tmp1 = x1

-	#else

-	#error Must define one of SAMPLES_8_BIT or SAMPLES_16_BIT.

-	#endif

+	.endif

 

 	ADD		tmp2, phaseIncrement, phaseFrac	@ increment pointer here to avoid pipeline stall

 

@@ -94,13 +93,11 @@ InterpolationLoop:
 @ saturation operation should take in the filter before scaling back to

 @ 16 bits or the signal path should be increased to 18 bits or more.

 

-	#ifdef	SAMPLES_8_BIT

+	.ifdef	SAMPLES_8_BIT

 	MOV		tmp0, tmp0, LSL #6							@ boost 8-bit signal by 36dB

-	#elif	SAMPLES_16_BIT

+	.else

 	MOV		tmp0, tmp0, ASR #2							@ reduce 16-bit signal by 12dB

-	#else

-	#error Must define one of SAMPLES_8_BIT or SAMPLES_16_BIT.

-	#endif

+	.endif															

 	

 	ADD		tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6)	@ tmp1 = tmp0 + (tmp1 >> (15-6))

 															@	   = x0 + f * (x1 - x0) == interpolated result

@@ -128,5 +125,6 @@ InterpolationLoop:
 	LDMFD	sp!,{r4-r9,lr}

 	BX		lr

 

+	.endfunc

 	.end

 	

diff --git a/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s b/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s
index e53bb99..f443fbb 100644
--- a/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s
+++ b/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s
@@ -40,6 +40,7 @@
 	.arm

 	.text

 

+	.func	SynthMasterGain

 SynthMasterGain:

 

 	.global	SynthMasterGain	@ allow other files to use this function

@@ -102,5 +103,7 @@ loop:
 

 @*****************************************************************************

 

+	.endfunc		@ end of function/procedure

+

 	.end		@ end of assembly code

 

diff --git a/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s b/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s
index 9e1fcce..6ca28b2 100644
--- a/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s
+++ b/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s
@@ -49,21 +49,22 @@ tmp3	.req	r6
 

 numSamples	.req	r9

 

-	#if	STEREO_OUTPUT

+	.if	STEREO_OUTPUT

 gainIncLeft	.req	r7

 gainIncRight	.req	r8

 gainLeft	.req	r10

 gainRight	.req	r11

-	#else

+	.else

 gainIncrement	.req	r7

 gain	.req	r8

-	#endif

+	.endif

 

 

 @ register context for local variables

 @SaveRegs	RLIST	{r4-r11,lr}

 @RestoreRegs	RLIST	{r4-r11,pc}

 

+	.func	WT_VoiceGain

 WT_VoiceGain:

 

 	STMFD	sp!, {r4-r11,lr}

@@ -79,7 +80,7 @@ WT_VoiceGain:
 @ due to storage and computational dependencies.

 @----------------------------------------------------------------

 

-	#if	STEREO_OUTPUT

+	.if	STEREO_OUTPUT

 

 	LDR		tmp0, [pWTFrame, #m_prevGain]

 	LDR		tmp1, [pWTFrame, #m_gainTarget]

@@ -131,7 +132,7 @@ StereoGainLoop:
 @----------------------------------------------------------------

 @ Mono version

 @----------------------------------------------------------------

-	#else

+	.else

 

 	LDR		gain, [pWTFrame, #m_prevGain]

 	MOV		gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)

@@ -155,10 +156,11 @@ MonoGainLoop:
 	SUBS	numSamples, numSamples, #1

 	BGT		MonoGainLoop

 

-	#endif	@end Mono version

+	.endif	@end Mono version

 

 	LDMFD	sp!,{r4-r11,lr}

 	BX		lr

 	

+	.endfunc

 	.end

 

diff --git a/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc b/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc
index 213944e..c0f8df3 100644
--- a/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc
+++ b/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc
@@ -12,45 +12,45 @@
 @****************************************************************
 
 
-    #ifdef  SAMPLE_RATE_8000
+    .ifdef  SAMPLE_RATE_8000
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 5
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 32
-    #endif
+    .endif
 
-    #ifdef  SAMPLE_RATE_16000
+    .ifdef  SAMPLE_RATE_16000
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 6
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 64
-    #endif
+    .endif
 
-    #ifdef  SAMPLE_RATE_20000
+    .ifdef  SAMPLE_RATE_20000
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
-    #endif
+    .endif
 
-    #ifdef  SAMPLE_RATE_22050
+    .ifdef  SAMPLE_RATE_22050
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
-    #endif
+    .endif
 
-    #ifdef  SAMPLE_RATE_24000
+    .ifdef  SAMPLE_RATE_24000
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
-    #endif
+    .endif
 
-    #ifdef  SAMPLE_RATE_32000
+    .ifdef  SAMPLE_RATE_32000
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
-    #endif
+    .endif
 
-    #ifdef  SAMPLE_RATE_44100
+    .ifdef  SAMPLE_RATE_44100
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 8
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 256
-    #endif
+    .endif
 
-    #ifdef  SAMPLE_RATE_48000
+    .ifdef  SAMPLE_RATE_48000
     .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 8
     .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 256
-    #endif
+    .endif
 
 
 @ if the OUTPUT PCM sample is 16-bits, then when using indexed addressing,
@@ -64,13 +64,13 @@
     .equ    PHASE_FRAC_MASK, 0x7FFF
 
 @ shift for phase accumulator when fraction carries over
-    #ifdef  SAMPLES_8_BIT
+    .ifdef  SAMPLES_8_BIT
     .equ    NEXT_INPUT_PCM_SHIFT, 0
-    #endif
+    .endif
 
-    #ifdef  SAMPLES_16_BIT
+    .ifdef  SAMPLES_16_BIT
     .equ    NEXT_INPUT_PCM_SHIFT, 1
-    #endif
+    .endif
 
 @****************************************************************************
     .equ    NUM_MIXER_GUARD_BITS, 4
@@ -90,19 +90,19 @@
 @ handle a struct in a compatible fashion. Switching to old fashion EQU
 @
 
-    #if FILTER_ENABLED
+    .if FILTER_ENABLED
 @**************************************
 @ typedef struct s_filter_tag
     .equ    m_z1, 0
     .equ    m_z2, 2
-    #endif
+    .endif
 
 @**************************************
 @ typedef struct s_wt_frame_tag
     .equ    m_gainTarget, 0
     .equ    m_phaseIncrement, 4
 
-    #if FILTER_ENABLED
+    .if FILTER_ENABLED
     .equ    m_k, 8
     .equ    m_b1, 12
     .equ    m_b2, 16
@@ -110,12 +110,12 @@
     .equ    m_pMixBuffer, 24
     .equ    m_numSamples, 28
     .equ    m_prevGain, 32
-    #else
+    .else
     .equ    m_pAudioBuffer, 8
     .equ    m_pMixBuffer, 12
     .equ    m_numSamples, 16
     .equ    m_prevGain, 20
-    #endif
+    .endif
 
 
 @**************************************
@@ -125,10 +125,10 @@
     .equ    m_pPhaseAccum, 8    @ /* points to first sample at start of loop */
     .equ    m_phaseFrac, 12 @ /* points to first sample at start of loop */
 
-    #if STEREO_OUTPUT
+    .if STEREO_OUTPUT
     .equ    m_gainLeft, 16  @ /* current gain, left ch  */
     .equ    m_gainRight, 18 @ /* current gain, right ch */
-    #endif
+    .endif
 
 
 @****************************************************************************
diff --git a/arm-wt-22k/lib_src/eas_data.h b/arm-wt-22k/lib_src/eas_data.h
index 5fe52a9..4191678 100644
--- a/arm-wt-22k/lib_src/eas_data.h
+++ b/arm-wt-22k/lib_src/eas_data.h
@@ -31,8 +31,6 @@
 #ifndef _EAS_DATA_H
 #define _EAS_DATA_H
 
-#include <stdint.h>
-
 #include "eas_types.h"
 #include "eas_synthcfg.h"
 #include "eas.h"
diff --git a/arm-wt-22k/lib_src/eas_rtttl.c b/arm-wt-22k/lib_src/eas_rtttl.c
index 1419d6d..79d1be8 100644
--- a/arm-wt-22k/lib_src/eas_rtttl.c
+++ b/arm-wt-22k/lib_src/eas_rtttl.c
@@ -439,12 +439,6 @@ static EAS_RESULT RTTTL_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS
         /* dotted note */
         else if (c == '.')
         {
-            /* Number of ticks must not be greater than 32-bits */
-            if ((ticks >> 1) > (INT32_MAX - ticks))
-            {
-                return EAS_ERROR_FILE_FORMAT;
-            }
-
             /*lint -e{704} shift for performance */
             ticks += ticks >> 1;
         }
@@ -496,22 +490,12 @@ static EAS_RESULT RTTTL_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS
                 }
 
                 /* next event is at end of this note */
-                if ((ticks - pData->restTicks) > (INT32_MAX - pData->time))
-                {
-                    return EAS_ERROR_FILE_FORMAT;
-                }
                 pData->time += ticks - pData->restTicks;
             }
 
             /* rest */
             else
-            {
-                if (ticks > (INT32_MAX - pData->time))
-                {
-                    return EAS_ERROR_FILE_FORMAT;
-                }
                 pData->time += ticks;
-            }
 
             /* event found, return to caller */
             break;
diff --git a/arm-wt-22k/lib_src/eas_smf.c b/arm-wt-22k/lib_src/eas_smf.c
index 0e70f01..e13e1d8 100644
--- a/arm-wt-22k/lib_src/eas_smf.c
+++ b/arm-wt-22k/lib_src/eas_smf.c
@@ -808,10 +808,6 @@ static EAS_RESULT SMF_GetDeltaTime (EAS_HW_DATA_HANDLE hwInstData, S_SMF_STREAM
     if ((result = SMF_GetVarLenData(hwInstData, pSMFStream->fileHandle, &ticks)) != EAS_SUCCESS)
         return result;
 
-    /* number of ticks must not exceed 32-bits */
-    if (ticks > (UINT32_MAX - pSMFStream->ticks))
-        return EAS_ERROR_FILE_FORMAT;
-
     pSMFStream->ticks += ticks;
     return EAS_SUCCESS;
 }
diff --git a/arm-wt-22k/lib_src/eas_wtengine.c b/arm-wt-22k/lib_src/eas_wtengine.c
index b1ee749..950616e 100644
--- a/arm-wt-22k/lib_src/eas_wtengine.c
+++ b/arm-wt-22k/lib_src/eas_wtengine.c
@@ -202,7 +202,7 @@ void WT_Interpolate (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame)
     loopEnd = (const EAS_SAMPLE*) pWTVoice->loopEnd + 1;
     pSamples = (const EAS_SAMPLE*) pWTVoice->phaseAccum;
     /*lint -e{713} truncation is OK */
-    phaseFrac = pWTVoice->phaseFrac & PHASE_FRAC_MASK;
+    phaseFrac = pWTVoice->phaseFrac;
     phaseInc = pWTIntFrame->frame.phaseIncrement;
 
     /* fetch adjacent samples */
@@ -218,8 +218,6 @@ void WT_Interpolate (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame)
 
     while (numSamples--) {
 
-        EAS_I32 nextSamplePhaseInc;
-
         /* linear interpolation */
         acc0 = samp2 - samp1;
         acc0 = acc0 * phaseFrac;
@@ -233,19 +231,19 @@ void WT_Interpolate (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame)
         /* increment phase */
         phaseFrac += phaseInc;
         /*lint -e{704} <avoid divide>*/
-        nextSamplePhaseInc = phaseFrac >> NUM_PHASE_FRAC_BITS;
+        acc0 = phaseFrac >> NUM_PHASE_FRAC_BITS;
 
         /* next sample */
-        if (nextSamplePhaseInc > 0) {
+        if (acc0 > 0) {
+
             /* advance sample pointer */
-            pSamples +=  nextSamplePhaseInc;
-            phaseFrac = phaseFrac & PHASE_FRAC_MASK;
+            pSamples += acc0;
+            phaseFrac = (EAS_I32)((EAS_U32)phaseFrac & PHASE_FRAC_MASK);
 
-            /* decrementing pSamples by entire buffer length until second pSample is within */
-            /* loopEnd                                                                      */
-            while (&pSamples[1] >= loopEnd) {
-                pSamples -= (loopEnd - (const EAS_SAMPLE*)pWTVoice->loopStart);
-            }
+            /* check for loop end */
+            acc0 = (EAS_I32) (pSamples - loopEnd);
+            if (acc0 >= 0)
+                pSamples = (const EAS_SAMPLE*) pWTVoice->loopStart + acc0;
 
             /* fetch new samples */
 #if defined(_8_BIT_SAMPLES)
diff --git a/arm-wt-22k/lib_src/eas_wtsynth.c b/arm-wt-22k/lib_src/eas_wtsynth.c
index 74f78f5..d3ca3af 100644
--- a/arm-wt-22k/lib_src/eas_wtsynth.c
+++ b/arm-wt-22k/lib_src/eas_wtsynth.c
@@ -482,7 +482,7 @@ EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, E
 #endif
         /* now account for the fractional portion */
         /*lint -e{703} use shift for performance */
-        numSamples = (numSamples << NUM_PHASE_FRAC_BITS) - (EAS_I32) pWTVoice->phaseFrac;
+        numSamples = (EAS_I32) ((numSamples << NUM_PHASE_FRAC_BITS) - pWTVoice->phaseFrac);
         if (pWTIntFrame->frame.phaseIncrement) {
             pWTIntFrame->numSamples = 1 + (numSamples / pWTIntFrame->frame.phaseIncrement);
         } else {
diff --git a/test/Android.bp b/test/Android.bp
deleted file mode 100644
index 4ce7e85..0000000
--- a/test/Android.bp
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package {
-    // See: http://go/android-license-faq
-    // A large-scale-change added 'default_applicable_licenses' to import
-    // all of the 'license_kinds' from "external_sonivox_license"
-    // to get the below license kinds:
-    //   SPDX-license-identifier-Apache-2.0
-    default_applicable_licenses: ["external_sonivox_license"],
-}
-
-cc_test {
-    name: "SonivoxTest",
-    gtest: true,
-
-    srcs: [ "SonivoxTest.cpp" ],
-
-    static_libs: [
-        "libsonivox",
-    ],
-
-    shared_libs: [
-        "liblog",
-    ],
-
-    cflags: [
-        "-Werror",
-        "-Wall",
-    ],
-
-    sanitize: {
-        cfi: false,
-        misc_undefined: [
-            "unsigned-integer-overflow",
-            "signed-integer-overflow",
-        ],
-    },
-}
diff --git a/test/AndroidTest.xml b/test/AndroidTest.xml
deleted file mode 100644
index 17a36bd..0000000
--- a/test/AndroidTest.xml
+++ /dev/null
@@ -1,30 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<!-- Copyright (C) 2020 The Android Open Source Project
-
-     Licensed under the Apache License, Version 2.0 (the "License");
-     you may not use this file except in compliance with the License.
-     You may obtain a copy of the License at
-
-          http://www.apache.org/licenses/LICENSE-2.0
-
-     Unless required by applicable law or agreed to in writing, software
-     distributed under the License is distributed on an "AS IS" BASIS,
-     WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-     See the License for the specific language governing permissions and
-     limitations under the License.
--->
-<configuration description="Test module config for SonivoxTest unit test">
-    <option name="test-suite-tag" value="SonivoxTest" />
-    <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
-        <option name="cleanup" value="true" />
-        <option name="push" value="SonivoxTest->/data/local/tmp/SonivoxTest" />
-        <option name="push-file"
-        key="https://storage.googleapis.com/android_media/external/sonivox/test/SonivoxTestRes-1.0.zip?unzip=true"
-        value="/data/local/tmp/SonivoxTestRes/" />
-    </target_preparer>
-    <test class="com.android.tradefed.testtype.GTest" >
-        <option name="native-test-device-path" value="/data/local/tmp" />
-        <option name="module-name" value="SonivoxTest" />
-        <option name="native-test-flag" value="-P /data/local/tmp/SonivoxTestRes/" />
-    </test>
-</configuration>
diff --git a/test/README.md b/test/README.md
deleted file mode 100644
index 9c0eed3..0000000
--- a/test/README.md
+++ /dev/null
@@ -1,39 +0,0 @@
-## Media Testing ##
----
-#### Sonivox Unit Test
-The Sonivox Unit Test Suite validates the Sonivox library available in external/sonivox/
-
-Run the following steps to build the test suite:
-```
-m SonivoxTest
-```
-
-The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
-
-The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
-
-To test 64-bit binary push binaries from nativetest64.
-```
-adb push ${OUT}/data/nativetest64/SonivoxTest/SonivoxTest /data/local/tmp/
-```
-
-To test 32-bit binary push binaries from nativetest.
-```
-adb push ${OUT}/data/nativetest/SonivoxTest/SonivoxTest /data/local/tmp/
-```
-
-The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/external/sonivox/test/SonivoxTestRes-1.0.zip). Download, unzip and push these files into device for testing.
-
-```
-adb push SonivoxTestRes-1.0/. /data/local/tmp/SonivoxTestRes/
-```
-
-usage: SonivoxTest -P \<path_to_res_folder\> -C <remove_output_file>
-```
-adb shell /data/local/tmp/SonivoxTest -P /data/local/tmp/SonivoxTestRes/ -C true
-```
-Alternatively, the test can also be run using atest command.
-
-```
-atest SonivoxTest -- --enable-module-dynamic-download=true
-```
diff --git a/test/SonivoxTest.cpp b/test/SonivoxTest.cpp
deleted file mode 100644
index 5894b50..0000000
--- a/test/SonivoxTest.cpp
+++ /dev/null
@@ -1,368 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "SonivoxTest"
-#include <utils/Log.h>
-
-#include <fcntl.h>
-#include <unistd.h>
-#include <fstream>
-
-#include <libsonivox/eas.h>
-#include <libsonivox/eas_reverb.h>
-
-#include "SonivoxTestEnvironment.h"
-
-#define OUTPUT_FILE "/data/local/tmp/output_midi.pcm"
-
-// number of Sonivox output buffers to aggregate into one MediaBuffer
-static constexpr uint32_t kNumBuffersToCombine = 4;
-static constexpr uint32_t kSeekBeyondPlayTimeOffsetMs = 10;
-
-static SonivoxTestEnvironment *gEnv = nullptr;
-static int readAt(void *, void *, int, int);
-static int getSize(void *);
-
-class SonivoxTest : public ::testing::TestWithParam<tuple</*fileName*/ string,
-                                                          /*audioPlayTimeMs*/ uint32_t,
-                                                          /*totalChannels*/ uint32_t,
-                                                          /*sampleRateHz*/ uint32_t>> {
-  public:
-    SonivoxTest()
-        : mFd(-1),
-          mInputFp(nullptr),
-          mEASDataHandle(nullptr),
-          mEASStreamHandle(nullptr),
-          mPCMBuffer(nullptr),
-          mAudioBuffer(nullptr),
-          mEASConfig(nullptr) {}
-
-    ~SonivoxTest() {
-        if (mInputFp) fclose(mInputFp);
-        if (mFd >= 0) close(mFd);
-        if (mPCMBuffer) {
-            delete[] mPCMBuffer;
-            mPCMBuffer = nullptr;
-        }
-        if (mAudioBuffer) {
-            delete[] mAudioBuffer;
-            mAudioBuffer = nullptr;
-        }
-        if (gEnv->cleanUp()) remove(OUTPUT_FILE);
-    }
-
-    virtual void SetUp() override {
-        tuple<string, uint32_t, uint32_t, uint32_t> params = GetParam();
-        mInputMediaFile = gEnv->getRes() + get<0>(params);
-        mAudioplayTimeMs = get<1>(params);
-        mTotalAudioChannels = get<2>(params);
-        mAudioSampleRate = get<3>(params);
-
-        mFd = open(mInputMediaFile.c_str(), O_RDONLY | O_LARGEFILE);
-        ASSERT_GE(mFd, 0) << "Failed to get the file descriptor for file: " << mInputMediaFile;
-
-        struct stat buf;
-        int8_t err = stat(mInputMediaFile.c_str(), &buf);
-        ASSERT_EQ(err, 0) << "Failed to get information for file: " << mInputMediaFile;
-
-        mBase = 0;
-        mLength = buf.st_size;
-        mEasFile.handle = this;
-        mEasFile.readAt = ::readAt;
-        mEasFile.size = ::getSize;
-
-        EAS_RESULT result = EAS_Init(&mEASDataHandle);
-        ASSERT_EQ(result, EAS_SUCCESS) << "Failed to initialize synthesizer library";
-
-        ASSERT_NE(mEASDataHandle, nullptr) << "Failed to initialize EAS data handle";
-
-        result = EAS_OpenFile(mEASDataHandle, &mEasFile, &mEASStreamHandle);
-        ASSERT_EQ(result, EAS_SUCCESS) << "Failed to open file";
-
-        ASSERT_NE(mEASStreamHandle, nullptr) << "Failed to initialize EAS stream handle";
-
-        result = EAS_Prepare(mEASDataHandle, mEASStreamHandle);
-        ASSERT_EQ(result, EAS_SUCCESS) << "Failed to prepare EAS data and stream handles";
-
-        EAS_I32 playTimeMs;
-        result = EAS_ParseMetaData(mEASDataHandle, mEASStreamHandle, &playTimeMs);
-        ASSERT_EQ(result, EAS_SUCCESS) << "Failed to parse meta data";
-
-        ASSERT_EQ(playTimeMs, mAudioplayTimeMs)
-                << "Invalid audio play time found for file: " << mInputMediaFile;
-
-        EAS_I32 locationMs = -1;
-        /* EAS_ParseMetaData resets the parser to the starting of file */
-        result = EAS_GetLocation(mEASDataHandle, mEASStreamHandle, &locationMs);
-        ASSERT_EQ(result, EAS_SUCCESS) << "Failed to get the location after parsing meta data";
-
-        ASSERT_EQ(locationMs, 0) << "Expected position: 0, found: " << locationMs;
-
-        mEASConfig = EAS_Config();
-        ASSERT_NE(mEASConfig, nullptr) << "Failed to configure the library";
-
-        ASSERT_GT(mEASConfig->mixBufferSize, 0) << "Mix buffer size must be greater than 0";
-
-        ASSERT_GT(mEASConfig->numChannels, 0) << "Number of channels must be greater than 0";
-
-        mPCMBufferSize = sizeof(EAS_PCM) * mEASConfig->mixBufferSize * mEASConfig->numChannels *
-                         kNumBuffersToCombine;
-
-        mPCMBuffer = new (std::nothrow) EAS_PCM[mPCMBufferSize];
-        ASSERT_NE(mPCMBuffer, nullptr) << "Failed to allocate a memory of size: " << mPCMBufferSize;
-
-        mAudioBuffer =
-                new (std::nothrow) EAS_PCM[mEASConfig->mixBufferSize * mEASConfig->numChannels];
-        ASSERT_NE(mAudioBuffer, nullptr) << "Failed to allocate a memory of size: "
-                                         << mEASConfig->mixBufferSize * mEASConfig->numChannels;
-    }
-
-    virtual void TearDown() {
-        EAS_RESULT result;
-        if (mEASDataHandle) {
-            if (mEASStreamHandle) {
-                result = EAS_CloseFile(mEASDataHandle, mEASStreamHandle);
-                ASSERT_EQ(result, EAS_SUCCESS) << "Failed to close audio file/stream";
-            }
-            result = EAS_Shutdown(mEASDataHandle);
-            ASSERT_EQ(result, EAS_SUCCESS)
-                    << "Failed to deallocate the resources for synthesizer library";
-        }
-    }
-
-    bool seekToLocation(EAS_I32);
-    bool renderAudio();
-    int readAt(void *buf, int offset, int size);
-    int getSize();
-
-    string mInputMediaFile;
-    uint32_t mAudioplayTimeMs;
-    uint32_t mTotalAudioChannels;
-    uint32_t mAudioSampleRate;
-    off64_t mBase;
-    int64_t mLength;
-    int mFd;
-
-    FILE *mInputFp;
-    EAS_DATA_HANDLE mEASDataHandle;
-    EAS_HANDLE mEASStreamHandle;
-    EAS_FILE mEasFile;
-    EAS_PCM *mPCMBuffer;
-    EAS_PCM *mAudioBuffer;
-    EAS_I32 mPCMBufferSize;
-    const S_EAS_LIB_CONFIG *mEASConfig;
-};
-
-static int readAt(void *handle, void *buffer, int offset, int size) {
-    return ((SonivoxTest *)handle)->readAt(buffer, offset, size);
-}
-
-static int getSize(void *handle) {
-    return ((SonivoxTest *)handle)->getSize();
-}
-
-int SonivoxTest::readAt(void *buffer, int offset, int size) {
-    if (offset > mLength) offset = mLength;
-    lseek(mFd, mBase + offset, SEEK_SET);
-    if (offset + size > mLength) {
-        size = mLength - offset;
-    }
-
-    return read(mFd, buffer, size);
-}
-
-int SonivoxTest::getSize() {
-    return mLength;
-}
-
-bool SonivoxTest::seekToLocation(EAS_I32 locationExpectedMs) {
-    EAS_RESULT result = EAS_Locate(mEASDataHandle, mEASStreamHandle, locationExpectedMs, false);
-    if (result != EAS_SUCCESS) return false;
-
-    // position in milliseconds
-    EAS_I32 locationReceivedMs;
-    result = EAS_GetLocation(mEASDataHandle, mEASStreamHandle, &locationReceivedMs);
-    if (result != EAS_SUCCESS) return false;
-
-    if (locationReceivedMs != locationExpectedMs) return false;
-
-    return true;
-}
-
-bool SonivoxTest::renderAudio() {
-    EAS_I32 count = -1;
-    EAS_PCM *pcm = mAudioBuffer;
-
-    EAS_RESULT result = EAS_Render(mEASDataHandle, pcm, mEASConfig->mixBufferSize, &count);
-    if (result != EAS_SUCCESS) {
-        ALOGE("Failed to render audio");
-        return false;
-    }
-    if (count != mEASConfig->mixBufferSize) {
-        ALOGE("%ld of %ld bytes rendered", count, mEASConfig->mixBufferSize);
-        return false;
-    }
-
-    return true;
-}
-
-TEST_P(SonivoxTest, DecodeTest) {
-    EAS_I32 totalChannels = mEASConfig->numChannels;
-    ASSERT_EQ(totalChannels, mTotalAudioChannels)
-            << "Expected: " << mTotalAudioChannels << " channels, Found: " << totalChannels;
-
-    EAS_I32 sampleRate = mEASConfig->sampleRate;
-    ASSERT_EQ(sampleRate, mAudioSampleRate)
-            << "Expected: " << mAudioSampleRate << " sample rate, Found: " << sampleRate;
-
-    // TODO(b/158231824): Check and verify the output with other parameters present at eas_reverb.h
-    // select reverb preset and enable
-    EAS_RESULT result = EAS_SetParameter(mEASDataHandle, EAS_MODULE_REVERB, EAS_PARAM_REVERB_PRESET,
-                                         EAS_PARAM_REVERB_CHAMBER);
-    ASSERT_EQ(result, EAS_SUCCESS)
-            << "Failed to set reverberation preset parameter in reverb module";
-
-    result =
-            EAS_SetParameter(mEASDataHandle, EAS_MODULE_REVERB, EAS_PARAM_REVERB_BYPASS, EAS_FALSE);
-    ASSERT_EQ(result, EAS_SUCCESS)
-            << "Failed to set reverberation bypass parameter in reverb module";
-
-    EAS_I32 count;
-    EAS_STATE state;
-
-    FILE *filePtr = fopen(OUTPUT_FILE, "wb");
-    ASSERT_NE(filePtr, nullptr) << "Failed to open file: " << OUTPUT_FILE;
-
-    while (1) {
-        EAS_PCM *pcm = mPCMBuffer;
-        int32_t numBytesOutput = 0;
-        result = EAS_State(mEASDataHandle, mEASStreamHandle, &state);
-        ASSERT_EQ(result, EAS_SUCCESS) << "Failed to get EAS State";
-
-        ASSERT_NE(state, EAS_STATE_ERROR) << "Error state found";
-
-        /* is playback complete */
-        if (state == EAS_STATE_STOPPED) {
-            break;
-        }
-
-        EAS_I32 locationMs;
-        result = EAS_GetLocation(mEASDataHandle, mEASStreamHandle, &locationMs);
-        ASSERT_EQ(result, EAS_SUCCESS) << "Failed to get the current location in ms";
-
-        if (locationMs >= mAudioplayTimeMs) {
-            ASSERT_NE(state, EAS_STATE_STOPPED)
-                    << "Invalid state reached when rendering is complete";
-
-            break;
-        }
-
-        for (uint32_t i = 0; i < kNumBuffersToCombine; i++) {
-            result = EAS_Render(mEASDataHandle, pcm, mEASConfig->mixBufferSize, &count);
-            ASSERT_EQ(result, EAS_SUCCESS) << "Failed to render the audio data";
-
-            pcm += count * mEASConfig->numChannels;
-            numBytesOutput += count * mEASConfig->numChannels * sizeof(EAS_PCM);
-        }
-        int32_t numBytes = fwrite(mPCMBuffer, 1, numBytesOutput, filePtr);
-        ASSERT_EQ(numBytes, numBytesOutput)
-                << "Wrote " << numBytes << " of " << numBytesOutput << " to file: " << OUTPUT_FILE;
-    }
-    fclose(filePtr);
-}
-
-TEST_P(SonivoxTest, SeekTest) {
-    bool status = seekToLocation(0);
-    ASSERT_TRUE(status) << "Seek test failed for location(ms): 0";
-
-    status = seekToLocation(mAudioplayTimeMs / 2);
-    ASSERT_TRUE(status) << "Seek test failed for location(ms): " << mAudioplayTimeMs / 2;
-
-    status = seekToLocation(mAudioplayTimeMs);
-    ASSERT_TRUE(status) << "Seek test failed for location(ms): " << mAudioplayTimeMs;
-
-    status = seekToLocation(mAudioplayTimeMs + kSeekBeyondPlayTimeOffsetMs);
-    ASSERT_FALSE(status) << "Invalid seek position: "
-                         << mAudioplayTimeMs + kSeekBeyondPlayTimeOffsetMs;
-}
-
-TEST_P(SonivoxTest, DecodePauseResumeTest) {
-    EAS_I32 seekPosition = mAudioplayTimeMs / 2;
-    // go to middle of the audio
-    EAS_RESULT result = EAS_Locate(mEASDataHandle, mEASStreamHandle, seekPosition, false);
-    ASSERT_EQ(result, EAS_SUCCESS) << "Failed to locate to location(ms): " << seekPosition;
-
-    bool status = renderAudio();
-    ASSERT_TRUE(status) << "Failed to render audio";
-
-    result = EAS_Pause(mEASDataHandle, mEASStreamHandle);
-    ASSERT_EQ(result, EAS_SUCCESS) << "Failed to pause";
-
-    // will render previous audio again, no change in audio position
-    status = renderAudio();
-    ASSERT_TRUE(status) << "should not move audio position, since we're paused";
-
-    // current position in milliseconds
-    EAS_I32 currentPosMs = -1;
-    result = EAS_GetLocation(mEASDataHandle, mEASStreamHandle, &currentPosMs);
-    ASSERT_EQ(result, EAS_SUCCESS) << "Failed to get current location";
-
-    ASSERT_EQ(currentPosMs, seekPosition) << "Must not move the audio position after pause";
-
-    EAS_STATE state;
-    result = EAS_State(mEASDataHandle, mEASStreamHandle, &state);
-    ASSERT_EQ(result, EAS_SUCCESS) << "Failed to get EAS state";
-
-    ASSERT_EQ(state, EAS_STATE_PAUSED) << "Invalid state reached when paused";
-
-    result = EAS_Resume(mEASDataHandle, mEASStreamHandle);
-    ASSERT_EQ(result, EAS_SUCCESS) << "Failed to resume";
-
-    status = renderAudio();
-    ASSERT_TRUE(status) << "Failed to render audio after resume";
-
-    currentPosMs = -1;
-    result = EAS_GetLocation(mEASDataHandle, mEASStreamHandle, &currentPosMs);
-    ASSERT_EQ(result, EAS_SUCCESS) << "Failed to get current location";
-
-    ASSERT_GT(currentPosMs, seekPosition) << "Invalid position after resuming";
-
-    result = EAS_State(mEASDataHandle, mEASStreamHandle, &state);
-    ASSERT_EQ(result, EAS_SUCCESS) << "Failed to get EAS state";
-
-    ASSERT_EQ(state, EAS_STATE_PLAY) << "Invalid state reached when resumed";
-}
-
-INSTANTIATE_TEST_SUITE_P(SonivoxTestAll, SonivoxTest,
-                         ::testing::Values(make_tuple("midi_a.mid", 2000, 2, 22050),
-                                           make_tuple("midi8sec.mid", 8002, 2, 22050),
-                                           make_tuple("midi_cs.mid", 2000, 2, 22050),
-                                           make_tuple("midi_gs.mid", 2000, 2, 22050),
-                                           make_tuple("ants.mid", 17233, 2, 22050),
-                                           make_tuple("testmxmf.mxmf", 29095, 2, 22050)));
-
-int main(int argc, char **argv) {
-    gEnv = new SonivoxTestEnvironment();
-    ::testing::AddGlobalTestEnvironment(gEnv);
-    ::testing::InitGoogleTest(&argc, argv);
-    int status = gEnv->initFromOptions(argc, argv);
-    if (status == 0) {
-        status = RUN_ALL_TESTS();
-        ALOGV("Test result = %d\n", status);
-    }
-    return status;
-}
diff --git a/test/SonivoxTestEnvironment.h b/test/SonivoxTestEnvironment.h
deleted file mode 100644
index 1b1690d..0000000
--- a/test/SonivoxTestEnvironment.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef __SONIVOX_TEST_ENVIRONMENT_H__
-#define __SONIVOX_TEST_ENVIRONMENT_H__
-
-#include <gtest/gtest.h>
-
-#include <getopt.h>
-
-using namespace std;
-
-class SonivoxTestEnvironment : public::testing::Environment {
-  public:
-    SonivoxTestEnvironment() : res("/data/local/tmp/"), deleteOutput(true){}
-
-    // Parses the command line arguments
-    int initFromOptions(int argc, char **argv);
-
-    void setRes(const char *_res) { res = _res; }
-
-    const string getRes() const { return res; }
-
-    bool cleanUp() const { return deleteOutput; }
-
-  private:
-    string res;
-    bool deleteOutput;
-};
-
-int SonivoxTestEnvironment::initFromOptions(int argc, char **argv) {
-    static struct option options[] = {{"res", required_argument, 0, 'P'},
-                                      {"cleanUp", optional_argument, 0, 'C'},
-                                      {0, 0, 0, 0}};
-
-    while (true) {
-        int index = 0;
-        int c = getopt_long(argc, argv, "P:C:", options, &index);
-        if (c == -1) {
-            break;
-        }
-
-        switch (c) {
-            case 'P': {
-                setRes(optarg);
-                break;
-            }
-            case 'C':
-                if (!strcmp(optarg, "false")) {
-                    deleteOutput = false;
-                }
-                break;
-            default:
-                break;
-        }
-    }
-
-    if (optind < argc) {
-        fprintf(stderr,
-                "unrecognized option: %s\n\n"
-                "usage: %s <gtest options> <test options>\n\n"
-                "test options are:\n\n"
-                "-P, --path: Resource files directory location\n"
-                "-C, default:true. Delete output file after test completes\n",
-                argv[optind ?: 1], argv[0]);
-        return 2;
-    }
-    return 0;
-}
-
-#endif  // __SONIVOX_TEST_ENVIRONMENT_H__