Snap for 8426163 from 7f5ca1faeccbd81eda47e36261d1fed1577ce327 to mainline-tzdata2-release
Change-Id: I29c1230bdf2c32fac2da050918fa57c9f902735f
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 @@
"liblog",
],
- host_supported: true,
-
- target: {
- darwin: {
- enabled: false,
- },
- },
-
arch: {
arm: {
instruction_set: "arm",
@@ -111,15 +86,22 @@
// 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 @@
"-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 @@
@RestoreRegs RLIST {r4-r10, pc}
+ .func WT_VoiceFilter
WT_VoiceFilter:
STMFD sp!, {r4-r10, lr}
@@ -111,7 +112,7 @@
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 @@
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 @@
@SaveRegs RLIST {r4-r11,lr}
@RestoreRegs RLIST {r4-r11,pc}
+ .func WT_Interpolate
WT_Interpolate:
STMFD sp!,{r4-r11,lr}
@@ -80,15 +81,13 @@
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 @@
@ 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 @@
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 @@
@SaveRegs RLIST {r4-r9,lr}
@RestoreRegs RLIST {r4-r9,pc}
+ .func WT_InterpolateNoLoop
WT_InterpolateNoLoop:
STMFD sp!, {r4-r9,lr}
@@ -72,15 +73,13 @@
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 @@
@ 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 @@
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 @@
@*****************************************************************************
+ .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 @@
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 @@
@ 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 @@
@----------------------------------------------------------------
@ Mono version
@----------------------------------------------------------------
- #else
+ .else
LDR gain, [pWTFrame, #m_prevGain]
MOV gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)
@@ -155,10 +156,11 @@
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 @@
/* 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 @@
}
/* 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 @@
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 @@
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 @@
while (numSamples--) {
- EAS_I32 nextSamplePhaseInc;
-
/* linear interpolation */
acc0 = samp2 - samp1;
acc0 = acc0 * phaseFrac;
@@ -233,19 +231,19 @@
/* 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) {
- /* advance sample pointer */
- pSamples += nextSamplePhaseInc;
- phaseFrac = phaseFrac & PHASE_FRAC_MASK;
+ if (acc0 > 0) {
- /* decrementing pSamples by entire buffer length until second pSample is within */
- /* loopEnd */
- while (&pSamples[1] >= loopEnd) {
- pSamples -= (loopEnd - (const EAS_SAMPLE*)pWTVoice->loopStart);
- }
+ /* advance sample pointer */
+ pSamples += acc0;
+ phaseFrac = (EAS_I32)((EAS_U32)phaseFrac & PHASE_FRAC_MASK);
+
+ /* 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 @@
#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, ¤tPosMs);
- 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, ¤tPosMs);
- 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__
