Audio: Fixes for FCC_LIMIT validation
Test: SoundPool(Aac|Haptic|Midi|Ogg)Test
Test: AudioEffectTest, AudioPreProcessingTest
Test: BassBoostTest, EnvReverbTest, EqualizerTest, LoudnessEnhancerTest
Test: PresetReverbTest, VirtualizerTest, VisualizerTest
Bug: 189325443
Change-Id: I0ce958a2e111d8f55391a00fafbc95135bee288e
diff --git a/media/libaudioprocessing/AudioResamplerDyn.cpp b/media/libaudioprocessing/AudioResamplerDyn.cpp
index 21d3d36..2292b19 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.cpp
+++ b/media/libaudioprocessing/AudioResamplerDyn.cpp
@@ -548,61 +548,73 @@
LOG_ALWAYS_FATAL_IF(mChannelCount < 1 || mChannelCount > FCC_LIMIT,
"Resampler channels(%d) must be between 1 to %d", mChannelCount, FCC_LIMIT);
// stride 16 (falls back to stride 2 for machines that do not support NEON)
+
+
+// For now use a #define as a compiler generated function table requires renaming.
+#pragma push_macro("AUDIORESAMPLERDYN_CASE")
+#undef AUDIORESAMPLERDYN_CASE
+#define AUDIORESAMPLERDYN_CASE(CHANNEL, LOCKED) \
+ case CHANNEL: if constexpr (CHANNEL <= FCC_LIMIT) {\
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<CHANNEL, LOCKED, 16>; \
+ } break
+
if (locked) {
switch (mChannelCount) {
- case 1:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>;
- break;
- case 2:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>;
- break;
- case 3:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, true, 16>;
- break;
- case 4:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, true, 16>;
- break;
- case 5:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, true, 16>;
- break;
- case 6:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, true, 16>;
- break;
- case 7:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, true, 16>;
- break;
- case 8:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, true, 16>;
- break;
+ AUDIORESAMPLERDYN_CASE(1, true);
+ AUDIORESAMPLERDYN_CASE(2, true);
+ AUDIORESAMPLERDYN_CASE(3, true);
+ AUDIORESAMPLERDYN_CASE(4, true);
+ AUDIORESAMPLERDYN_CASE(5, true);
+ AUDIORESAMPLERDYN_CASE(6, true);
+ AUDIORESAMPLERDYN_CASE(7, true);
+ AUDIORESAMPLERDYN_CASE(8, true);
+ AUDIORESAMPLERDYN_CASE(9, true);
+ AUDIORESAMPLERDYN_CASE(10, true);
+ AUDIORESAMPLERDYN_CASE(11, true);
+ AUDIORESAMPLERDYN_CASE(12, true);
+ AUDIORESAMPLERDYN_CASE(13, true);
+ AUDIORESAMPLERDYN_CASE(14, true);
+ AUDIORESAMPLERDYN_CASE(15, true);
+ AUDIORESAMPLERDYN_CASE(16, true);
+ AUDIORESAMPLERDYN_CASE(17, true);
+ AUDIORESAMPLERDYN_CASE(18, true);
+ AUDIORESAMPLERDYN_CASE(19, true);
+ AUDIORESAMPLERDYN_CASE(20, true);
+ AUDIORESAMPLERDYN_CASE(21, true);
+ AUDIORESAMPLERDYN_CASE(22, true);
+ AUDIORESAMPLERDYN_CASE(23, true);
+ AUDIORESAMPLERDYN_CASE(24, true);
}
} else {
switch (mChannelCount) {
- case 1:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>;
- break;
- case 2:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>;
- break;
- case 3:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, false, 16>;
- break;
- case 4:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, false, 16>;
- break;
- case 5:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, false, 16>;
- break;
- case 6:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, false, 16>;
- break;
- case 7:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, false, 16>;
- break;
- case 8:
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, false, 16>;
- break;
+ AUDIORESAMPLERDYN_CASE(1, false);
+ AUDIORESAMPLERDYN_CASE(2, false);
+ AUDIORESAMPLERDYN_CASE(3, false);
+ AUDIORESAMPLERDYN_CASE(4, false);
+ AUDIORESAMPLERDYN_CASE(5, false);
+ AUDIORESAMPLERDYN_CASE(6, false);
+ AUDIORESAMPLERDYN_CASE(7, false);
+ AUDIORESAMPLERDYN_CASE(8, false);
+ AUDIORESAMPLERDYN_CASE(9, false);
+ AUDIORESAMPLERDYN_CASE(10, false);
+ AUDIORESAMPLERDYN_CASE(11, false);
+ AUDIORESAMPLERDYN_CASE(12, false);
+ AUDIORESAMPLERDYN_CASE(13, false);
+ AUDIORESAMPLERDYN_CASE(14, false);
+ AUDIORESAMPLERDYN_CASE(15, false);
+ AUDIORESAMPLERDYN_CASE(16, false);
+ AUDIORESAMPLERDYN_CASE(17, false);
+ AUDIORESAMPLERDYN_CASE(18, false);
+ AUDIORESAMPLERDYN_CASE(19, false);
+ AUDIORESAMPLERDYN_CASE(20, false);
+ AUDIORESAMPLERDYN_CASE(21, false);
+ AUDIORESAMPLERDYN_CASE(22, false);
+ AUDIORESAMPLERDYN_CASE(23, false);
+ AUDIORESAMPLERDYN_CASE(24, false);
}
}
+#pragma pop_macro("AUDIORESAMPLERDYN_CASE")
+
#ifdef DEBUG_RESAMPLER
printf("channels:%d %s stride:%d %s coef:%d shift:%d\n",
mChannelCount, locked ? "locked" : "interpolated",