Merge "EffectDownmix: Move processing to audio_utils ChannelMix" into sc-v2-dev
diff --git a/media/libeffects/downmix/EffectDownmix.cpp b/media/libeffects/downmix/EffectDownmix.cpp
index f500bc3..90bb410 100644
--- a/media/libeffects/downmix/EffectDownmix.cpp
+++ b/media/libeffects/downmix/EffectDownmix.cpp
@@ -19,7 +19,7 @@
#include <log/log.h>
#include "EffectDownmix.h"
-#include <math.h>
+#include <audio_utils/ChannelMix.h>
// Do not submit with DOWNMIX_TEST_CHANNEL_INDEX defined, strictly for testing
//#define DOWNMIX_TEST_CHANNEL_INDEX 0
@@ -35,12 +35,13 @@
} downmix_state_t;
/* parameters for each downmixer */
-typedef struct {
+struct downmix_object_t {
downmix_state_t state;
downmix_type_t type;
bool apply_volume_correction;
uint8_t input_channel_count;
-} downmix_object_t;
+ android::audio_utils::channels::ChannelMix channelMix;
+};
typedef struct downmix_module_s {
const struct effect_interface_s *itfe;
@@ -77,11 +78,6 @@
downmix_object_t *pDownmixer, int32_t param, uint32_t size, void *pValue);
static int Downmix_getParameter(
downmix_object_t *pDownmixer, int32_t param, uint32_t *pSize, void *pValue);
-static void Downmix_foldFromQuad(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
-static void Downmix_foldFrom5Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
-static void Downmix_foldFrom7Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
-static bool Downmix_foldGeneric(
- uint32_t mask, float *pSrc, float *pDst, size_t numFrames, bool accumulate);
// effect_handle_t interface implementation for downmix effect
const struct effect_interface_s gDownmixInterface = {
@@ -315,7 +311,8 @@
audio_buffer_t *inBuffer, audio_buffer_t *outBuffer) {
downmix_object_t *pDownmixer;
- float *pSrc, *pDst;
+ const float *pSrc;
+ float *pDst;
downmix_module_t *pDwmModule = (downmix_module_t *)self;
if (pDwmModule == NULL) {
@@ -344,7 +341,8 @@
const bool accumulate =
(pDwmModule->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE);
- const uint32_t downmixInputChannelMask = pDwmModule->config.inputCfg.channels;
+ const audio_channel_mask_t downmixInputChannelMask =
+ (audio_channel_mask_t)pDwmModule->config.inputCfg.channels;
switch(pDownmixer->type) {
@@ -368,38 +366,13 @@
}
break;
- case DOWNMIX_TYPE_FOLD:
-#ifdef DOWNMIX_ALWAYS_USE_GENERIC_DOWNMIXER
- // bypass the optimized downmix routines for the common formats
- if (!Downmix_foldGeneric(
- downmixInputChannelMask, pSrc, pDst, numFrames, accumulate)) {
- ALOGE("Multichannel configuration %#x is not supported",
- downmixInputChannelMask);
- return -EINVAL;
- }
- break;
-#endif
- // optimize for the common formats
- switch (downmixInputChannelMask) {
- case AUDIO_CHANNEL_OUT_QUAD_BACK:
- case AUDIO_CHANNEL_OUT_QUAD_SIDE:
- Downmix_foldFromQuad(pSrc, pDst, numFrames, accumulate);
- break;
- case AUDIO_CHANNEL_OUT_5POINT1_BACK:
- case AUDIO_CHANNEL_OUT_5POINT1_SIDE:
- Downmix_foldFrom5Point1(pSrc, pDst, numFrames, accumulate);
- break;
- case AUDIO_CHANNEL_OUT_7POINT1:
- Downmix_foldFrom7Point1(pSrc, pDst, numFrames, accumulate);
- break;
- default:
- if (!Downmix_foldGeneric(
- downmixInputChannelMask, pSrc, pDst, numFrames, accumulate)) {
+ case DOWNMIX_TYPE_FOLD: {
+ if (!pDownmixer->channelMix.process(
+ pSrc, pDst, numFrames, accumulate, downmixInputChannelMask)) {
ALOGE("Multichannel configuration %#x is not supported",
downmixInputChannelMask);
return -EINVAL;
}
- break;
}
break;
@@ -780,7 +753,6 @@
return 0;
} /* end Downmix_setParameter */
-
/*----------------------------------------------------------------------------
* Downmix_getParameter()
*----------------------------------------------------------------------------
@@ -829,299 +801,3 @@
return 0;
} /* end Downmix_getParameter */
-
-/*----------------------------------------------------------------------------
- * Downmix_foldFromQuad()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix a quad signal to stereo
- *
- * Inputs:
- * pSrc quad audio samples to downmix
- * numFrames the number of quad frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- *----------------------------------------------------------------------------
- */
-void Downmix_foldFromQuad(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
- // sample at index 0 is FL
- // sample at index 1 is FR
- // sample at index 2 is RL
- // sample at index 3 is RR
- if (accumulate) {
- while (numFrames) {
- // FL + RL
- pDst[0] = clamp_float(pDst[0] + ((pSrc[0] + pSrc[2]) / 2.0f));
- // FR + RR
- pDst[1] = clamp_float(pDst[1] + ((pSrc[1] + pSrc[3]) / 2.0f));
- pSrc += 4;
- pDst += 2;
- numFrames--;
- }
- } else { // same code as above but without adding and clamping pDst[i] to itself
- while (numFrames) {
- // FL + RL
- pDst[0] = clamp_float((pSrc[0] + pSrc[2]) / 2.0f);
- // FR + RR
- pDst[1] = clamp_float((pSrc[1] + pSrc[3]) / 2.0f);
- pSrc += 4;
- pDst += 2;
- numFrames--;
- }
- }
-}
-
-/*----------------------------------------------------------------------------
- * Downmix_foldFrom5Point1()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix a 5.1 signal to stereo
- *
- * Inputs:
- * pSrc 5.1 audio samples to downmix
- * numFrames the number of 5.1 frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- *----------------------------------------------------------------------------
- */
-void Downmix_foldFrom5Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
- float lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
- // sample at index 0 is FL
- // sample at index 1 is FR
- // sample at index 2 is FC
- // sample at index 3 is LFE
- // sample at index 4 is RL
- // sample at index 5 is RR
- // code is mostly duplicated between the two values of accumulate to avoid repeating the test
- // for every sample
- if (accumulate) {
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[4];
- // FR + centerPlusLfeContrib + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[5];
- // accumulate in destination
- pDst[0] = clamp_float(pDst[0] + (lt / 2.0f));
- pDst[1] = clamp_float(pDst[1] + (rt / 2.0f));
- pSrc += 6;
- pDst += 2;
- numFrames--;
- }
- } else { // same code as above but without adding and clamping pDst[i] to itself
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[4];
- // FR + centerPlusLfeContrib + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[5];
- // store in destination
- pDst[0] = clamp_float(lt / 2.0f); // differs from when accumulate is true above
- pDst[1] = clamp_float(rt / 2.0f); // differs from when accumulate is true above
- pSrc += 6;
- pDst += 2;
- numFrames--;
- }
- }
-}
-
-/*----------------------------------------------------------------------------
- * Downmix_foldFrom7Point1()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix a 7.1 signal to stereo
- *
- * Inputs:
- * pSrc 7.1 audio samples to downmix
- * numFrames the number of 7.1 frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- *----------------------------------------------------------------------------
- */
-void Downmix_foldFrom7Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
- float lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
- // sample at index 0 is FL
- // sample at index 1 is FR
- // sample at index 2 is FC
- // sample at index 3 is LFE
- // sample at index 4 is RL
- // sample at index 5 is RR
- // sample at index 6 is SL
- // sample at index 7 is SR
- // code is mostly duplicated between the two values of accumulate to avoid repeating the test
- // for every sample
- if (accumulate) {
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + SL + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[6] + pSrc[4];
- // FR + centerPlusLfeContrib + SR + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[7] + pSrc[5];
- //accumulate in destination
- pDst[0] = clamp_float(pDst[0] + (lt / 2.0f));
- pDst[1] = clamp_float(pDst[1] + (rt / 2.0f));
- pSrc += 8;
- pDst += 2;
- numFrames--;
- }
- } else { // same code as above but without adding and clamping pDst[i] to itself
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + SL + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[6] + pSrc[4];
- // FR + centerPlusLfeContrib + SR + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[7] + pSrc[5];
- // store in destination
- pDst[0] = clamp_float(lt / 2.0f); // differs from when accumulate is true above
- pDst[1] = clamp_float(rt / 2.0f); // differs from when accumulate is true above
- pSrc += 8;
- pDst += 2;
- numFrames--;
- }
- }
-}
-
-/*----------------------------------------------------------------------------
- * Downmix_foldGeneric()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix to stereo a multichannel signal of arbitrary channel position mask.
- *
- * Inputs:
- * mask the channel mask of pSrc
- * pSrc multichannel audio buffer to downmix
- * numFrames the number of multichannel frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- * Returns: false if multichannel format is not supported
- *
- *----------------------------------------------------------------------------
- */
-bool Downmix_foldGeneric(
- uint32_t mask, float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
-
- if (!Downmix_validChannelMask(mask)) {
- return false;
- }
- const int numChan = audio_channel_count_from_out_mask(mask);
-
- // compute at what index each channel is: samples will be in the following order:
- // FL FR FC LFE BL BR BC SL SR
- //
- // (transfer matrix)
- // FL FR FC LFE BL BR BC SL SR
- // 0.5 0.353 0.353 0.5 0.353 0.5
- // 0.5 0.353 0.353 0.5 0.353 0.5
-
- // derive the indices for the transfer matrix columns that have non-zero values.
- int indexFL = -1;
- int indexFR = -1;
- int indexFC = -1;
- int indexLFE = -1;
- int indexBL = -1;
- int indexBR = -1;
- int indexBC = -1;
- int indexSL = -1;
- int indexSR = -1;
- int index = 0;
- for (unsigned tmp = mask;
- (tmp & (AUDIO_CHANNEL_OUT_7POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER)) != 0;
- ++index) {
- const unsigned lowestBit = tmp & -(signed)tmp;
- switch (lowestBit) {
- case AUDIO_CHANNEL_OUT_FRONT_LEFT:
- indexFL = index;
- break;
- case AUDIO_CHANNEL_OUT_FRONT_RIGHT:
- indexFR = index;
- break;
- case AUDIO_CHANNEL_OUT_FRONT_CENTER:
- indexFC = index;
- break;
- case AUDIO_CHANNEL_OUT_LOW_FREQUENCY:
- indexLFE = index;
- break;
- case AUDIO_CHANNEL_OUT_BACK_LEFT:
- indexBL = index;
- break;
- case AUDIO_CHANNEL_OUT_BACK_RIGHT:
- indexBR = index;
- break;
- case AUDIO_CHANNEL_OUT_BACK_CENTER:
- indexBC = index;
- break;
- case AUDIO_CHANNEL_OUT_SIDE_LEFT:
- indexSL = index;
- break;
- case AUDIO_CHANNEL_OUT_SIDE_RIGHT:
- indexSR = index;
- break;
- }
- tmp ^= lowestBit;
- }
-
- // With good branch prediction, this should run reasonably fast.
- // Also consider using a transfer matrix form.
- while (numFrames) {
- // compute contribution of FC, BC and LFE
- float centersLfeContrib = 0;
- if (indexFC >= 0) centersLfeContrib = pSrc[indexFC];
- if (indexLFE >= 0) centersLfeContrib += pSrc[indexLFE];
- if (indexBC >= 0) centersLfeContrib += pSrc[indexBC];
- centersLfeContrib *= MINUS_3_DB_IN_FLOAT;
-
- float ch[2];
- ch[0] = centersLfeContrib;
- ch[1] = centersLfeContrib;
-
- // mix in left / right channels
- if (indexFL >= 0) ch[0] += pSrc[indexFL];
- if (indexFR >= 0) ch[1] += pSrc[indexFR];
-
- if (indexSL >= 0) ch[0] += pSrc[indexSL];
- if (indexSR >= 0) ch[1] += pSrc[indexSR]; // note pair checks enforce this if indexSL != 0
-
- if (indexBL >= 0) ch[0] += pSrc[indexBL];
- if (indexBR >= 0) ch[1] += pSrc[indexBR]; // note pair checks enforce this if indexBL != 0
-
- // scale to prevent overflow.
- ch[0] *= 0.5f;
- ch[1] *= 0.5f;
-
- if (accumulate) {
- ch[0] += pDst[0];
- ch[1] += pDst[1];
- }
-
- pDst[0] = clamp_float(ch[0]);
- pDst[1] = clamp_float(ch[1]);
- pSrc += numChan;
- pDst += 2;
- numFrames--;
- }
- return true;
-}
diff --git a/media/libeffects/downmix/benchmark/downmix_benchmark.cpp b/media/libeffects/downmix/benchmark/downmix_benchmark.cpp
index ee169c2..d4adf5e 100644
--- a/media/libeffects/downmix/benchmark/downmix_benchmark.cpp
+++ b/media/libeffects/downmix/benchmark/downmix_benchmark.cpp
@@ -35,16 +35,14 @@
AUDIO_CHANNEL_OUT_STEREO,
AUDIO_CHANNEL_OUT_2POINT1,
AUDIO_CHANNEL_OUT_2POINT0POINT2,
- AUDIO_CHANNEL_OUT_QUAD,
- AUDIO_CHANNEL_OUT_QUAD_BACK,
+ AUDIO_CHANNEL_OUT_QUAD, // AUDIO_CHANNEL_OUT_QUAD_BACK
AUDIO_CHANNEL_OUT_QUAD_SIDE,
AUDIO_CHANNEL_OUT_SURROUND,
AUDIO_CHANNEL_OUT_2POINT1POINT2,
AUDIO_CHANNEL_OUT_3POINT0POINT2,
AUDIO_CHANNEL_OUT_PENTA,
AUDIO_CHANNEL_OUT_3POINT1POINT2,
- AUDIO_CHANNEL_OUT_5POINT1,
- AUDIO_CHANNEL_OUT_5POINT1_BACK,
+ AUDIO_CHANNEL_OUT_5POINT1, // AUDIO_CHANNEL_OUT_5POINT1_BACK
AUDIO_CHANNEL_OUT_5POINT1_SIDE,
AUDIO_CHANNEL_OUT_6POINT1,
AUDIO_CHANNEL_OUT_5POINT1POINT2,
@@ -62,58 +60,32 @@
static constexpr size_t kFrameCount = 1000;
/*
-Pixel 3XL
-downmix_benchmark:
- #BM_Downmix/0 4723 ns 4708 ns 148694
- #BM_Downmix/1 4717 ns 4702 ns 148873
- #BM_Downmix/2 4803 ns 4788 ns 145893
- #BM_Downmix/3 5056 ns 5041 ns 139110
- #BM_Downmix/4 4710 ns 4696 ns 149625
- #BM_Downmix/5 1514 ns 1509 ns 463694
- #BM_Downmix/6 1513 ns 1509 ns 463451
- #BM_Downmix/7 1516 ns 1511 ns 463899
- #BM_Downmix/8 4445 ns 4431 ns 157831
- #BM_Downmix/9 5081 ns 5065 ns 138412
- #BM_Downmix/10 4354 ns 4341 ns 161247
- #BM_Downmix/11 4411 ns 4397 ns 158893
- #BM_Downmix/12 4434 ns 4420 ns 157992
- #BM_Downmix/13 4845 ns 4830 ns 144873
- #BM_Downmix/14 4851 ns 4835 ns 144954
- #BM_Downmix/15 4884 ns 4870 ns 144233
- #BM_Downmix/16 5832 ns 5813 ns 120565
- #BM_Downmix/17 5241 ns 5224 ns 133927
- #BM_Downmix/18 5044 ns 5028 ns 139131
- #BM_Downmix/19 5244 ns 5227 ns 132315
- #BM_Downmix/20 5943 ns 5923 ns 117759
- #BM_Downmix/21 5990 ns 5971 ns 117263
- #BM_Downmix/22 4468 ns 4454 ns 156689
- #BM_Downmix/23 7306 ns 7286 ns 95911
---
-downmix_benchmark: (generic fold)
- #BM_Downmix/0 4722 ns 4707 ns 149847
- #BM_Downmix/1 4714 ns 4698 ns 148748
- #BM_Downmix/2 4794 ns 4779 ns 145661
- #BM_Downmix/3 5053 ns 5035 ns 139172
- #BM_Downmix/4 4695 ns 4678 ns 149762
- #BM_Downmix/5 4381 ns 4368 ns 159675
- #BM_Downmix/6 4387 ns 4373 ns 160267
- #BM_Downmix/7 4732 ns 4717 ns 148514
- #BM_Downmix/8 4430 ns 4415 ns 158133
- #BM_Downmix/9 5101 ns 5084 ns 138353
- #BM_Downmix/10 4356 ns 4343 ns 160821
- #BM_Downmix/11 4397 ns 4383 ns 159995
- #BM_Downmix/12 4438 ns 4424 ns 158117
- #BM_Downmix/13 5243 ns 5226 ns 133863
- #BM_Downmix/14 5259 ns 5242 ns 131855
- #BM_Downmix/15 5245 ns 5228 ns 133686
- #BM_Downmix/16 5829 ns 5809 ns 120543
- #BM_Downmix/17 5245 ns 5228 ns 133533
- #BM_Downmix/18 5935 ns 5916 ns 118282
- #BM_Downmix/19 5263 ns 5245 ns 133657
- #BM_Downmix/20 5998 ns 5978 ns 114693
- #BM_Downmix/21 5989 ns 5969 ns 117450
- #BM_Downmix/22 4442 ns 4431 ns 157913
- #BM_Downmix/23 7309 ns 7290 ns 95797
+Pixel 4XL
+--------------------------------------------------------
+Benchmark Time CPU Iterations
+--------------------------------------------------------
+BM_Downmix/0 3553 ns 3545 ns 197503 AUDIO_CHANNEL_OUT_MONO
+BM_Downmix/1 2846 ns 2840 ns 202849
+BM_Downmix/2 4436 ns 4426 ns 158176 AUDIO_CHANNEL_OUT_STEREO
+BM_Downmix/3 5320 ns 5307 ns 131870 AUDIO_CHANNEL_OUT_2POINT1
+BM_Downmix/4 4437 ns 4426 ns 159523 AUDIO_CHANNEL_OUT_2POINT0POINT2
+BM_Downmix/5 2493 ns 2487 ns 281496 AUDIO_CHANNEL_OUT_QUAD
+BM_Downmix/6 2493 ns 2487 ns 281456 AUDIO_CHANNEL_OUT_QUAD_SIDE
+BM_Downmix/7 6204 ns 6188 ns 115044 AUDIO_CHANNEL_OUT_SURROUND
+BM_Downmix/8 5320 ns 5307 ns 100000 AUDIO_CHANNEL_OUT_2POINT1POINT2
+BM_Downmix/9 5320 ns 5307 ns 100000 AUDIO_CHANNEL_OUT_3POINT0POINT2
+BM_Downmix/10 7088 ns 7071 ns 108264 AUDIO_CHANNEL_OUT_PENTA
+BM_Downmix/11 6203 ns 6188 ns 117021 AUDIO_CHANNEL_OUT_3POINT1POINT2
+BM_Downmix/12 3105 ns 3097 ns 226182 AUDIO_CHANNEL_OUT_5POINT1
+BM_Downmix/13 3112 ns 3105 ns 225488 AUDIO_CHANNEL_OUT_5POINT1_SIDE
+BM_Downmix/14 8855 ns 8831 ns 79265 AUDIO_CHANNEL_OUT_6POINT1
+BM_Downmix/15 7971 ns 7951 ns 90918 AUDIO_CHANNEL_OUT_5POINT1POINT2
+BM_Downmix/16 3547 ns 3539 ns 197780 AUDIO_CHANNEL_OUT_7POINT1
+BM_Downmix/17 7972 ns 7953 ns 90101 AUDIO_CHANNEL_OUT_5POINT1POINT4
+BM_Downmix/18 9737 ns 9714 ns 72773 AUDIO_CHANNEL_OUT_7POINT1POINT2
+BM_Downmix/19 9745 ns 9721 ns 72015 AUDIO_CHANNEL_OUT_7POINT1POINT4
+BM_Downmix/20 7070 ns 7053 ns 109476 AUDIO_CHANNEL_OUT_13POINT_360RA
+BM_Downmix/21 12413 ns 12381 ns 57455 AUDIO_CHANNEL_OUT_22POINT2
*/
static void BM_Downmix(benchmark::State& state) {
@@ -125,7 +97,7 @@
std::minstd_rand gen(channelMask);
std::uniform_real_distribution<> dis(-1.0f, 1.0f);
std::vector<float> input(kFrameCount * channelCount);
- std::vector<float> output(kFrameCount * 2);
+ std::vector<float> output(kFrameCount * FCC_2);
for (auto& in : input) {
in = dis(gen);
}
@@ -187,7 +159,8 @@
benchmark::ClobberMemory();
}
- state.SetComplexityN(state.range(0));
+ state.SetComplexityN(channelCount);
+ state.SetLabel(audio_channel_out_mask_to_string(channelMask));
if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle); status != 0) {
ALOGE("release_effect returned an error = %d\n", status);
diff --git a/media/libeffects/downmix/tests/downmix_tests.cpp b/media/libeffects/downmix/tests/downmix_tests.cpp
index d4b7a3a..d378044 100644
--- a/media/libeffects/downmix/tests/downmix_tests.cpp
+++ b/media/libeffects/downmix/tests/downmix_tests.cpp
@@ -33,16 +33,14 @@
AUDIO_CHANNEL_OUT_STEREO,
AUDIO_CHANNEL_OUT_2POINT1,
AUDIO_CHANNEL_OUT_2POINT0POINT2,
- AUDIO_CHANNEL_OUT_QUAD,
- AUDIO_CHANNEL_OUT_QUAD_BACK,
+ AUDIO_CHANNEL_OUT_QUAD, // AUDIO_CHANNEL_OUT_QUAD_BACK
AUDIO_CHANNEL_OUT_QUAD_SIDE,
AUDIO_CHANNEL_OUT_SURROUND,
AUDIO_CHANNEL_OUT_2POINT1POINT2,
AUDIO_CHANNEL_OUT_3POINT0POINT2,
AUDIO_CHANNEL_OUT_PENTA,
AUDIO_CHANNEL_OUT_3POINT1POINT2,
- AUDIO_CHANNEL_OUT_5POINT1,
- AUDIO_CHANNEL_OUT_5POINT1_BACK,
+ AUDIO_CHANNEL_OUT_5POINT1, // AUDIO_CHANNEL_OUT_5POINT1_BACK
AUDIO_CHANNEL_OUT_5POINT1_SIDE,
AUDIO_CHANNEL_OUT_6POINT1,
AUDIO_CHANNEL_OUT_5POINT1POINT2,
@@ -57,6 +55,33 @@
static constexpr audio_channel_mask_t kConsideredChannels =
(audio_channel_mask_t)(AUDIO_CHANNEL_OUT_7POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER);
+constexpr inline float kScaleFromChannelIdx[] = {
+ 0.5f, // AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1u,
+ 0.5f, // AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2u,
+ 0.5f * M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_CENTER = 0x4u,
+ 0.5f * M_SQRT1_2, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 0x8u,
+ 0.5f, // AUDIO_CHANNEL_OUT_BACK_LEFT = 0x10u,
+ 0.5f, // AUDIO_CHANNEL_OUT_BACK_RIGHT = 0x20u,
+ 0, // AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40u,
+ 0, // AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80u,
+ 0.5f * M_SQRT1_2, // AUDIO_CHANNEL_OUT_BACK_CENTER = 0x100u,
+ 0.5f, // AUDIO_CHANNEL_OUT_SIDE_LEFT = 0x200u,
+ 0.5f, // AUDIO_CHANNEL_OUT_SIDE_RIGHT = 0x400u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_CENTER = 0x800u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 0x1000u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 0x2000u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 0x4000u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 0x8000u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_SIDE_LEFT = 0x40000u,
+ 0, // AUDIO_CHANNEL_OUT_TOP_SIDE_RIGHT = 0x80000u,
+ 0, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_LEFT = 0x100000u,
+ 0, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_CENTER = 0x200000u,
+ 0, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_RIGHT = 0x400000u,
+ 0, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2 = 0x800000u,
+};
+
// Downmix doesn't change with sample rate
static constexpr size_t kSampleRates[] = {
48000,
@@ -93,8 +118,8 @@
void testBalance(int sampleRate, audio_channel_mask_t channelMask) {
using namespace ::android::audio_utils::channels;
- size_t frames = 100;
- unsigned outChannels = 2;
+ size_t frames = 100; // set to an even number (2, 4, 6 ... ) stream alternates +1, -1.
+ constexpr unsigned outChannels = 2;
unsigned inChannels = audio_channel_count_from_out_mask(channelMask);
std::vector<float> input(frames * inChannels);
std::vector<float> output(frames * outChannels);
@@ -119,7 +144,7 @@
auto stats = channelStatistics(output, 2 /* channels */);
// printf("power: %s %s\n", stats[0].toString().c_str(), stats[1].toString().c_str());
- double power[2] = { stats[0].getVariance(), stats[1].getVariance() };
+ double power[2] = { stats[0].getPopVariance(), stats[1].getPopVariance() };
// Check symmetric power for pair channels on exchange of left/right position.
// to do this, we save previous power measurements.
@@ -139,20 +164,21 @@
EXPECT_EQ(0.f, power[1]);
continue;
}
- constexpr float POWER_TOLERANCE = 0.01; // for variance sum error.
+
+ constexpr float POWER_TOLERANCE = 0.001;
+ const float expectedPower = kScaleFromChannelIdx[index] * kScaleFromChannelIdx[index];
switch (side) {
case AUDIO_GEOMETRY_SIDE_LEFT:
- EXPECT_NEAR(0.25f, power[0], POWER_TOLERANCE);
- EXPECT_EQ(0.f, power[1]);
+ EXPECT_EQ(0.f, power[1]); // always true
+ EXPECT_NEAR(expectedPower, power[0], POWER_TOLERANCE);
break;
case AUDIO_GEOMETRY_SIDE_RIGHT:
- EXPECT_EQ(0.f, power[0]);
- EXPECT_NEAR(0.25f, power[1], POWER_TOLERANCE);
+ EXPECT_EQ(0.f, power[0]); // always true
+ EXPECT_NEAR(expectedPower, power[1], POWER_TOLERANCE);
break;
case AUDIO_GEOMETRY_SIDE_CENTER:
- EXPECT_NEAR(0.125f, power[0], POWER_TOLERANCE);
- EXPECT_NEAR(0.125f, power[1], POWER_TOLERANCE);
- EXPECT_NEAR_EPSILON(power[0], power[1]);
+ EXPECT_NEAR_EPSILON(power[0], power[1]); // always true
+ EXPECT_NEAR(expectedPower, power[0], POWER_TOLERANCE);
break;
}
}
@@ -244,10 +270,11 @@
::testing::Combine(
::testing::Range(0, (int)std::size(kSampleRates)),
::testing::Range(0, (int)std::size(kChannelPositionMasks))
- ));
-
-int main(int argc, /* const */ char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
- int status = RUN_ALL_TESTS();
- return status;
-}
+ ),
+ [](const testing::TestParamInfo<DownmixTest::ParamType>& info) {
+ const int index = std::get<1>(info.param);
+ const audio_channel_mask_t channelMask = kChannelPositionMasks[index];
+ const std::string name = std::string(audio_channel_out_mask_to_string(channelMask))
+ + "_" + std::to_string(std::get<0>(info.param)) + "_" + std::to_string(index);
+ return name;
+ });