Audio: Support FCC_LIMIT for 22.2 testing
Test: Basic audio checks
Bug: 189325443
Change-Id: Ie49f077e5deb7dcbc6acc32502d9ce82bcb85c04
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index 0d60120..acfac24 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -25,8 +25,7 @@
// TODO These defines should be moved to a central place in audio.
#define SAMPLES_PER_FRAME_MIN 1
-// TODO Remove 8 channel limitation.
-#define SAMPLES_PER_FRAME_MAX FCC_8
+#define SAMPLES_PER_FRAME_MAX FCC_LIMIT
#define SAMPLE_RATE_HZ_MIN 8000
// HDMI supports up to 32 channels at 1536000 Hz.
#define SAMPLE_RATE_HZ_MAX 1600000
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 2135c54..e015592 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -43,8 +43,7 @@
// on the edge of being ridiculous.
// TODO These defines should be moved to a central place in audio.
#define SAMPLES_PER_FRAME_MIN 1
-// TODO Remove 8 channel limitation.
-#define SAMPLES_PER_FRAME_MAX FCC_8
+#define SAMPLES_PER_FRAME_MAX FCC_LIMIT
#define SAMPLE_RATE_HZ_MIN 8000
// HDMI supports up to 32 channels at 1536000 Hz.
#define SAMPLE_RATE_HZ_MAX 1600000
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index 8d374c9..a56d9cb 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -17,6 +17,8 @@
#ifndef ANDROID_AUDIO_MIXER_OPS_H
#define ANDROID_AUDIO_MIXER_OPS_H
+#include <system/audio.h>
+
namespace android {
// Hack to make static_assert work in a constexpr
@@ -231,7 +233,7 @@
typename TO, typename TI, typename TV,
typename F>
void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
- static_assert(NCHAN > 0 && NCHAN <= 8);
+ static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
|| MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
|| MIXTYPE == MIXTYPE_STEREOEXPAND
diff --git a/media/libaudioprocessing/AudioResampler.cpp b/media/libaudioprocessing/AudioResampler.cpp
index c761b38..51673d7 100644
--- a/media/libaudioprocessing/AudioResampler.cpp
+++ b/media/libaudioprocessing/AudioResampler.cpp
@@ -268,7 +268,7 @@
mPhaseFraction(0),
mQuality(quality) {
- const int maxChannels = quality < DYN_LOW_QUALITY ? 2 : 8;
+ const int maxChannels = quality < DYN_LOW_QUALITY ? FCC_2 : FCC_LIMIT;
if (inChannelCount < 1
|| inChannelCount > maxChannels) {
LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels",
diff --git a/media/libaudioprocessing/AudioResamplerDyn.cpp b/media/libaudioprocessing/AudioResamplerDyn.cpp
index 1aacfd1..21d3d36 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.cpp
+++ b/media/libaudioprocessing/AudioResamplerDyn.cpp
@@ -545,8 +545,8 @@
// Note: A stride of 2 is achieved with non-SIMD processing.
int stride = ((c.mHalfNumCoefs & 7) == 0) ? 16 : 2;
LOG_ALWAYS_FATAL_IF(stride < 16, "Resampler stride must be 16 or more");
- LOG_ALWAYS_FATAL_IF(mChannelCount < 1 || mChannelCount > 8,
- "Resampler channels(%d) must be between 1 to 8", mChannelCount);
+ 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)
if (locked) {
switch (mChannelCount) {
diff --git a/media/libaudioprocessing/include/media/AudioMixerBase.h b/media/libaudioprocessing/include/media/AudioMixerBase.h
index cf84b83..3419816 100644
--- a/media/libaudioprocessing/include/media/AudioMixerBase.h
+++ b/media/libaudioprocessing/include/media/AudioMixerBase.h
@@ -45,8 +45,7 @@
{
public:
// Do not change these unless underlying code changes.
- // This mixer has a hard-coded upper limit of 8 channels for output.
- static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
+ static constexpr uint32_t MAX_NUM_CHANNELS = FCC_LIMIT;
static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
static const uint16_t UNITY_GAIN_INT = 0x1000;
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index f838892..1551e33 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -157,7 +157,7 @@
if (pConfig->inputCfg.format != pConfig->outputCfg.format) return -EINVAL;
const uint32_t channelCount = audio_channel_count_from_out_mask(pConfig->inputCfg.channels);
#ifdef SUPPORT_MC
- if (channelCount < 1 || channelCount > FCC_8) return -EINVAL;
+ if (channelCount < 1 || channelCount > FCC_LIMIT) return -EINVAL;
#else
if (channelCount != FCC_2) return -EINVAL;
#endif
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index a89088a..e325dfa 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2896,8 +2896,8 @@
audio_is_linear_pcm(config->format) &&
audio_is_linear_pcm(halconfig.format) &&
(halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
- (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) &&
- (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) {
+ (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_LIMIT) &&
+ (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT)) {
// FIXME describe the change proposed by HAL (save old values so we can log them here)
ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
inStream.clear();
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index d6d6e25..2963202 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -107,7 +107,7 @@
mSampleRate = Format_sampleRate(mFormat);
#if !LOG_NDEBUG
unsigned channelCount = Format_channelCount(mFormat);
- ALOG_ASSERT(channelCount >= 1 && channelCount <= FCC_8);
+ ALOG_ASSERT(channelCount >= 1 && channelCount <= FCC_LIMIT);
#endif
}
dumpState->mSampleRate = mSampleRate;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 47b4b18..5e2b4e7 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -8573,7 +8573,7 @@
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
audio_channel_mask_t mask = (audio_channel_mask_t) value;
if (!audio_is_input_channel(mask) ||
- audio_channel_count_from_in_mask(mask) > FCC_8) {
+ audio_channel_count_from_in_mask(mask) > FCC_LIMIT) {
status = BAD_VALUE;
} else {
channelMask = mask;
@@ -8610,7 +8610,7 @@
if (mInput->stream->getAudioProperties(&config) == OK &&
audio_is_linear_pcm(config.format) && audio_is_linear_pcm(reqFormat) &&
config.sample_rate <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate) &&
- audio_channel_count_from_in_mask(config.channel_mask) <= FCC_8) {
+ audio_channel_count_from_in_mask(config.channel_mask) <= FCC_LIMIT) {
status = NO_ERROR;
}
}
@@ -8672,10 +8672,10 @@
mFormat = mHALFormat;
mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
if (audio_is_linear_pcm(mFormat)) {
- LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_8, "HAL channel count %d > %d",
- mChannelCount, FCC_8);
+ LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_LIMIT, "HAL channel count %d > %d",
+ mChannelCount, FCC_LIMIT);
} else {
- // Can have more that FCC_8 channels in encoded streams.
+ // Can have more that FCC_LIMIT channels in encoded streams.
ALOGI("HAL format %#x is not linear pcm", mFormat);
}
result = mInput->stream->getFrameSize(&mFrameSize);
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 552919d..577f641 100644
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -42,7 +42,7 @@
// For mixed output and inputs, the policy will use max mixer channel count.
// Do not limit channel count otherwise
-#define MAX_MIXER_CHANNEL_COUNT FCC_8
+#define MAX_MIXER_CHANNEL_COUNT FCC_LIMIT
/**
* Alias to AUDIO_DEVICE_OUT_DEFAULT defined for clarification when this value is used by volume