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 8e0de7e..54a6425 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 8ab72a9..f62082e 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