Remove symlinks from include dir
Test: build
Change-Id: Ibe4eee4fe49b7884e6d720e626d88125bbee0eb2
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 81291a1..93a336a 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -73,7 +73,7 @@
local_include_dirs: ["aidl"],
export_aidl_headers: true,
},
-
+
local_include_dirs: [
"include",
],
diff --git a/media/libmedia/include/media/RecordBufferConverter.h b/media/libmedia/include/media/RecordBufferConverter.h
deleted file mode 100644
index 2abc45e..0000000
--- a/media/libmedia/include/media/RecordBufferConverter.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (C) 2017 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 ANDROID_RECORD_BUFFER_CONVERTER_H
-#define ANDROID_RECORD_BUFFER_CONVERTER_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <media/AudioBufferProvider.h>
-#include <system/audio.h>
-
-class AudioResampler;
-class PassthruBufferProvider;
-
-namespace android {
-
-/* The RecordBufferConverter is used for format, channel, and sample rate
- * conversion for a RecordTrack.
- *
- * RecordBufferConverter uses the convert() method rather than exposing a
- * buffer provider interface; this is to save a memory copy.
- *
- * There are legacy conversion requirements for this converter, specifically
- * due to mono handling, so be careful about modifying.
- *
- * Original source audioflinger/Threads.{h,cpp}
- */
-class RecordBufferConverter
-{
-public:
- RecordBufferConverter(
- audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
- uint32_t srcSampleRate,
- audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
- uint32_t dstSampleRate);
-
- ~RecordBufferConverter();
-
- /* Converts input data from an AudioBufferProvider by format, channelMask,
- * and sampleRate to a destination buffer.
- *
- * Parameters
- * dst: buffer to place the converted data.
- * provider: buffer provider to obtain source data.
- * frames: number of frames to convert
- *
- * Returns the number of frames converted.
- */
- size_t convert(void *dst, AudioBufferProvider *provider, size_t frames);
-
- // returns NO_ERROR if constructor was successful
- status_t initCheck() const {
- // mSrcChannelMask set on successful updateParameters
- return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT;
- }
-
- // allows dynamic reconfigure of all parameters
- status_t updateParameters(
- audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
- uint32_t srcSampleRate,
- audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
- uint32_t dstSampleRate);
-
- // called to reset resampler buffers on record track discontinuity
- void reset();
-
-private:
- // format conversion when not using resampler
- void convertNoResampler(void *dst, const void *src, size_t frames);
-
- // format conversion when using resampler; modifies src in-place
- void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames);
-
- // user provided information
- audio_channel_mask_t mSrcChannelMask;
- audio_format_t mSrcFormat;
- uint32_t mSrcSampleRate;
- audio_channel_mask_t mDstChannelMask;
- audio_format_t mDstFormat;
- uint32_t mDstSampleRate;
-
- // derived information
- uint32_t mSrcChannelCount;
- uint32_t mDstChannelCount;
- size_t mDstFrameSize;
-
- // format conversion buffer
- void *mBuf;
- size_t mBufFrames;
- size_t mBufFrameSize;
-
- // resampler info
- AudioResampler *mResampler;
-
- bool mIsLegacyDownmix; // legacy stereo to mono conversion needed
- bool mIsLegacyUpmix; // legacy mono to stereo conversion needed
- bool mRequiresFloat; // data processing requires float (e.g. resampler)
- PassthruBufferProvider *mInputConverterProvider; // converts input to float
- int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion
-};
-
-// ----------------------------------------------------------------------------
-} // namespace android
-
-#endif // ANDROID_RECORD_BUFFER_CONVERTER_H