move native services under services/
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
new file mode 100644
index 0000000..aee3e17
--- /dev/null
+++ b/services/audioflinger/AudioMixer.h
@@ -0,0 +1,207 @@
+/* //device/include/server/AudioFlinger/AudioMixer.h
+**
+** Copyright 2007, 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_AUDIO_MIXER_H
+#define ANDROID_AUDIO_MIXER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include "AudioBufferProvider.h"
+#include "AudioResampler.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
+#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+
+// ----------------------------------------------------------------------------
+
+class AudioMixer
+{
+public:
+ AudioMixer(size_t frameCount, uint32_t sampleRate);
+
+ ~AudioMixer();
+
+ static const uint32_t MAX_NUM_TRACKS = 32;
+ static const uint32_t MAX_NUM_CHANNELS = 2;
+
+ static const uint16_t UNITY_GAIN = 0x1000;
+
+ enum { // names
+
+ // track units (32 units)
+ TRACK0 = 0x1000,
+
+ // enable/disable
+ MIXING = 0x2000,
+
+ // setParameter targets
+ TRACK = 0x3000,
+ RESAMPLE = 0x3001,
+ RAMP_VOLUME = 0x3002, // ramp to new volume
+ VOLUME = 0x3003, // don't ramp
+
+ // set Parameter names
+ // for target TRACK
+ CHANNEL_COUNT = 0x4000,
+ FORMAT = 0x4001,
+ MAIN_BUFFER = 0x4002,
+ AUX_BUFFER = 0x4003,
+ // for TARGET RESAMPLE
+ SAMPLE_RATE = 0x4100,
+ // for TARGET VOLUME (8 channels max)
+ VOLUME0 = 0x4200,
+ VOLUME1 = 0x4201,
+ AUXLEVEL = 0x4210,
+ };
+
+
+ int getTrackName();
+ void deleteTrackName(int name);
+
+ status_t enable(int name);
+ status_t disable(int name);
+
+ status_t setActiveTrack(int track);
+ status_t setParameter(int target, int name, void *value);
+
+ status_t setBufferProvider(AudioBufferProvider* bufferProvider);
+ void process();
+
+ uint32_t trackNames() const { return mTrackNames; }
+
+ static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
+
+private:
+
+ enum {
+ NEEDS_CHANNEL_COUNT__MASK = 0x00000003,
+ NEEDS_FORMAT__MASK = 0x000000F0,
+ NEEDS_MUTE__MASK = 0x00000100,
+ NEEDS_RESAMPLE__MASK = 0x00001000,
+ NEEDS_AUX__MASK = 0x00010000,
+ };
+
+ enum {
+ NEEDS_CHANNEL_1 = 0x00000000,
+ NEEDS_CHANNEL_2 = 0x00000001,
+
+ NEEDS_FORMAT_16 = 0x00000010,
+
+ NEEDS_MUTE_DISABLED = 0x00000000,
+ NEEDS_MUTE_ENABLED = 0x00000100,
+
+ NEEDS_RESAMPLE_DISABLED = 0x00000000,
+ NEEDS_RESAMPLE_ENABLED = 0x00001000,
+
+ NEEDS_AUX_DISABLED = 0x00000000,
+ NEEDS_AUX_ENABLED = 0x00010000,
+ };
+
+ static inline int32_t applyVolume(int32_t in, int32_t v) {
+ return in * v;
+ }
+
+
+ struct state_t;
+ struct track_t;
+
+ typedef void (*mix_t)(state_t* state);
+ typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
+ static const int BLOCKSIZE = 16; // 4 cache lines
+
+ struct track_t {
+ uint32_t needs;
+
+ union {
+ int16_t volume[2]; // [0]3.12 fixed point
+ int32_t volumeRL;
+ };
+
+ int32_t prevVolume[2];
+
+ int32_t volumeInc[2];
+ int32_t auxLevel;
+ int32_t auxInc;
+ int32_t prevAuxLevel;
+
+ uint16_t frameCount;
+
+ uint8_t channelCount : 4;
+ uint8_t enabled : 1;
+ uint8_t reserved0 : 3;
+ uint8_t format;
+
+ AudioBufferProvider* bufferProvider;
+ mutable AudioBufferProvider::Buffer buffer;
+
+ hook_t hook;
+ void const* in; // current location in buffer
+
+ AudioResampler* resampler;
+ uint32_t sampleRate;
+ int32_t* mainBuffer;
+ int32_t* auxBuffer;
+
+ bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
+ bool doesResample() const;
+ void adjustVolumeRamp(bool aux);
+ };
+
+ // pad to 32-bytes to fill cache line
+ struct state_t {
+ uint32_t enabledTracks;
+ uint32_t needsChanged;
+ size_t frameCount;
+ mix_t hook;
+ int32_t *outputTemp;
+ int32_t *resampleTemp;
+ int32_t reserved[2];
+ track_t tracks[32]; __attribute__((aligned(32)));
+ };
+
+ int mActiveTrack;
+ uint32_t mTrackNames;
+ const uint32_t mSampleRate;
+
+ state_t mState __attribute__((aligned(32)));
+
+ void invalidateState(uint32_t mask);
+
+ static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+ static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+
+ static void process__validate(state_t* state);
+ static void process__nop(state_t* state);
+ static void process__genericNoResampling(state_t* state);
+ static void process__genericResampling(state_t* state);
+ static void process__OneTrack16BitsStereoNoResampling(state_t* state);
+ static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H