9d91ad5d99
Initial commit for review. Integrated comments after patch set 1 review. Fixed lockup in AudioFlinger::ThreadBase::exit() Fixed lockup when playing tone with AudioPlocyService startTone()
194 lines
5.9 KiB
C++
194 lines
5.9 KiB
C++
/* //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,
|
|
// for TARGET RESAMPLE
|
|
SAMPLE_RATE = 0x4100,
|
|
// for TARGET VOLUME (8 channels max)
|
|
VOLUME0 = 0x4200,
|
|
VOLUME1 = 0x4201,
|
|
};
|
|
|
|
|
|
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, int value);
|
|
|
|
status_t setBufferProvider(AudioBufferProvider* bufferProvider);
|
|
void process(void* output);
|
|
|
|
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,
|
|
};
|
|
|
|
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,
|
|
};
|
|
|
|
static inline int32_t applyVolume(int32_t in, int32_t v) {
|
|
return in * v;
|
|
}
|
|
|
|
|
|
struct state_t;
|
|
|
|
typedef void (*mix_t)(state_t* state, void* output);
|
|
|
|
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];
|
|
|
|
uint16_t frameCount;
|
|
|
|
uint8_t channelCount : 4;
|
|
uint8_t enabled : 1;
|
|
uint8_t reserved0 : 3;
|
|
uint8_t format;
|
|
|
|
AudioBufferProvider* bufferProvider;
|
|
mutable AudioBufferProvider::Buffer buffer;
|
|
|
|
void (*hook)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp);
|
|
void const* in; // current location in buffer
|
|
|
|
AudioResampler* resampler;
|
|
uint32_t sampleRate;
|
|
|
|
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
|
|
bool doesResample() const;
|
|
void adjustVolumeRamp();
|
|
};
|
|
|
|
// 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);
|
|
static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
|
|
static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp);
|
|
static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
|
|
static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
|
|
|
|
static void process__validate(state_t* state, void* output);
|
|
static void process__nop(state_t* state, void* output);
|
|
static void process__genericNoResampling(state_t* state, void* output);
|
|
static void process__genericResampling(state_t* state, void* output);
|
|
static void process__OneTrack16BitsStereoNoResampling(state_t* state, void* output);
|
|
static void process__TwoTracks16BitsStereoNoResampling(state_t* state, void* output);
|
|
};
|
|
|
|
// ----------------------------------------------------------------------------
|
|
}; // namespace android
|
|
|
|
#endif // ANDROID_AUDIO_MIXER_H
|