/*
**
** 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 <utils/threads.h>
#include <media/AudioBufferProvider.h>
#include "AudioResampler.h"
#include <audio_effects/effect_downmix.h>
#include <system/audio.h>
namespace android {
// ----------------------------------------------------------------------------
class AudioMixer
{
public:
AudioMixer(size_t frameCount, uint32_t sampleRate,
uint32_t maxNumTracks = MAX_NUM_TRACKS);
/*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
static const uint32_t MAX_NUM_TRACKS = 32;
// maximum number of channels supported by the mixer
static const uint32_t MAX_NUM_CHANNELS = 2;
// maximum number of channels supported for the content
static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
static const uint16_t UNITY_GAIN = 0x1000;
enum { // names
// track names (MAX_NUM_TRACKS units)
TRACK0 = 0x1000,
// 0x2000 is unused
// 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_MASK = 0x4000,
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
DOWNMIX_TYPE = 0X4004,
// for target RESAMPLE
SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
// parameter 'value' is the new sample rate in Hz.
// Only creates a sample rate converter the first time that
// the track sample rate is different from the mix sample rate.
// If the new sample rate is the same as the mix sample rate,
// and a sample rate converter already exists,
// then the sample rate converter remains present but is a no-op.
RESET = 0x4101, // Reset sample rate converter without changing sample rate.
// This clears out the resampler's input buffer.
REMOVE = 0x4102, // Remove the sample rate converter on this track name;
// the track is restored to the mix sample rate.
// for target RAMP_VOLUME and VOLUME (8 channels max)
VOLUME0 = 0x4200,
VOLUME1 = 0x4201,
AUXLEVEL = 0x4210,
};
// For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
// Allocate a track name. Returns new track name if successful, -1 on failure.
int getTrackName(audio_channel_mask_t channelMask, int sessionId);
// Free an allocated track by name
void deleteTrackName(int name);
// Enable or disable an allocated track by name
void enable(int name);
void disable(int name);
void setParameter(int name, int target, int param, void *value);
void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
void process(int64_t pts);
uint32_t trackNames() const { return mTrackNames; }
size_t getUnreleasedFrames(int name) const;
private:
enum {
NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
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,
};
struct state_t;
struct track_t;
class DownmixerBufferProvider;
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[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
int32_t volumeRL;
};
int32_t prevVolume[MAX_NUM_CHANNELS];
// 16-byte boundary
int32_t volumeInc[MAX_NUM_CHANNELS];
int32_t auxInc;
int32_t prevAuxLevel;
// 16-byte boundary
int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
uint16_t frameCount;
uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
uint8_t format; // always 16
uint16_t enabled; // actually bool
audio_channel_mask_t channelMask;
// actual buffer provider used by the track hooks, see DownmixerBufferProvider below
// for how the Track buffer provider is wrapped by another one when dowmixing is required
AudioBufferProvider* bufferProvider;
// 16-byte boundary
mutable AudioBufferProvider::Buffer buffer; // 8 bytes
hook_t hook;
const void* in; // current location in buffer
// 16-byte boundary
AudioResampler* resampler;
uint32_t sampleRate;
int32_t* mainBuffer;
int32_t* auxBuffer;
// 16-byte boundary
uint64_t localTimeFreq;
DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
int32_t sessionId;
// 16-byte boundary
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
bool doesResample() const { return resampler != NULL; }
void resetResampler() { if (resampler != NULL) resampler->reset(); }
void adjustVolumeRamp(bool aux);
size_t getUnreleasedFrames() const { return resampler != NULL ?
resampler->getUnreleasedFrames() : 0; };
};
// pad to 32-bytes to fill cache line
struct state_t {
uint32_t enabledTracks;
uint32_t needsChanged;
size_t frameCount;
void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL
int32_t *outputTemp;
int32_t *resampleTemp;
int32_t reserved[2];
// FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
};
// AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
class DownmixerBufferProvider : public AudioBufferProvider {
public:
virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
virtual void releaseBuffer(Buffer* buffer);
DownmixerBufferProvider();
virtual ~DownmixerBufferProvider();
AudioBufferProvider* mTrackBufferProvider;
effect_handle_t mDownmixHandle;
effect_config_t mDownmixConfig;
};
// bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
uint32_t mTrackNames;
// bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
// but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
const uint32_t mConfiguredNames;
const uint32_t mSampleRate;
state_t mState __attribute__((aligned(32)));
// effect descriptor for the downmixer used by the mixer
static effect_descriptor_t dwnmFxDesc;
// indicates whether a downmix effect has been found and is usable by this mixer
static bool isMultichannelCapable;
// Call after changing either the enabled status of a track, or parameters of an enabled track.
// OK to call more often than that, but unnecessary.
void invalidateState(uint32_t mask);
static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
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, int64_t pts);
static void process__nop(state_t* state, int64_t pts);
static void process__genericNoResampling(state_t* state, int64_t pts);
static void process__genericResampling(state_t* state, int64_t pts);
static void process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts);
#if 0
static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
int64_t pts);
#endif
static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
int outputFrameIndex);
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif // ANDROID_AUDIO_MIXER_H