4f22c05eac
uint16_t enabled is (mostly) changed to bool in a separate CL Change-Id: Ied9f8c034b2479cee9a8778cee7b8ff92ae75b7b
224 lines
7.4 KiB
C++
224 lines
7.4 KiB
C++
/*
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**
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** Copyright 2007, The Android Open Source Project
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_MIXER_H
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#define ANDROID_AUDIO_MIXER_H
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#include <stdint.h>
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#include <sys/types.h>
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#include "AudioBufferProvider.h"
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#include "AudioResampler.h"
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namespace android {
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// ----------------------------------------------------------------------------
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class AudioMixer
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{
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public:
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AudioMixer(size_t frameCount, uint32_t sampleRate);
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/*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
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static const uint32_t MAX_NUM_TRACKS = 32;
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static const uint32_t MAX_NUM_CHANNELS = 2;
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static const uint16_t UNITY_GAIN = 0x1000;
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enum { // names
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// track names (MAX_NUM_TRACKS units)
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TRACK0 = 0x1000,
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// 0x2000 is unused
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// setParameter targets
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TRACK = 0x3000,
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RESAMPLE = 0x3001,
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RAMP_VOLUME = 0x3002, // ramp to new volume
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VOLUME = 0x3003, // don't ramp
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// set Parameter names
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// for target TRACK
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CHANNEL_MASK = 0x4000,
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FORMAT = 0x4001,
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MAIN_BUFFER = 0x4002,
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AUX_BUFFER = 0x4003,
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// for target RESAMPLE
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SAMPLE_RATE = 0x4100,
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RESET = 0x4101,
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// for target RAMP_VOLUME and VOLUME (8 channels max)
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VOLUME0 = 0x4200,
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VOLUME1 = 0x4201,
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AUXLEVEL = 0x4210,
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};
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// For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
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int getTrackName();
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void deleteTrackName(int name);
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void enable(int name);
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void disable(int name);
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void setParameter(int name, int target, int param, void *value);
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void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
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void process(int64_t pts);
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uint32_t trackNames() const { return mTrackNames; }
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size_t getUnreleasedFrames(int name) const;
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private:
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enum {
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NEEDS_CHANNEL_COUNT__MASK = 0x00000003,
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NEEDS_FORMAT__MASK = 0x000000F0,
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NEEDS_MUTE__MASK = 0x00000100,
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NEEDS_RESAMPLE__MASK = 0x00001000,
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NEEDS_AUX__MASK = 0x00010000,
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};
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enum {
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NEEDS_CHANNEL_1 = 0x00000000,
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NEEDS_CHANNEL_2 = 0x00000001,
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NEEDS_FORMAT_16 = 0x00000010,
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NEEDS_MUTE_DISABLED = 0x00000000,
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NEEDS_MUTE_ENABLED = 0x00000100,
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NEEDS_RESAMPLE_DISABLED = 0x00000000,
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NEEDS_RESAMPLE_ENABLED = 0x00001000,
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NEEDS_AUX_DISABLED = 0x00000000,
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NEEDS_AUX_ENABLED = 0x00010000,
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};
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struct state_t;
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struct track_t;
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typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
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static const int BLOCKSIZE = 16; // 4 cache lines
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struct track_t {
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uint32_t needs;
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union {
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int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
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int32_t volumeRL;
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};
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int32_t prevVolume[MAX_NUM_CHANNELS];
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// 16-byte boundary
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int32_t volumeInc[MAX_NUM_CHANNELS];
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int32_t auxInc;
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int32_t prevAuxLevel;
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// 16-byte boundary
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int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
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uint16_t frameCount;
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uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
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uint8_t format; // always 16
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uint16_t enabled; // actually bool
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uint32_t channelMask; // currently under-used
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AudioBufferProvider* bufferProvider;
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// 16-byte boundary
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mutable AudioBufferProvider::Buffer buffer; // 8 bytes
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hook_t hook;
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const void* in; // current location in buffer
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// 16-byte boundary
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AudioResampler* resampler;
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uint32_t sampleRate;
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int32_t* mainBuffer;
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int32_t* auxBuffer;
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// 16-byte boundary
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uint64_t localTimeFreq;
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int64_t padding;
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// 16-byte boundary
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bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
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bool doesResample() const { return resampler != NULL; }
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void resetResampler() { if (resampler != NULL) resampler->reset(); }
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void adjustVolumeRamp(bool aux);
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size_t getUnreleasedFrames() const { return resampler != NULL ?
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resampler->getUnreleasedFrames() : 0; };
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};
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// pad to 32-bytes to fill cache line
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struct state_t {
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uint32_t enabledTracks;
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uint32_t needsChanged;
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size_t frameCount;
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void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL
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int32_t *outputTemp;
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int32_t *resampleTemp;
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int32_t reserved[2];
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track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
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};
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// bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
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uint32_t mTrackNames;
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const uint32_t mSampleRate;
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state_t mState __attribute__((aligned(32)));
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void invalidateState(uint32_t mask);
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static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
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static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
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static void process__validate(state_t* state, int64_t pts);
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static void process__nop(state_t* state, int64_t pts);
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static void process__genericNoResampling(state_t* state, int64_t pts);
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static void process__genericResampling(state_t* state, int64_t pts);
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static void process__OneTrack16BitsStereoNoResampling(state_t* state,
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int64_t pts);
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#if 0
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static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
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int64_t pts);
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#endif
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static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
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int outputFrameIndex);
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};
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// ----------------------------------------------------------------------------
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}; // namespace android
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#endif // ANDROID_AUDIO_MIXER_H
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