1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18#ifndef ANDROID_AUDIO_MIXER_H
19#define ANDROID_AUDIO_MIXER_H
20
21#include <stdint.h>
22#include <sys/types.h>
23
24#include <utils/threads.h>
25
26#include <media/AudioBufferProvider.h>
27#include "AudioResampler.h"
28
29#include <audio_effects/effect_downmix.h>
30#include <system/audio.h>
31
32namespace android {
33
34// ----------------------------------------------------------------------------
35
36class AudioMixer
37{
38public:
39                            AudioMixer(size_t frameCount, uint32_t sampleRate,
40                                       uint32_t maxNumTracks = MAX_NUM_TRACKS);
41
42    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
43
44    static const uint32_t MAX_NUM_TRACKS = 32;
45    // maximum number of channels supported by the mixer
46    static const uint32_t MAX_NUM_CHANNELS = 2;
47    // maximum number of channels supported for the content
48    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
49
50    static const uint16_t UNITY_GAIN = 0x1000;
51
52    enum { // names
53
54        // track names (MAX_NUM_TRACKS units)
55        TRACK0          = 0x1000,
56
57        // 0x2000 is unused
58
59        // setParameter targets
60        TRACK           = 0x3000,
61        RESAMPLE        = 0x3001,
62        RAMP_VOLUME     = 0x3002, // ramp to new volume
63        VOLUME          = 0x3003, // don't ramp
64
65        // set Parameter names
66        // for target TRACK
67        CHANNEL_MASK    = 0x4000,
68        FORMAT          = 0x4001,
69        MAIN_BUFFER     = 0x4002,
70        AUX_BUFFER      = 0x4003,
71        DOWNMIX_TYPE    = 0X4004,
72        // for target RESAMPLE
73        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
74                                  // parameter 'value' is the new sample rate in Hz.
75                                  // Only creates a sample rate converter the first time that
76                                  // the track sample rate is different from the mix sample rate.
77                                  // If the new sample rate is the same as the mix sample rate,
78                                  // and a sample rate converter already exists,
79                                  // then the sample rate converter remains present but is a no-op.
80        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
81                                  // This clears out the resampler's input buffer.
82        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
83                                  // the track is restored to the mix sample rate.
84        // for target RAMP_VOLUME and VOLUME (8 channels max)
85        VOLUME0         = 0x4200,
86        VOLUME1         = 0x4201,
87        AUXLEVEL        = 0x4210,
88    };
89
90
91    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
92
93    // Allocate a track name.  Returns new track name if successful, -1 on failure.
94    int         getTrackName(audio_channel_mask_t channelMask, int sessionId);
95
96    // Free an allocated track by name
97    void        deleteTrackName(int name);
98
99    // Enable or disable an allocated track by name
100    void        enable(int name);
101    void        disable(int name);
102
103    void        setParameter(int name, int target, int param, void *value);
104
105    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
106    void        process(int64_t pts);
107
108    uint32_t    trackNames() const { return mTrackNames; }
109
110    size_t      getUnreleasedFrames(int name) const;
111
112private:
113
114    enum {
115        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
116        NEEDS_FORMAT__MASK          = 0x000000F0,
117        NEEDS_MUTE__MASK            = 0x00000100,
118        NEEDS_RESAMPLE__MASK        = 0x00001000,
119        NEEDS_AUX__MASK             = 0x00010000,
120    };
121
122    enum {
123        NEEDS_CHANNEL_1             = 0x00000000,
124        NEEDS_CHANNEL_2             = 0x00000001,
125
126        NEEDS_FORMAT_16             = 0x00000010,
127
128        NEEDS_MUTE_DISABLED         = 0x00000000,
129        NEEDS_MUTE_ENABLED          = 0x00000100,
130
131        NEEDS_RESAMPLE_DISABLED     = 0x00000000,
132        NEEDS_RESAMPLE_ENABLED      = 0x00001000,
133
134        NEEDS_AUX_DISABLED     = 0x00000000,
135        NEEDS_AUX_ENABLED      = 0x00010000,
136    };
137
138    struct state_t;
139    struct track_t;
140    class DownmixerBufferProvider;
141
142    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
143    static const int BLOCKSIZE = 16; // 4 cache lines
144
145    struct track_t {
146        uint32_t    needs;
147
148        union {
149        int16_t     volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
150        int32_t     volumeRL;
151        };
152
153        int32_t     prevVolume[MAX_NUM_CHANNELS];
154
155        // 16-byte boundary
156
157        int32_t     volumeInc[MAX_NUM_CHANNELS];
158        int32_t     auxInc;
159        int32_t     prevAuxLevel;
160
161        // 16-byte boundary
162
163        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
164        uint16_t    frameCount;
165
166        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
167        uint8_t     format;         // always 16
168        uint16_t    enabled;        // actually bool
169        audio_channel_mask_t channelMask;
170
171        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
172        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
173        AudioBufferProvider*                bufferProvider;
174
175        // 16-byte boundary
176
177        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
178
179        hook_t      hook;
180        const void* in;             // current location in buffer
181
182        // 16-byte boundary
183
184        AudioResampler*     resampler;
185        uint32_t            sampleRate;
186        int32_t*           mainBuffer;
187        int32_t*           auxBuffer;
188
189        // 16-byte boundary
190
191        uint64_t    localTimeFreq;
192
193        DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
194
195        int32_t     sessionId;
196
197        // 16-byte boundary
198
199        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
200        bool        doesResample() const { return resampler != NULL; }
201        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
202        void        adjustVolumeRamp(bool aux);
203        size_t      getUnreleasedFrames() const { return resampler != NULL ?
204                                                    resampler->getUnreleasedFrames() : 0; };
205    };
206
207    // pad to 32-bytes to fill cache line
208    struct state_t {
209        uint32_t        enabledTracks;
210        uint32_t        needsChanged;
211        size_t          frameCount;
212        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL
213        int32_t         *outputTemp;
214        int32_t         *resampleTemp;
215        int32_t         reserved[2];
216        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
217        track_t         tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
218    };
219
220    // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
221    class DownmixerBufferProvider : public AudioBufferProvider {
222    public:
223        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
224        virtual void releaseBuffer(Buffer* buffer);
225        DownmixerBufferProvider();
226        virtual ~DownmixerBufferProvider();
227
228        AudioBufferProvider* mTrackBufferProvider;
229        effect_handle_t    mDownmixHandle;
230        effect_config_t    mDownmixConfig;
231    };
232
233    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
234    uint32_t        mTrackNames;
235
236    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
237    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
238    const uint32_t  mConfiguredNames;
239
240    const uint32_t  mSampleRate;
241
242    state_t         mState __attribute__((aligned(32)));
243
244    // effect descriptor for the downmixer used by the mixer
245    static effect_descriptor_t dwnmFxDesc;
246    // indicates whether a downmix effect has been found and is usable by this mixer
247    static bool                isMultichannelCapable;
248
249    // Call after changing either the enabled status of a track, or parameters of an enabled track.
250    // OK to call more often than that, but unnecessary.
251    void invalidateState(uint32_t mask);
252
253    static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
254    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
255    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
256
257    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
258    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
259    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
260    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
261    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
262    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
263
264    static void process__validate(state_t* state, int64_t pts);
265    static void process__nop(state_t* state, int64_t pts);
266    static void process__genericNoResampling(state_t* state, int64_t pts);
267    static void process__genericResampling(state_t* state, int64_t pts);
268    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
269                                                          int64_t pts);
270#if 0
271    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
272                                                           int64_t pts);
273#endif
274
275    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
276                                      int outputFrameIndex);
277};
278
279// ----------------------------------------------------------------------------
280}; // namespace android
281
282#endif // ANDROID_AUDIO_MIXER_H
283