AudioMixer.h revision 49c34acbef91800930b0fe561e0787145a10cfcc
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#include <media/nbaio/NBLog.h>
32
33namespace android {
34
35// ----------------------------------------------------------------------------
36
37class AudioMixer
38{
39public:
40                            AudioMixer(size_t frameCount, uint32_t sampleRate,
41                                       uint32_t maxNumTracks = MAX_NUM_TRACKS);
42
43    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
44
45
46    // This mixer has a hard-coded upper limit of 32 active track inputs.
47    // Adding support for > 32 tracks would require more than simply changing this value.
48    static const uint32_t MAX_NUM_TRACKS = 32;
49    // maximum number of channels supported by the mixer
50
51    // This mixer has a hard-coded upper limit of 2 channels for output.
52    // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
53    // Adding support for > 2 channel output would require more than simply changing this value.
54    static const uint32_t MAX_NUM_CHANNELS = 2;
55    // maximum number of channels supported for the content
56    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
57
58    static const uint16_t UNITY_GAIN = 0x1000;
59
60    enum { // names
61
62        // track names (MAX_NUM_TRACKS units)
63        TRACK0          = 0x1000,
64
65        // 0x2000 is unused
66
67        // setParameter targets
68        TRACK           = 0x3000,
69        RESAMPLE        = 0x3001,
70        RAMP_VOLUME     = 0x3002, // ramp to new volume
71        VOLUME          = 0x3003, // don't ramp
72
73        // set Parameter names
74        // for target TRACK
75        CHANNEL_MASK    = 0x4000,
76        FORMAT          = 0x4001,
77        MAIN_BUFFER     = 0x4002,
78        AUX_BUFFER      = 0x4003,
79        DOWNMIX_TYPE    = 0X4004,
80        // for target RESAMPLE
81        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
82                                  // parameter 'value' is the new sample rate in Hz.
83                                  // Only creates a sample rate converter the first time that
84                                  // the track sample rate is different from the mix sample rate.
85                                  // If the new sample rate is the same as the mix sample rate,
86                                  // and a sample rate converter already exists,
87                                  // then the sample rate converter remains present but is a no-op.
88        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
89                                  // This clears out the resampler's input buffer.
90        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
91                                  // the track is restored to the mix sample rate.
92        // for target RAMP_VOLUME and VOLUME (8 channels max)
93        VOLUME0         = 0x4200,
94        VOLUME1         = 0x4201,
95        AUXLEVEL        = 0x4210,
96    };
97
98
99    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
100
101    // Allocate a track name.  Returns new track name if successful, -1 on failure.
102    int         getTrackName(audio_channel_mask_t channelMask, int sessionId);
103
104    // Free an allocated track by name
105    void        deleteTrackName(int name);
106
107    // Enable or disable an allocated track by name
108    void        enable(int name);
109    void        disable(int name);
110
111    void        setParameter(int name, int target, int param, void *value);
112
113    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
114    void        process(int64_t pts);
115
116    uint32_t    trackNames() const { return mTrackNames; }
117
118    size_t      getUnreleasedFrames(int name) const;
119
120private:
121
122    enum {
123        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
124        NEEDS_FORMAT__MASK          = 0x000000F0,
125        NEEDS_MUTE__MASK            = 0x00000100,
126        NEEDS_RESAMPLE__MASK        = 0x00001000,
127        NEEDS_AUX__MASK             = 0x00010000,
128    };
129
130    enum {
131        NEEDS_CHANNEL_1             = 0x00000000,
132        NEEDS_CHANNEL_2             = 0x00000001,
133
134        NEEDS_FORMAT_16             = 0x00000010,
135
136        NEEDS_MUTE_DISABLED         = 0x00000000,
137        NEEDS_MUTE_ENABLED          = 0x00000100,
138
139        NEEDS_RESAMPLE_DISABLED     = 0x00000000,
140        NEEDS_RESAMPLE_ENABLED      = 0x00001000,
141
142        NEEDS_AUX_DISABLED     = 0x00000000,
143        NEEDS_AUX_ENABLED      = 0x00010000,
144    };
145
146    struct state_t;
147    struct track_t;
148    class DownmixerBufferProvider;
149
150    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
151                           int32_t* aux);
152    static const int BLOCKSIZE = 16; // 4 cache lines
153
154    struct track_t {
155        uint32_t    needs;
156
157        union {
158        int16_t     volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
159        int32_t     volumeRL;
160        };
161
162        int32_t     prevVolume[MAX_NUM_CHANNELS];
163
164        // 16-byte boundary
165
166        int32_t     volumeInc[MAX_NUM_CHANNELS];
167        int32_t     auxInc;
168        int32_t     prevAuxLevel;
169
170        // 16-byte boundary
171
172        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
173        uint16_t    frameCount;
174
175        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
176        uint8_t     format;         // always 16
177        uint16_t    enabled;        // actually bool
178        audio_channel_mask_t channelMask;
179
180        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
181        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
182        AudioBufferProvider*                bufferProvider;
183
184        // 16-byte boundary
185
186        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
187
188        hook_t      hook;
189        const void* in;             // current location in buffer
190
191        // 16-byte boundary
192
193        AudioResampler*     resampler;
194        uint32_t            sampleRate;
195        int32_t*           mainBuffer;
196        int32_t*           auxBuffer;
197
198        // 16-byte boundary
199
200        DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
201
202        int32_t     sessionId;
203
204        int32_t     padding[2];
205
206        // 16-byte boundary
207
208        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
209        bool        doesResample() const { return resampler != NULL; }
210        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
211        void        adjustVolumeRamp(bool aux);
212        size_t      getUnreleasedFrames() const { return resampler != NULL ?
213                                                    resampler->getUnreleasedFrames() : 0; };
214    };
215
216    // pad to 32-bytes to fill cache line
217    struct state_t {
218        uint32_t        enabledTracks;
219        uint32_t        needsChanged;
220        size_t          frameCount;
221        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL
222        int32_t         *outputTemp;
223        int32_t         *resampleTemp;
224        NBLog::Writer*  mLog;
225        int32_t         reserved[1];
226        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
227        track_t         tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
228    };
229
230    // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
231    class DownmixerBufferProvider : public AudioBufferProvider {
232    public:
233        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
234        virtual void releaseBuffer(Buffer* buffer);
235        DownmixerBufferProvider();
236        virtual ~DownmixerBufferProvider();
237
238        AudioBufferProvider* mTrackBufferProvider;
239        effect_handle_t    mDownmixHandle;
240        effect_config_t    mDownmixConfig;
241    };
242
243    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
244    uint32_t        mTrackNames;
245
246    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
247    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
248    const uint32_t  mConfiguredNames;
249
250    const uint32_t  mSampleRate;
251
252    NBLog::Writer   mDummyLog;
253public:
254    void            setLog(NBLog::Writer* log);
255private:
256    state_t         mState __attribute__((aligned(32)));
257
258    // effect descriptor for the downmixer used by the mixer
259    static effect_descriptor_t sDwnmFxDesc;
260    // indicates whether a downmix effect has been found and is usable by this mixer
261    static bool                sIsMultichannelCapable;
262
263    // Call after changing either the enabled status of a track, or parameters of an enabled track.
264    // OK to call more often than that, but unnecessary.
265    void invalidateState(uint32_t mask);
266
267    static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
268    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
269    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
270
271    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
272            int32_t* aux);
273    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
274    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
275            int32_t* aux);
276    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
277            int32_t* aux);
278    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
279            int32_t* aux);
280    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
281            int32_t* aux);
282
283    static void process__validate(state_t* state, int64_t pts);
284    static void process__nop(state_t* state, int64_t pts);
285    static void process__genericNoResampling(state_t* state, int64_t pts);
286    static void process__genericResampling(state_t* state, int64_t pts);
287    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
288                                                          int64_t pts);
289#if 0
290    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
291                                                           int64_t pts);
292#endif
293
294    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
295                                      int outputFrameIndex);
296
297    static uint64_t         sLocalTimeFreq;
298    static pthread_once_t   sOnceControl;
299    static void             sInitRoutine();
300};
301
302// ----------------------------------------------------------------------------
303}; // namespace android
304
305#endif // ANDROID_AUDIO_MIXER_H
306