AudioMixer.h revision e8a1ced4da17dc6c07803dc2af8060f62a8389c1 
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        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
81        // for target RESAMPLE
82        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
83                                  // parameter 'value' is the new sample rate in Hz.
84                                  // Only creates a sample rate converter the first time that
85                                  // the track sample rate is different from the mix sample rate.
86                                  // If the new sample rate is the same as the mix sample rate,
87                                  // and a sample rate converter already exists,
88                                  // then the sample rate converter remains present but is a no-op.
89        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
90                                  // This clears out the resampler's input buffer.
91        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
92                                  // the track is restored to the mix sample rate.
93        // for target RAMP_VOLUME and VOLUME (8 channels max)
94        VOLUME0         = 0x4200,
95        VOLUME1         = 0x4201,
96        AUXLEVEL        = 0x4210,
97    };
98
99
100    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
101
102    // Allocate a track name.  Returns new track name if successful, -1 on failure.
103    // The failure could be because of an invalid channelMask or format, or that
104    // the track capacity of the mixer is exceeded.
105    int         getTrackName(audio_channel_mask_t channelMask,
106                             audio_format_t format, int sessionId);
107
108    // Free an allocated track by name
109    void        deleteTrackName(int name);
110
111    // Enable or disable an allocated track by name
112    void        enable(int name);
113    void        disable(int name);
114
115    void        setParameter(int name, int target, int param, void *value);
116
117    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
118    void        process(int64_t pts);
119
120    uint32_t    trackNames() const { return mTrackNames; }
121
122    size_t      getUnreleasedFrames(int name) const;
123
124    static inline bool isValidPcmTrackFormat(audio_format_t format) {
125        return format == AUDIO_FORMAT_PCM_16_BIT;
126    }
127
128private:
129
130    enum {
131        // FIXME this representation permits up to 8 channels
132        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
133    };
134
135    enum {
136        NEEDS_CHANNEL_1             = 0x00000000,   // mono
137        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
138
139        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
140
141        NEEDS_MUTE                  = 0x00000100,
142        NEEDS_RESAMPLE              = 0x00001000,
143        NEEDS_AUX                   = 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        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
205        audio_format_t mFormat;          // input track format
206
207        // 16-byte boundary
208
209        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
210        bool        doesResample() const { return resampler != NULL; }
211        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
212        void        adjustVolumeRamp(bool aux);
213        size_t      getUnreleasedFrames() const { return resampler != NULL ?
214                                                    resampler->getUnreleasedFrames() : 0; };
215    };
216
217    // pad to 32-bytes to fill cache line
218    struct state_t {
219        uint32_t        enabledTracks;
220        uint32_t        needsChanged;
221        size_t          frameCount;
222        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL
223        int32_t         *outputTemp;
224        int32_t         *resampleTemp;
225        NBLog::Writer*  mLog;
226        int32_t         reserved[1];
227        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
228        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
229    };
230
231    // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
232    class DownmixerBufferProvider : public AudioBufferProvider {
233    public:
234        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
235        virtual void releaseBuffer(Buffer* buffer);
236        DownmixerBufferProvider();
237        virtual ~DownmixerBufferProvider();
238
239        AudioBufferProvider* mTrackBufferProvider;
240        effect_handle_t    mDownmixHandle;
241        effect_config_t    mDownmixConfig;
242    };
243
244    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
245    uint32_t        mTrackNames;
246
247    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
248    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
249    const uint32_t  mConfiguredNames;
250
251    const uint32_t  mSampleRate;
252
253    NBLog::Writer   mDummyLog;
254public:
255    void            setLog(NBLog::Writer* log);
256private:
257    state_t         mState __attribute__((aligned(32)));
258
259    // effect descriptor for the downmixer used by the mixer
260    static effect_descriptor_t sDwnmFxDesc;
261    // indicates whether a downmix effect has been found and is usable by this mixer
262    static bool                sIsMultichannelCapable;
263
264    // Call after changing either the enabled status of a track, or parameters of an enabled track.
265    // OK to call more often than that, but unnecessary.
266    void invalidateState(uint32_t mask);
267
268    static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
269    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
270    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
271
272    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
273            int32_t* aux);
274    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
275    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
276            int32_t* aux);
277    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
278            int32_t* aux);
279    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
280            int32_t* aux);
281    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
282            int32_t* aux);
283
284    static void process__validate(state_t* state, int64_t pts);
285    static void process__nop(state_t* state, int64_t pts);
286    static void process__genericNoResampling(state_t* state, int64_t pts);
287    static void process__genericResampling(state_t* state, int64_t pts);
288    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
289                                                          int64_t pts);
290#if 0
291    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
292                                                           int64_t pts);
293#endif
294
295    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
296                                      int outputFrameIndex);
297
298    static uint64_t         sLocalTimeFreq;
299    static pthread_once_t   sOnceControl;
300    static void             sInitRoutine();
301};
302
303// ----------------------------------------------------------------------------
304}; // namespace android
305
306#endif // ANDROID_AUDIO_MIXER_H
307