AudioMixer.h revision ef7c7fbd0e3fb36af14cd7d39f64c949031516a5 
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                format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
127                format == AUDIO_FORMAT_PCM_32_BIT ||
128                format == AUDIO_FORMAT_PCM_FLOAT;
129    }
130
131private:
132
133    enum {
134        // FIXME this representation permits up to 8 channels
135        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
136    };
137
138    enum {
139        NEEDS_CHANNEL_1             = 0x00000000,   // mono
140        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
141
142        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
143
144        NEEDS_MUTE                  = 0x00000100,
145        NEEDS_RESAMPLE              = 0x00001000,
146        NEEDS_AUX                   = 0x00010000,
147    };
148
149    struct state_t;
150    struct track_t;
151    class DownmixerBufferProvider;
152    class ReformatBufferProvider;
153
154    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
155                           int32_t* aux);
156    static const int BLOCKSIZE = 16; // 4 cache lines
157
158    struct track_t {
159        uint32_t    needs;
160
161        union {
162        int16_t     volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
163        int32_t     volumeRL;
164        };
165
166        int32_t     prevVolume[MAX_NUM_CHANNELS];
167
168        // 16-byte boundary
169
170        int32_t     volumeInc[MAX_NUM_CHANNELS];
171        int32_t     auxInc;
172        int32_t     prevAuxLevel;
173
174        // 16-byte boundary
175
176        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
177        uint16_t    frameCount;
178
179        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
180        uint8_t     unused_padding; // formerly format, was always 16
181        uint16_t    enabled;        // actually bool
182        audio_channel_mask_t channelMask;
183
184        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
185        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
186        AudioBufferProvider*                bufferProvider;
187
188        // 16-byte boundary
189
190        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
191
192        hook_t      hook;
193        const void* in;             // current location in buffer
194
195        // 16-byte boundary
196
197        AudioResampler*     resampler;
198        uint32_t            sampleRate;
199        int32_t*           mainBuffer;
200        int32_t*           auxBuffer;
201
202        // 16-byte boundary
203        AudioBufferProvider*     mInputBufferProvider;    // 4 bytes
204        ReformatBufferProvider*  mReformatBufferProvider; // 4 bytes
205        DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
206
207        int32_t     sessionId;
208
209        // 16-byte boundary
210        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
211        audio_format_t mFormat;          // input track format
212        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
213                                         // each track must be converted to this format.
214
215        int32_t        mUnused[1];       // alignment padding
216
217        // 16-byte boundary
218
219        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
220        bool        doesResample() const { return resampler != NULL; }
221        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
222        void        adjustVolumeRamp(bool aux);
223        size_t      getUnreleasedFrames() const { return resampler != NULL ?
224                                                    resampler->getUnreleasedFrames() : 0; };
225    };
226
227    // pad to 32-bytes to fill cache line
228    struct state_t {
229        uint32_t        enabledTracks;
230        uint32_t        needsChanged;
231        size_t          frameCount;
232        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL
233        int32_t         *outputTemp;
234        int32_t         *resampleTemp;
235        NBLog::Writer*  mLog;
236        int32_t         reserved[1];
237        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
238        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
239    };
240
241    // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
242    class DownmixerBufferProvider : public AudioBufferProvider {
243    public:
244        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
245        virtual void releaseBuffer(Buffer* buffer);
246        DownmixerBufferProvider();
247        virtual ~DownmixerBufferProvider();
248
249        AudioBufferProvider* mTrackBufferProvider;
250        effect_handle_t    mDownmixHandle;
251        effect_config_t    mDownmixConfig;
252    };
253
254    // AudioBufferProvider wrapper that reformats track to acceptable mixer input type
255    class ReformatBufferProvider : public AudioBufferProvider {
256    public:
257        ReformatBufferProvider(int32_t channels,
258                audio_format_t inputFormat, audio_format_t outputFormat);
259        virtual ~ReformatBufferProvider();
260
261        // overrides AudioBufferProvider methods
262        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
263        virtual void releaseBuffer(Buffer* buffer);
264
265        void reset();
266        inline bool requiresInternalBuffers() {
267            return true; //mInputFrameSize < mOutputFrameSize;
268        }
269
270        AudioBufferProvider* mTrackBufferProvider;
271        int32_t              mChannels;
272        audio_format_t       mInputFormat;
273        audio_format_t       mOutputFormat;
274        size_t               mInputFrameSize;
275        size_t               mOutputFrameSize;
276        // (only) required for reformatting to a larger size.
277        AudioBufferProvider::Buffer mBuffer;
278        void*                mOutputData;
279        size_t               mOutputCount;
280        size_t               mConsumed;
281    };
282
283    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
284    uint32_t        mTrackNames;
285
286    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
287    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
288    const uint32_t  mConfiguredNames;
289
290    const uint32_t  mSampleRate;
291
292    NBLog::Writer   mDummyLog;
293public:
294    void            setLog(NBLog::Writer* log);
295private:
296    state_t         mState __attribute__((aligned(32)));
297
298    // effect descriptor for the downmixer used by the mixer
299    static effect_descriptor_t sDwnmFxDesc;
300    // indicates whether a downmix effect has been found and is usable by this mixer
301    static bool                sIsMultichannelCapable;
302
303    // Call after changing either the enabled status of a track, or parameters of an enabled track.
304    // OK to call more often than that, but unnecessary.
305    void invalidateState(uint32_t mask);
306
307    static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
308    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
309    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
310    static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
311    static void unprepareTrackForReformat(track_t* pTrack, int trackName);
312    static void reconfigureBufferProviders(track_t* pTrack);
313
314    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
315            int32_t* aux);
316    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
317    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
318            int32_t* aux);
319    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
320            int32_t* aux);
321    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
322            int32_t* aux);
323    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
324            int32_t* aux);
325
326    static void process__validate(state_t* state, int64_t pts);
327    static void process__nop(state_t* state, int64_t pts);
328    static void process__genericNoResampling(state_t* state, int64_t pts);
329    static void process__genericResampling(state_t* state, int64_t pts);
330    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
331                                                          int64_t pts);
332#if 0
333    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
334                                                           int64_t pts);
335#endif
336
337    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
338                                      int outputFrameIndex);
339
340    static uint64_t         sLocalTimeFreq;
341    static pthread_once_t   sOnceControl;
342    static void             sInitRoutine();
343};
344
345// ----------------------------------------------------------------------------
346}; // namespace android
347
348#endif // ANDROID_AUDIO_MIXER_H
349