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 <media/AudioBufferProvider.h>
25#include <media/AudioResampler.h>
26#include <media/AudioResamplerPublic.h>
27#include <media/BufferProviders.h>
28#include <media/nbaio/NBLog.h>
29#include <system/audio.h>
30#include <utils/Compat.h>
31#include <utils/threads.h>
32
33// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
34#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
35
36namespace android {
37
38// ----------------------------------------------------------------------------
39
40class AudioMixer
41{
42public:
43                            AudioMixer(size_t frameCount, uint32_t sampleRate,
44                                       uint32_t maxNumTracks = MAX_NUM_TRACKS);
45
46    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
47
48
49    // This mixer has a hard-coded upper limit of 32 active track inputs.
50    // Adding support for > 32 tracks would require more than simply changing this value.
51    static const uint32_t MAX_NUM_TRACKS = 32;
52    // maximum number of channels supported by the mixer
53
54    // This mixer has a hard-coded upper limit of 8 channels for output.
55    static const uint32_t MAX_NUM_CHANNELS = 8;
56    static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
57    // maximum number of channels supported for the content
58    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
59
60    static const uint16_t UNITY_GAIN_INT = 0x1000;
61    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
62
63    enum { // names
64
65        // track names (MAX_NUM_TRACKS units)
66        TRACK0          = 0x1000,
67
68        // 0x2000 is unused
69
70        // setParameter targets
71        TRACK           = 0x3000,
72        RESAMPLE        = 0x3001,
73        RAMP_VOLUME     = 0x3002, // ramp to new volume
74        VOLUME          = 0x3003, // don't ramp
75        TIMESTRETCH     = 0x3004,
76
77        // set Parameter names
78        // for target TRACK
79        CHANNEL_MASK    = 0x4000,
80        FORMAT          = 0x4001,
81        MAIN_BUFFER     = 0x4002,
82        AUX_BUFFER      = 0x4003,
83        DOWNMIX_TYPE    = 0X4004,
84        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
85        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
86        // for target RESAMPLE
87        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
88                                  // parameter 'value' is the new sample rate in Hz.
89                                  // Only creates a sample rate converter the first time that
90                                  // the track sample rate is different from the mix sample rate.
91                                  // If the new sample rate is the same as the mix sample rate,
92                                  // and a sample rate converter already exists,
93                                  // then the sample rate converter remains present but is a no-op.
94        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
95                                  // This clears out the resampler's input buffer.
96        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
97                                  // the track is restored to the mix sample rate.
98        // for target RAMP_VOLUME and VOLUME (8 channels max)
99        // FIXME use float for these 3 to improve the dynamic range
100        VOLUME0         = 0x4200,
101        VOLUME1         = 0x4201,
102        AUXLEVEL        = 0x4210,
103        // for target TIMESTRETCH
104        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
105                                  // parameter 'value' is a pointer to the new playback rate.
106    };
107
108
109    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
110
111    // Allocate a track name.  Returns new track name if successful, -1 on failure.
112    // The failure could be because of an invalid channelMask or format, or that
113    // the track capacity of the mixer is exceeded.
114    int         getTrackName(audio_channel_mask_t channelMask,
115                             audio_format_t format, int sessionId);
116
117    // Free an allocated track by name
118    void        deleteTrackName(int name);
119
120    // Enable or disable an allocated track by name
121    void        enable(int name);
122    void        disable(int name);
123
124    void        setParameter(int name, int target, int param, void *value);
125
126    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
127    void        process();
128
129    uint32_t    trackNames() const { return mTrackNames; }
130
131    size_t      getUnreleasedFrames(int name) const;
132
133    static inline bool isValidPcmTrackFormat(audio_format_t format) {
134        switch (format) {
135        case AUDIO_FORMAT_PCM_8_BIT:
136        case AUDIO_FORMAT_PCM_16_BIT:
137        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
138        case AUDIO_FORMAT_PCM_32_BIT:
139        case AUDIO_FORMAT_PCM_FLOAT:
140            return true;
141        default:
142            return false;
143        }
144    }
145
146private:
147
148    enum {
149        // FIXME this representation permits up to 8 channels
150        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
151    };
152
153    enum {
154        NEEDS_CHANNEL_1             = 0x00000000,   // mono
155        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
156
157        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
158
159        NEEDS_MUTE                  = 0x00000100,
160        NEEDS_RESAMPLE              = 0x00001000,
161        NEEDS_AUX                   = 0x00010000,
162    };
163
164    struct state_t;
165    struct track_t;
166
167    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
168                           int32_t* aux);
169    static const int BLOCKSIZE = 16; // 4 cache lines
170
171    struct track_t {
172        uint32_t    needs;
173
174        // TODO: Eventually remove legacy integer volume settings
175        union {
176        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
177        int32_t     volumeRL;
178        };
179
180        int32_t     prevVolume[MAX_NUM_VOLUMES];
181
182        // 16-byte boundary
183
184        int32_t     volumeInc[MAX_NUM_VOLUMES];
185        int32_t     auxInc;
186        int32_t     prevAuxLevel;
187
188        // 16-byte boundary
189
190        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
191        uint16_t    frameCount;
192
193        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
194        uint8_t     unused_padding; // formerly format, was always 16
195        uint16_t    enabled;        // actually bool
196        audio_channel_mask_t channelMask;
197
198        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
199        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
200        AudioBufferProvider*                bufferProvider;
201
202        // 16-byte boundary
203
204        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
205
206        hook_t      hook;
207        const void* in;             // current location in buffer
208
209        // 16-byte boundary
210
211        AudioResampler*     resampler;
212        uint32_t            sampleRate;
213        int32_t*           mainBuffer;
214        int32_t*           auxBuffer;
215
216        // 16-byte boundary
217
218        /* Buffer providers are constructed to translate the track input data as needed.
219         *
220         * TODO: perhaps make a single PlaybackConverterProvider class to move
221         * all pre-mixer track buffer conversions outside the AudioMixer class.
222         *
223         * 1) mInputBufferProvider: The AudioTrack buffer provider.
224         * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
225         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
226         *    requires reformat. For example, it may convert floating point input to
227         *    PCM_16_bit if that's required by the downmixer.
228         * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
229         *    the number of channels required by the mixer sink.
230         * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
231         *    the downmixer requirements to the mixer engine input requirements.
232         * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
233         */
234        AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
235        PassthruBufferProvider*  mReformatBufferProvider; // provider wrapper for reformatting.
236        PassthruBufferProvider*  downmixerBufferProvider; // wrapper for channel conversion.
237        PassthruBufferProvider*  mPostDownmixReformatBufferProvider;
238        PassthruBufferProvider*  mTimestretchBufferProvider;
239
240        int32_t     sessionId;
241
242        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
243        audio_format_t mFormat;          // input track format
244        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
245                                         // each track must be converted to this format.
246        audio_format_t mDownmixRequiresFormat;  // required downmixer format
247                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
248                                                // AUDIO_FORMAT_INVALID if no required format
249
250        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
251        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
252        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
253
254        float          mAuxLevel;                     // floating point set aux level
255        float          mPrevAuxLevel;                 // floating point prev aux level
256        float          mAuxInc;                       // floating point aux increment
257
258        audio_channel_mask_t mMixerChannelMask;
259        uint32_t             mMixerChannelCount;
260
261        AudioPlaybackRate    mPlaybackRate;
262
263        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
264        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
265        bool        doesResample() const { return resampler != NULL; }
266        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
267        void        adjustVolumeRamp(bool aux, bool useFloat = false);
268        size_t      getUnreleasedFrames() const { return resampler != NULL ?
269                                                    resampler->getUnreleasedFrames() : 0; };
270
271        status_t    prepareForDownmix();
272        void        unprepareForDownmix();
273        status_t    prepareForReformat();
274        void        unprepareForReformat();
275        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
276        void        reconfigureBufferProviders();
277    };
278
279    typedef void (*process_hook_t)(state_t* state);
280
281    // pad to 32-bytes to fill cache line
282    struct state_t {
283        uint32_t        enabledTracks;
284        uint32_t        needsChanged;
285        size_t          frameCount;
286        process_hook_t  hook;   // one of process__*, never NULL
287        int32_t         *outputTemp;
288        int32_t         *resampleTemp;
289        NBLog::Writer*  mLog;
290        int32_t         reserved[1];
291        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
292        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
293    };
294
295    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
296    uint32_t        mTrackNames;
297
298    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
299    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
300    const uint32_t  mConfiguredNames;
301
302    const uint32_t  mSampleRate;
303
304    NBLog::Writer   mDummyLog;
305public:
306    void            setLog(NBLog::Writer* log);
307private:
308    state_t         mState __attribute__((aligned(32)));
309
310    // Call after changing either the enabled status of a track, or parameters of an enabled track.
311    // OK to call more often than that, but unnecessary.
312    void invalidateState(uint32_t mask);
313
314    bool setChannelMasks(int name,
315            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
316
317    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
318            int32_t* aux);
319    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
320    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
321            int32_t* aux);
322    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
323            int32_t* aux);
324    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
325            int32_t* aux);
326    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
327            int32_t* aux);
328
329    static void process__validate(state_t* state);
330    static void process__nop(state_t* state);
331    static void process__genericNoResampling(state_t* state);
332    static void process__genericResampling(state_t* state);
333    static void process__OneTrack16BitsStereoNoResampling(state_t* state);
334
335    static pthread_once_t   sOnceControl;
336    static void             sInitRoutine();
337
338    /* multi-format volume mixing function (calls template functions
339     * in AudioMixerOps.h).  The template parameters are as follows:
340     *
341     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
342     *   USEFLOATVOL (set to true if float volume is used)
343     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
344     *   TO: int32_t (Q4.27) or float
345     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
346     *   TA: int32_t (Q4.27)
347     */
348    template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
349        typename TO, typename TI, typename TA>
350    static void volumeMix(TO *out, size_t outFrames,
351            const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
352
353    // multi-format process hooks
354    template <int MIXTYPE, typename TO, typename TI, typename TA>
355    static void process_NoResampleOneTrack(state_t* state);
356
357    // multi-format track hooks
358    template <int MIXTYPE, typename TO, typename TI, typename TA>
359    static void track__Resample(track_t* t, TO* out, size_t frameCount,
360            TO* temp __unused, TA* aux);
361    template <int MIXTYPE, typename TO, typename TI, typename TA>
362    static void track__NoResample(track_t* t, TO* out, size_t frameCount,
363            TO* temp __unused, TA* aux);
364
365    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
366            void *in, audio_format_t mixerInFormat, size_t sampleCount);
367
368    // hook types
369    enum {
370        PROCESSTYPE_NORESAMPLEONETRACK,
371    };
372    enum {
373        TRACKTYPE_NOP,
374        TRACKTYPE_RESAMPLE,
375        TRACKTYPE_NORESAMPLE,
376        TRACKTYPE_NORESAMPLEMONO,
377    };
378
379    // functions for determining the proper process and track hooks.
380    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
381            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
382    static hook_t getTrackHook(int trackType, uint32_t channelCount,
383            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
384};
385
386// ----------------------------------------------------------------------------
387} // namespace android
388
389#endif // ANDROID_AUDIO_MIXER_H
390