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 <hardware/audio_effect.h>
25#include <media/AudioBufferProvider.h>
26#include <media/AudioResamplerPublic.h>
27#include <media/nbaio/NBLog.h>
28#include <system/audio.h>
29#include <utils/Compat.h>
30#include <utils/threads.h>
31
32#include "AudioResampler.h"
33#include "BufferProviders.h"
34
35// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
36#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
37
38namespace android {
39
40// ----------------------------------------------------------------------------
41
42class AudioMixer
43{
44public:
45                            AudioMixer(size_t frameCount, uint32_t sampleRate,
46                                       uint32_t maxNumTracks = MAX_NUM_TRACKS);
47
48    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
49
50
51    // This mixer has a hard-coded upper limit of 32 active track inputs.
52    // Adding support for > 32 tracks would require more than simply changing this value.
53    static const uint32_t MAX_NUM_TRACKS = 32;
54    // maximum number of channels supported by the mixer
55
56    // This mixer has a hard-coded upper limit of 8 channels for output.
57    static const uint32_t MAX_NUM_CHANNELS = 8;
58    static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
59    // maximum number of channels supported for the content
60    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
61
62    static const uint16_t UNITY_GAIN_INT = 0x1000;
63    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
64
65    enum { // names
66
67        // track names (MAX_NUM_TRACKS units)
68        TRACK0          = 0x1000,
69
70        // 0x2000 is unused
71
72        // setParameter targets
73        TRACK           = 0x3000,
74        RESAMPLE        = 0x3001,
75        RAMP_VOLUME     = 0x3002, // ramp to new volume
76        VOLUME          = 0x3003, // don't ramp
77        TIMESTRETCH     = 0x3004,
78
79        // set Parameter names
80        // for target TRACK
81        CHANNEL_MASK    = 0x4000,
82        FORMAT          = 0x4001,
83        MAIN_BUFFER     = 0x4002,
84        AUX_BUFFER      = 0x4003,
85        DOWNMIX_TYPE    = 0X4004,
86        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
87        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
88        // for target RESAMPLE
89        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
90                                  // parameter 'value' is the new sample rate in Hz.
91                                  // Only creates a sample rate converter the first time that
92                                  // the track sample rate is different from the mix sample rate.
93                                  // If the new sample rate is the same as the mix sample rate,
94                                  // and a sample rate converter already exists,
95                                  // then the sample rate converter remains present but is a no-op.
96        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
97                                  // This clears out the resampler's input buffer.
98        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
99                                  // the track is restored to the mix sample rate.
100        // for target RAMP_VOLUME and VOLUME (8 channels max)
101        // FIXME use float for these 3 to improve the dynamic range
102        VOLUME0         = 0x4200,
103        VOLUME1         = 0x4201,
104        AUXLEVEL        = 0x4210,
105        // for target TIMESTRETCH
106        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
107                                  // parameter 'value' is a pointer to the new playback rate.
108    };
109
110
111    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
112
113    // Allocate a track name.  Returns new track name if successful, -1 on failure.
114    // The failure could be because of an invalid channelMask or format, or that
115    // the track capacity of the mixer is exceeded.
116    int         getTrackName(audio_channel_mask_t channelMask,
117                             audio_format_t format, int sessionId);
118
119    // Free an allocated track by name
120    void        deleteTrackName(int name);
121
122    // Enable or disable an allocated track by name
123    void        enable(int name);
124    void        disable(int name);
125
126    void        setParameter(int name, int target, int param, void *value);
127
128    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
129    void        process();
130
131    uint32_t    trackNames() const { return mTrackNames; }
132
133    size_t      getUnreleasedFrames(int name) const;
134
135    static inline bool isValidPcmTrackFormat(audio_format_t format) {
136        switch (format) {
137        case AUDIO_FORMAT_PCM_8_BIT:
138        case AUDIO_FORMAT_PCM_16_BIT:
139        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
140        case AUDIO_FORMAT_PCM_32_BIT:
141        case AUDIO_FORMAT_PCM_FLOAT:
142            return true;
143        default:
144            return false;
145        }
146    }
147
148private:
149
150    enum {
151        // FIXME this representation permits up to 8 channels
152        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
153    };
154
155    enum {
156        NEEDS_CHANNEL_1             = 0x00000000,   // mono
157        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
158
159        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
160
161        NEEDS_MUTE                  = 0x00000100,
162        NEEDS_RESAMPLE              = 0x00001000,
163        NEEDS_AUX                   = 0x00010000,
164    };
165
166    struct state_t;
167    struct track_t;
168
169    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
170                           int32_t* aux);
171    static const int BLOCKSIZE = 16; // 4 cache lines
172
173    struct track_t {
174        uint32_t    needs;
175
176        // TODO: Eventually remove legacy integer volume settings
177        union {
178        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
179        int32_t     volumeRL;
180        };
181
182        int32_t     prevVolume[MAX_NUM_VOLUMES];
183
184        // 16-byte boundary
185
186        int32_t     volumeInc[MAX_NUM_VOLUMES];
187        int32_t     auxInc;
188        int32_t     prevAuxLevel;
189
190        // 16-byte boundary
191
192        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
193        uint16_t    frameCount;
194
195        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
196        uint8_t     unused_padding; // formerly format, was always 16
197        uint16_t    enabled;        // actually bool
198        audio_channel_mask_t channelMask;
199
200        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
201        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
202        AudioBufferProvider*                bufferProvider;
203
204        // 16-byte boundary
205
206        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
207
208        hook_t      hook;
209        const void* in;             // current location in buffer
210
211        // 16-byte boundary
212
213        AudioResampler*     resampler;
214        uint32_t            sampleRate;
215        int32_t*           mainBuffer;
216        int32_t*           auxBuffer;
217
218        // 16-byte boundary
219
220        /* Buffer providers are constructed to translate the track input data as needed.
221         *
222         * TODO: perhaps make a single PlaybackConverterProvider class to move
223         * all pre-mixer track buffer conversions outside the AudioMixer class.
224         *
225         * 1) mInputBufferProvider: The AudioTrack buffer provider.
226         * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
227         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
228         *    requires reformat. For example, it may convert floating point input to
229         *    PCM_16_bit if that's required by the downmixer.
230         * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
231         *    the number of channels required by the mixer sink.
232         * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
233         *    the downmixer requirements to the mixer engine input requirements.
234         * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
235         */
236        AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
237        PassthruBufferProvider*  mReformatBufferProvider; // provider wrapper for reformatting.
238        PassthruBufferProvider*  downmixerBufferProvider; // wrapper for channel conversion.
239        PassthruBufferProvider*  mPostDownmixReformatBufferProvider;
240        PassthruBufferProvider*  mTimestretchBufferProvider;
241
242        int32_t     sessionId;
243
244        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
245        audio_format_t mFormat;          // input track format
246        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
247                                         // each track must be converted to this format.
248        audio_format_t mDownmixRequiresFormat;  // required downmixer format
249                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
250                                                // AUDIO_FORMAT_INVALID if no required format
251
252        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
253        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
254        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
255
256        float          mAuxLevel;                     // floating point set aux level
257        float          mPrevAuxLevel;                 // floating point prev aux level
258        float          mAuxInc;                       // floating point aux increment
259
260        audio_channel_mask_t mMixerChannelMask;
261        uint32_t             mMixerChannelCount;
262
263        AudioPlaybackRate    mPlaybackRate;
264
265        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
266        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
267        bool        doesResample() const { return resampler != NULL; }
268        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
269        void        adjustVolumeRamp(bool aux, bool useFloat = false);
270        size_t      getUnreleasedFrames() const { return resampler != NULL ?
271                                                    resampler->getUnreleasedFrames() : 0; };
272
273        status_t    prepareForDownmix();
274        void        unprepareForDownmix();
275        status_t    prepareForReformat();
276        void        unprepareForReformat();
277        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
278        void        reconfigureBufferProviders();
279    };
280
281    typedef void (*process_hook_t)(state_t* state);
282
283    // pad to 32-bytes to fill cache line
284    struct state_t {
285        uint32_t        enabledTracks;
286        uint32_t        needsChanged;
287        size_t          frameCount;
288        process_hook_t  hook;   // one of process__*, never NULL
289        int32_t         *outputTemp;
290        int32_t         *resampleTemp;
291        NBLog::Writer*  mLog;
292        int32_t         reserved[1];
293        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
294        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
295    };
296
297    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
298    uint32_t        mTrackNames;
299
300    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
301    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
302    const uint32_t  mConfiguredNames;
303
304    const uint32_t  mSampleRate;
305
306    NBLog::Writer   mDummyLog;
307public:
308    void            setLog(NBLog::Writer* log);
309private:
310    state_t         mState __attribute__((aligned(32)));
311
312    // Call after changing either the enabled status of a track, or parameters of an enabled track.
313    // OK to call more often than that, but unnecessary.
314    void invalidateState(uint32_t mask);
315
316    bool setChannelMasks(int name,
317            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
318
319    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
320            int32_t* aux);
321    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
322    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
323            int32_t* aux);
324    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
325            int32_t* aux);
326    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
327            int32_t* aux);
328    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
329            int32_t* aux);
330
331    static void process__validate(state_t* state);
332    static void process__nop(state_t* state);
333    static void process__genericNoResampling(state_t* state);
334    static void process__genericResampling(state_t* state);
335    static void process__OneTrack16BitsStereoNoResampling(state_t* state);
336
337    static pthread_once_t   sOnceControl;
338    static void             sInitRoutine();
339
340    /* multi-format volume mixing function (calls template functions
341     * in AudioMixerOps.h).  The template parameters are as follows:
342     *
343     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
344     *   USEFLOATVOL (set to true if float volume is used)
345     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
346     *   TO: int32_t (Q4.27) or float
347     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
348     *   TA: int32_t (Q4.27)
349     */
350    template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
351        typename TO, typename TI, typename TA>
352    static void volumeMix(TO *out, size_t outFrames,
353            const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
354
355    // multi-format process hooks
356    template <int MIXTYPE, typename TO, typename TI, typename TA>
357    static void process_NoResampleOneTrack(state_t* state);
358
359    // multi-format track hooks
360    template <int MIXTYPE, typename TO, typename TI, typename TA>
361    static void track__Resample(track_t* t, TO* out, size_t frameCount,
362            TO* temp __unused, TA* aux);
363    template <int MIXTYPE, typename TO, typename TI, typename TA>
364    static void track__NoResample(track_t* t, TO* out, size_t frameCount,
365            TO* temp __unused, TA* aux);
366
367    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
368            void *in, audio_format_t mixerInFormat, size_t sampleCount);
369
370    // hook types
371    enum {
372        PROCESSTYPE_NORESAMPLEONETRACK,
373    };
374    enum {
375        TRACKTYPE_NOP,
376        TRACKTYPE_RESAMPLE,
377        TRACKTYPE_NORESAMPLE,
378        TRACKTYPE_NORESAMPLEMONO,
379    };
380
381    // functions for determining the proper process and track hooks.
382    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
383            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
384    static hook_t getTrackHook(int trackType, uint32_t channelCount,
385            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
386};
387
388// ----------------------------------------------------------------------------
389} // namespace android
390
391#endif // ANDROID_AUDIO_MIXER_H
392