AudioMixer.h revision c56f3426099a3cf2d07ccff8886050c7fbce140f 
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
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
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 2 channels for output.
55    // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
56    // Adding support for > 2 channel output would require more than simply changing this value.
57    static const uint32_t MAX_NUM_CHANNELS = 2;
58    // maximum number of channels supported for the content
59    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
60
61    static const uint16_t UNITY_GAIN = 0x1000;
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
76        // set Parameter names
77        // for target TRACK
78        CHANNEL_MASK    = 0x4000,
79        FORMAT          = 0x4001,
80        MAIN_BUFFER     = 0x4002,
81        AUX_BUFFER      = 0x4003,
82        DOWNMIX_TYPE    = 0X4004,
83        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
84        // for target RESAMPLE
85        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
86                                  // parameter 'value' is the new sample rate in Hz.
87                                  // Only creates a sample rate converter the first time that
88                                  // the track sample rate is different from the mix sample rate.
89                                  // If the new sample rate is the same as the mix sample rate,
90                                  // and a sample rate converter already exists,
91                                  // then the sample rate converter remains present but is a no-op.
92        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
93                                  // This clears out the resampler's input buffer.
94        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
95                                  // the track is restored to the mix sample rate.
96        // for target RAMP_VOLUME and VOLUME (8 channels max)
97        // FIXME use float for these 3 to improve the dynamic range
98        VOLUME0         = 0x4200,
99        VOLUME1         = 0x4201,
100        AUXLEVEL        = 0x4210,
101    };
102
103
104    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
105
106    // Allocate a track name.  Returns new track name if successful, -1 on failure.
107    int         getTrackName(audio_channel_mask_t channelMask, int sessionId);
108
109    // Free an allocated track by name
110    void        deleteTrackName(int name);
111
112    // Enable or disable an allocated track by name
113    void        enable(int name);
114    void        disable(int name);
115
116    void        setParameter(int name, int target, int param, void *value);
117
118    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
119    void        process(int64_t pts);
120
121    uint32_t    trackNames() const { return mTrackNames; }
122
123    size_t      getUnreleasedFrames(int name) const;
124
125private:
126
127    enum {
128        // FIXME this representation permits up to 8 channels
129        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
130    };
131
132    enum {
133        NEEDS_CHANNEL_1             = 0x00000000,   // mono
134        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
135
136        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
137
138        NEEDS_MUTE                  = 0x00000100,
139        NEEDS_RESAMPLE              = 0x00001000,
140        NEEDS_AUX                   = 0x00010000,
141    };
142
143    struct state_t;
144    struct track_t;
145    class DownmixerBufferProvider;
146
147    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
148                           int32_t* aux);
149    static const int BLOCKSIZE = 16; // 4 cache lines
150
151    struct track_t {
152        uint32_t    needs;
153
154        union {
155        int16_t     volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
156        int32_t     volumeRL;
157        };
158
159        int32_t     prevVolume[MAX_NUM_CHANNELS];
160
161        // 16-byte boundary
162
163        int32_t     volumeInc[MAX_NUM_CHANNELS];
164        int32_t     auxInc;
165        int32_t     prevAuxLevel;
166
167        // 16-byte boundary
168
169        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
170        uint16_t    frameCount;
171
172        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
173        uint8_t     format;         // always 16
174        uint16_t    enabled;        // actually bool
175        audio_channel_mask_t channelMask;
176
177        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
178        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
179        AudioBufferProvider*                bufferProvider;
180
181        // 16-byte boundary
182
183        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
184
185        hook_t      hook;
186        const void* in;             // current location in buffer
187
188        // 16-byte boundary
189
190        AudioResampler*     resampler;
191        uint32_t            sampleRate;
192        int32_t*           mainBuffer;
193        int32_t*           auxBuffer;
194
195        // 16-byte boundary
196
197        DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
198
199        int32_t     sessionId;
200
201        audio_format_t mMixerFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
202
203        int32_t     padding[1];
204
205        // 16-byte boundary
206
207        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
208        bool        doesResample() const { return resampler != NULL; }
209        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
210        void        adjustVolumeRamp(bool aux);
211        size_t      getUnreleasedFrames() const { return resampler != NULL ?
212                                                    resampler->getUnreleasedFrames() : 0; };
213    };
214
215    // pad to 32-bytes to fill cache line
216    struct state_t {
217        uint32_t        enabledTracks;
218        uint32_t        needsChanged;
219        size_t          frameCount;
220        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL
221        int32_t         *outputTemp;
222        int32_t         *resampleTemp;
223        NBLog::Writer*  mLog;
224        int32_t         reserved[1];
225        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
226        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
227    };
228
229    // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
230    class DownmixerBufferProvider : public AudioBufferProvider {
231    public:
232        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
233        virtual void releaseBuffer(Buffer* buffer);
234        DownmixerBufferProvider();
235        virtual ~DownmixerBufferProvider();
236
237        AudioBufferProvider* mTrackBufferProvider;
238        effect_handle_t    mDownmixHandle;
239        effect_config_t    mDownmixConfig;
240    };
241
242    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
243    uint32_t        mTrackNames;
244
245    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
246    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
247    const uint32_t  mConfiguredNames;
248
249    const uint32_t  mSampleRate;
250
251    NBLog::Writer   mDummyLog;
252public:
253    void            setLog(NBLog::Writer* log);
254private:
255    state_t         mState __attribute__((aligned(32)));
256
257    // effect descriptor for the downmixer used by the mixer
258    static effect_descriptor_t sDwnmFxDesc;
259    // indicates whether a downmix effect has been found and is usable by this mixer
260    static bool                sIsMultichannelCapable;
261
262    // Call after changing either the enabled status of a track, or parameters of an enabled track.
263    // OK to call more often than that, but unnecessary.
264    void invalidateState(uint32_t mask);
265
266    static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
267    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
268    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
269
270    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
271            int32_t* aux);
272    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
273    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
274            int32_t* aux);
275    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
276            int32_t* aux);
277    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
278            int32_t* aux);
279    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
280            int32_t* aux);
281
282    static void process__validate(state_t* state, int64_t pts);
283    static void process__nop(state_t* state, int64_t pts);
284    static void process__genericNoResampling(state_t* state, int64_t pts);
285    static void process__genericResampling(state_t* state, int64_t pts);
286    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
287                                                          int64_t pts);
288#if 0
289    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
290                                                           int64_t pts);
291#endif
292
293    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
294                                      int outputFrameIndex);
295
296    static uint64_t         sLocalTimeFreq;
297    static pthread_once_t   sOnceControl;
298    static void             sInitRoutine();
299};
300
301// ----------------------------------------------------------------------------
302}; // namespace android
303
304#endif // ANDROID_AUDIO_MIXER_H
305