AudioMixer.h revision a1ab7cc8611c83427b57f6d4d4ce7aad9d1c0330
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        SINK_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    int         getTrackName(audio_channel_mask_t channelMask, int sessionId);
104
105    // Free an allocated track by name
106    void        deleteTrackName(int name);
107
108    // Enable or disable an allocated track by name
109    void        enable(int name);
110    void        disable(int name);
111
112    void        setParameter(int name, int target, int param, void *value);
113
114    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
115    void        process(int64_t pts);
116
117    uint32_t    trackNames() const { return mTrackNames; }
118
119    size_t      getUnreleasedFrames(int name) const;
120
121private:
122
123    enum {
124        // FIXME this representation permits up to 8 channels
125        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
126    };
127
128    enum {
129        NEEDS_CHANNEL_1             = 0x00000000,   // mono
130        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
131
132        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
133
134        NEEDS_MUTE                  = 0x00000100,
135        NEEDS_RESAMPLE              = 0x00001000,
136        NEEDS_AUX                   = 0x00010000,
137    };
138
139    struct state_t;
140    struct track_t;
141    class DownmixerBufferProvider;
142
143    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
144                           int32_t* aux);
145    static const int BLOCKSIZE = 16; // 4 cache lines
146
147    struct track_t {
148        uint32_t    needs;
149
150        union {
151        int16_t     volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
152        int32_t     volumeRL;
153        };
154
155        int32_t     prevVolume[MAX_NUM_CHANNELS];
156
157        // 16-byte boundary
158
159        int32_t     volumeInc[MAX_NUM_CHANNELS];
160        int32_t     auxInc;
161        int32_t     prevAuxLevel;
162
163        // 16-byte boundary
164
165        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
166        uint16_t    frameCount;
167
168        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
169        uint8_t     format;         // always 16
170        uint16_t    enabled;        // actually bool
171        audio_channel_mask_t channelMask;
172
173        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
174        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
175        AudioBufferProvider*                bufferProvider;
176
177        // 16-byte boundary
178
179        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
180
181        hook_t      hook;
182        const void* in;             // current location in buffer
183
184        // 16-byte boundary
185
186        AudioResampler*     resampler;
187        uint32_t            sampleRate;
188        int32_t*           mainBuffer;
189        int32_t*           auxBuffer;
190
191        // 16-byte boundary
192
193        DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
194
195        int32_t     sessionId;
196
197        audio_format_t mSinkFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
198
199        int32_t     padding[1];
200
201        // 16-byte boundary
202
203        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
204        bool        doesResample() const { return resampler != NULL; }
205        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
206        void        adjustVolumeRamp(bool aux);
207        size_t      getUnreleasedFrames() const { return resampler != NULL ?
208                                                    resampler->getUnreleasedFrames() : 0; };
209    };
210
211    // pad to 32-bytes to fill cache line
212    struct state_t {
213        uint32_t        enabledTracks;
214        uint32_t        needsChanged;
215        size_t          frameCount;
216        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL
217        int32_t         *outputTemp;
218        int32_t         *resampleTemp;
219        NBLog::Writer*  mLog;
220        int32_t         reserved[1];
221        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
222        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
223    };
224
225    // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
226    class DownmixerBufferProvider : public AudioBufferProvider {
227    public:
228        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
229        virtual void releaseBuffer(Buffer* buffer);
230        DownmixerBufferProvider();
231        virtual ~DownmixerBufferProvider();
232
233        AudioBufferProvider* mTrackBufferProvider;
234        effect_handle_t    mDownmixHandle;
235        effect_config_t    mDownmixConfig;
236    };
237
238    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
239    uint32_t        mTrackNames;
240
241    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
242    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
243    const uint32_t  mConfiguredNames;
244
245    const uint32_t  mSampleRate;
246
247    NBLog::Writer   mDummyLog;
248public:
249    void            setLog(NBLog::Writer* log);
250private:
251    state_t         mState __attribute__((aligned(32)));
252
253    // effect descriptor for the downmixer used by the mixer
254    static effect_descriptor_t sDwnmFxDesc;
255    // indicates whether a downmix effect has been found and is usable by this mixer
256    static bool                sIsMultichannelCapable;
257
258    // Call after changing either the enabled status of a track, or parameters of an enabled track.
259    // OK to call more often than that, but unnecessary.
260    void invalidateState(uint32_t mask);
261
262    static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
263    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
264    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
265
266    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
267            int32_t* aux);
268    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
269    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
270            int32_t* aux);
271    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
272            int32_t* aux);
273    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
274            int32_t* aux);
275    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
276            int32_t* aux);
277
278    static void process__validate(state_t* state, int64_t pts);
279    static void process__nop(state_t* state, int64_t pts);
280    static void process__genericNoResampling(state_t* state, int64_t pts);
281    static void process__genericResampling(state_t* state, int64_t pts);
282    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
283                                                          int64_t pts);
284#if 0
285    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
286                                                           int64_t pts);
287#endif
288
289    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
290                                      int outputFrameIndex);
291
292    static uint64_t         sLocalTimeFreq;
293    static pthread_once_t   sOnceControl;
294    static void             sInitRoutine();
295};
296
297// ----------------------------------------------------------------------------
298}; // namespace android
299
300#endif // ANDROID_AUDIO_MIXER_H
301