AudioMixer.cpp revision 395db4bfa5b43a839f95632676d59cde99a9840d 
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#define LOG_TAG "AudioMixer"
19//#define LOG_NDEBUG 0
20
21#include "Configuration.h"
22#include <stdint.h>
23#include <string.h>
24#include <stdlib.h>
25#include <math.h>
26#include <sys/types.h>
27
28#include <utils/Errors.h>
29#include <utils/Log.h>
30
31#include <cutils/bitops.h>
32#include <cutils/compiler.h>
33#include <utils/Debug.h>
34
35#include <system/audio.h>
36
37#include <audio_utils/primitives.h>
38#include <audio_utils/format.h>
39#include <common_time/local_clock.h>
40#include <common_time/cc_helper.h>
41
42#include <media/EffectsFactoryApi.h>
43#include <audio_effects/effect_downmix.h>
44
45#include "AudioMixerOps.h"
46#include "AudioMixer.h"
47
48// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
49#ifndef FCC_2
50#define FCC_2 2
51#endif
52
53// Look for MONO_HACK for any Mono hack involving legacy mono channel to
54// stereo channel conversion.
55
56/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
57 * being used. This is a considerable amount of log spam, so don't enable unless you
58 * are verifying the hook based code.
59 */
60//#define VERY_VERY_VERBOSE_LOGGING
61#ifdef VERY_VERY_VERBOSE_LOGGING
62#define ALOGVV ALOGV
63//define ALOGVV printf  // for test-mixer.cpp
64#else
65#define ALOGVV(a...) do { } while (0)
66#endif
67
68#ifndef ARRAY_SIZE
69#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
70#endif
71
72// Set kUseNewMixer to true to use the new mixer engine. Otherwise the
73// original code will be used.  This is false for now.
74static const bool kUseNewMixer = false;
75
76// Set kUseFloat to true to allow floating input into the mixer engine.
77// If kUseNewMixer is false, this is ignored or may be overridden internally
78// because of downmix/upmix support.
79static const bool kUseFloat = true;
80
81// Set to default copy buffer size in frames for input processing.
82static const size_t kCopyBufferFrameCount = 256;
83
84namespace android {
85
86// ----------------------------------------------------------------------------
87
88template <typename T>
89T min(const T& a, const T& b)
90{
91    return a < b ? a : b;
92}
93
94AudioMixer::CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize,
95        size_t outputFrameSize, size_t bufferFrameCount) :
96        mInputFrameSize(inputFrameSize),
97        mOutputFrameSize(outputFrameSize),
98        mLocalBufferFrameCount(bufferFrameCount),
99        mLocalBufferData(NULL),
100        mConsumed(0)
101{
102    ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this,
103            inputFrameSize, outputFrameSize, bufferFrameCount);
104    LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0,
105            "Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)",
106            inputFrameSize, outputFrameSize);
107    if (mLocalBufferFrameCount) {
108        (void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize);
109    }
110    mBuffer.frameCount = 0;
111}
112
113AudioMixer::CopyBufferProvider::~CopyBufferProvider()
114{
115    ALOGV("~CopyBufferProvider(%p)", this);
116    if (mBuffer.frameCount != 0) {
117        mTrackBufferProvider->releaseBuffer(&mBuffer);
118    }
119    free(mLocalBufferData);
120}
121
122status_t AudioMixer::CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
123        int64_t pts)
124{
125    //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
126    //        this, pBuffer, pBuffer->frameCount, pts);
127    if (mLocalBufferFrameCount == 0) {
128        status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
129        if (res == OK) {
130            copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
131        }
132        return res;
133    }
134    if (mBuffer.frameCount == 0) {
135        mBuffer.frameCount = pBuffer->frameCount;
136        status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
137        // At one time an upstream buffer provider had
138        // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
139        //
140        // By API spec, if res != OK, then mBuffer.frameCount == 0.
141        // but there may be improper implementations.
142        ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
143        if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
144            pBuffer->raw = NULL;
145            pBuffer->frameCount = 0;
146            return res;
147        }
148        mConsumed = 0;
149    }
150    ALOG_ASSERT(mConsumed < mBuffer.frameCount);
151    size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed);
152    count = min(count, pBuffer->frameCount);
153    pBuffer->raw = mLocalBufferData;
154    pBuffer->frameCount = count;
155    copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize,
156            pBuffer->frameCount);
157    return OK;
158}
159
160void AudioMixer::CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
161{
162    //ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))",
163    //        this, pBuffer, pBuffer->frameCount);
164    if (mLocalBufferFrameCount == 0) {
165        mTrackBufferProvider->releaseBuffer(pBuffer);
166        return;
167    }
168    // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
169    mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
170    if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
171        mTrackBufferProvider->releaseBuffer(&mBuffer);
172        ALOG_ASSERT(mBuffer.frameCount == 0);
173    }
174    pBuffer->raw = NULL;
175    pBuffer->frameCount = 0;
176}
177
178void AudioMixer::CopyBufferProvider::reset()
179{
180    if (mBuffer.frameCount != 0) {
181        mTrackBufferProvider->releaseBuffer(&mBuffer);
182    }
183    mConsumed = 0;
184}
185
186AudioMixer::DownmixerBufferProvider::DownmixerBufferProvider(
187        audio_channel_mask_t inputChannelMask,
188        audio_channel_mask_t outputChannelMask, audio_format_t format,
189        uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) :
190        CopyBufferProvider(
191            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
192            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
193            bufferFrameCount)  // set bufferFrameCount to 0 to do in-place
194{
195    ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d)",
196            this, inputChannelMask, outputChannelMask, format,
197            sampleRate, sessionId);
198    if (!sIsMultichannelCapable
199            || EffectCreate(&sDwnmFxDesc.uuid,
200                    sessionId,
201                    SESSION_ID_INVALID_AND_IGNORED,
202                    &mDownmixHandle) != 0) {
203         ALOGE("DownmixerBufferProvider() error creating downmixer effect");
204         mDownmixHandle = NULL;
205         return;
206     }
207     // channel input configuration will be overridden per-track
208     mDownmixConfig.inputCfg.channels = inputChannelMask;   // FIXME: Should be bits
209     mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits
210     mDownmixConfig.inputCfg.format = format;
211     mDownmixConfig.outputCfg.format = format;
212     mDownmixConfig.inputCfg.samplingRate = sampleRate;
213     mDownmixConfig.outputCfg.samplingRate = sampleRate;
214     mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
215     mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
216     // input and output buffer provider, and frame count will not be used as the downmix effect
217     // process() function is called directly (see DownmixerBufferProvider::getNextBuffer())
218     mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
219             EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
220     mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask;
221
222     int cmdStatus;
223     uint32_t replySize = sizeof(int);
224
225     // Configure downmixer
226     status_t status = (*mDownmixHandle)->command(mDownmixHandle,
227             EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
228             &mDownmixConfig /*pCmdData*/,
229             &replySize, &cmdStatus /*pReplyData*/);
230     if (status != 0 || cmdStatus != 0) {
231         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer",
232                 status, cmdStatus);
233         EffectRelease(mDownmixHandle);
234         mDownmixHandle = NULL;
235         return;
236     }
237
238     // Enable downmixer
239     replySize = sizeof(int);
240     status = (*mDownmixHandle)->command(mDownmixHandle,
241             EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
242             &replySize, &cmdStatus /*pReplyData*/);
243     if (status != 0 || cmdStatus != 0) {
244         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer",
245                 status, cmdStatus);
246         EffectRelease(mDownmixHandle);
247         mDownmixHandle = NULL;
248         return;
249     }
250
251     // Set downmix type
252     // parameter size rounded for padding on 32bit boundary
253     const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int);
254     const int downmixParamSize =
255             sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t);
256     effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize);
257     param->psize = sizeof(downmix_params_t);
258     const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE;
259     memcpy(param->data, &downmixParam, param->psize);
260     const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD;
261     param->vsize = sizeof(downmix_type_t);
262     memcpy(param->data + psizePadded, &downmixType, param->vsize);
263     replySize = sizeof(int);
264     status = (*mDownmixHandle)->command(mDownmixHandle,
265             EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */,
266             param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/);
267     free(param);
268     if (status != 0 || cmdStatus != 0) {
269         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type",
270                 status, cmdStatus);
271         EffectRelease(mDownmixHandle);
272         mDownmixHandle = NULL;
273         return;
274     }
275     ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType);
276}
277
278AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider()
279{
280    ALOGV("~DownmixerBufferProvider (%p)", this);
281    EffectRelease(mDownmixHandle);
282    mDownmixHandle = NULL;
283}
284
285void AudioMixer::DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
286{
287    mDownmixConfig.inputCfg.buffer.frameCount = frames;
288    mDownmixConfig.inputCfg.buffer.raw = const_cast<void *>(src);
289    mDownmixConfig.outputCfg.buffer.frameCount = frames;
290    mDownmixConfig.outputCfg.buffer.raw = dst;
291    // may be in-place if src == dst.
292    status_t res = (*mDownmixHandle)->process(mDownmixHandle,
293            &mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer);
294    ALOGE_IF(res != OK, "DownmixBufferProvider error %d", res);
295}
296
297/* call once in a pthread_once handler. */
298/*static*/ status_t AudioMixer::DownmixerBufferProvider::init()
299{
300    // find multichannel downmix effect if we have to play multichannel content
301    uint32_t numEffects = 0;
302    int ret = EffectQueryNumberEffects(&numEffects);
303    if (ret != 0) {
304        ALOGE("AudioMixer() error %d querying number of effects", ret);
305        return NO_INIT;
306    }
307    ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
308
309    for (uint32_t i = 0 ; i < numEffects ; i++) {
310        if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) {
311            ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
312            if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
313                ALOGI("found effect \"%s\" from %s",
314                        sDwnmFxDesc.name, sDwnmFxDesc.implementor);
315                sIsMultichannelCapable = true;
316                break;
317            }
318        }
319    }
320    ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
321    return NO_INIT;
322}
323
324/*static*/ bool AudioMixer::DownmixerBufferProvider::sIsMultichannelCapable = false;
325/*static*/ effect_descriptor_t AudioMixer::DownmixerBufferProvider::sDwnmFxDesc;
326
327AudioMixer::RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask,
328        audio_channel_mask_t outputChannelMask, audio_format_t format,
329        size_t bufferFrameCount) :
330        CopyBufferProvider(
331                audio_bytes_per_sample(format)
332                    * audio_channel_count_from_out_mask(inputChannelMask),
333                audio_bytes_per_sample(format)
334                    * audio_channel_count_from_out_mask(outputChannelMask),
335                bufferFrameCount),
336        mFormat(format),
337        mSampleSize(audio_bytes_per_sample(format)),
338        mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)),
339        mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask))
340{
341    ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
342            this, format, inputChannelMask, outputChannelMask,
343            mInputChannels, mOutputChannels);
344    // TODO: consider channel representation in index array formulation
345    // We ignore channel representation, and just use the bits.
346    memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry),
347            audio_channel_mask_get_bits(outputChannelMask),
348            audio_channel_mask_get_bits(inputChannelMask));
349}
350
351void AudioMixer::RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
352{
353    memcpy_by_index_array(dst, mOutputChannels,
354            src, mInputChannels, mIdxAry, mSampleSize, frames);
355}
356
357AudioMixer::ReformatBufferProvider::ReformatBufferProvider(int32_t channels,
358        audio_format_t inputFormat, audio_format_t outputFormat,
359        size_t bufferFrameCount) :
360        CopyBufferProvider(
361            channels * audio_bytes_per_sample(inputFormat),
362            channels * audio_bytes_per_sample(outputFormat),
363            bufferFrameCount),
364        mChannels(channels),
365        mInputFormat(inputFormat),
366        mOutputFormat(outputFormat)
367{
368    ALOGV("ReformatBufferProvider(%p)(%d, %#x, %#x)", this, channels, inputFormat, outputFormat);
369}
370
371void AudioMixer::ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
372{
373    memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannels);
374}
375
376// ----------------------------------------------------------------------------
377
378// Ensure mConfiguredNames bitmask is initialized properly on all architectures.
379// The value of 1 << x is undefined in C when x >= 32.
380
381AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)
382    :   mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1),
383        mSampleRate(sampleRate)
384{
385    ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u",
386            maxNumTracks, MAX_NUM_TRACKS);
387
388    // AudioMixer is not yet capable of more than 32 active track inputs
389    ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS);
390
391    pthread_once(&sOnceControl, &sInitRoutine);
392
393    mState.enabledTracks= 0;
394    mState.needsChanged = 0;
395    mState.frameCount   = frameCount;
396    mState.hook         = process__nop;
397    mState.outputTemp   = NULL;
398    mState.resampleTemp = NULL;
399    mState.mLog         = &mDummyLog;
400    // mState.reserved
401
402    // FIXME Most of the following initialization is probably redundant since
403    // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0
404    // and mTrackNames is initially 0.  However, leave it here until that's verified.
405    track_t* t = mState.tracks;
406    for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
407        t->resampler = NULL;
408        t->downmixerBufferProvider = NULL;
409        t->mReformatBufferProvider = NULL;
410        t++;
411    }
412
413}
414
415AudioMixer::~AudioMixer()
416{
417    track_t* t = mState.tracks;
418    for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
419        delete t->resampler;
420        delete t->downmixerBufferProvider;
421        delete t->mReformatBufferProvider;
422        t++;
423    }
424    delete [] mState.outputTemp;
425    delete [] mState.resampleTemp;
426}
427
428void AudioMixer::setLog(NBLog::Writer *log)
429{
430    mState.mLog = log;
431}
432
433int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
434        audio_format_t format, int sessionId)
435{
436    if (!isValidPcmTrackFormat(format)) {
437        ALOGE("AudioMixer::getTrackName invalid format (%#x)", format);
438        return -1;
439    }
440    uint32_t names = (~mTrackNames) & mConfiguredNames;
441    if (names != 0) {
442        int n = __builtin_ctz(names);
443        ALOGV("add track (%d)", n);
444        // assume default parameters for the track, except where noted below
445        track_t* t = &mState.tracks[n];
446        t->needs = 0;
447
448        // Integer volume.
449        // Currently integer volume is kept for the legacy integer mixer.
450        // Will be removed when the legacy mixer path is removed.
451        t->volume[0] = UNITY_GAIN_INT;
452        t->volume[1] = UNITY_GAIN_INT;
453        t->prevVolume[0] = UNITY_GAIN_INT << 16;
454        t->prevVolume[1] = UNITY_GAIN_INT << 16;
455        t->volumeInc[0] = 0;
456        t->volumeInc[1] = 0;
457        t->auxLevel = 0;
458        t->auxInc = 0;
459        t->prevAuxLevel = 0;
460
461        // Floating point volume.
462        t->mVolume[0] = UNITY_GAIN_FLOAT;
463        t->mVolume[1] = UNITY_GAIN_FLOAT;
464        t->mPrevVolume[0] = UNITY_GAIN_FLOAT;
465        t->mPrevVolume[1] = UNITY_GAIN_FLOAT;
466        t->mVolumeInc[0] = 0.;
467        t->mVolumeInc[1] = 0.;
468        t->mAuxLevel = 0.;
469        t->mAuxInc = 0.;
470        t->mPrevAuxLevel = 0.;
471
472        // no initialization needed
473        // t->frameCount
474        t->channelCount = audio_channel_count_from_out_mask(channelMask);
475        t->enabled = false;
476        ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
477                "Non-stereo channel mask: %d\n", channelMask);
478        t->channelMask = channelMask;
479        t->sessionId = sessionId;
480        // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
481        t->bufferProvider = NULL;
482        t->buffer.raw = NULL;
483        // no initialization needed
484        // t->buffer.frameCount
485        t->hook = NULL;
486        t->in = NULL;
487        t->resampler = NULL;
488        t->sampleRate = mSampleRate;
489        // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
490        t->mainBuffer = NULL;
491        t->auxBuffer = NULL;
492        t->mInputBufferProvider = NULL;
493        t->mReformatBufferProvider = NULL;
494        t->downmixerBufferProvider = NULL;
495        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
496        t->mFormat = format;
497        t->mMixerInFormat = kUseFloat && kUseNewMixer
498                ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
499        t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
500                AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
501        t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
502        // Check the downmixing (or upmixing) requirements.
503        status_t status = initTrackDownmix(t, n);
504        if (status != OK) {
505            ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
506            return -1;
507        }
508        // initTrackDownmix() may change the input format requirement.
509        // If you desire floating point input to the mixer, it may change
510        // to integer because the downmixer requires integer to process.
511        ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
512        prepareTrackForReformat(t, n);
513        mTrackNames |= 1 << n;
514        return TRACK0 + n;
515    }
516    ALOGE("AudioMixer::getTrackName out of available tracks");
517    return -1;
518}
519
520void AudioMixer::invalidateState(uint32_t mask)
521{
522    if (mask != 0) {
523        mState.needsChanged |= mask;
524        mState.hook = process__validate;
525    }
526 }
527
528// Called when channel masks have changed for a track name
529// TODO: Fix Downmixbufferprofider not to (possibly) change mixer input format,
530// which will simplify this logic.
531bool AudioMixer::setChannelMasks(int name,
532        audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
533    track_t &track = mState.tracks[name];
534
535    if (trackChannelMask == track.channelMask
536            && mixerChannelMask == track.mMixerChannelMask) {
537        return false;  // no need to change
538    }
539    // always recompute for both channel masks even if only one has changed.
540    const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
541    const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
542    const bool mixerChannelCountChanged = track.mMixerChannelCount != mixerChannelCount;
543
544    ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX)
545            && trackChannelCount
546            && mixerChannelCount);
547    track.channelMask = trackChannelMask;
548    track.channelCount = trackChannelCount;
549    track.mMixerChannelMask = mixerChannelMask;
550    track.mMixerChannelCount = mixerChannelCount;
551
552    // channel masks have changed, does this track need a downmixer?
553    // update to try using our desired format (if we aren't already using it)
554    const audio_format_t prevMixerInFormat = track.mMixerInFormat;
555    track.mMixerInFormat = kUseFloat && kUseNewMixer
556            ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
557    const status_t status = initTrackDownmix(&mState.tracks[name], name);
558    ALOGE_IF(status != OK,
559            "initTrackDownmix error %d, track channel mask %#x, mixer channel mask %#x",
560            status, track.channelMask, track.mMixerChannelMask);
561
562    const bool mixerInFormatChanged = prevMixerInFormat != track.mMixerInFormat;
563    if (mixerInFormatChanged) {
564        prepareTrackForReformat(&track, name); // because of downmixer, track format may change!
565    }
566
567    if (track.resampler && (mixerInFormatChanged || mixerChannelCountChanged)) {
568        // resampler input format or channels may have changed.
569        const uint32_t resetToSampleRate = track.sampleRate;
570        delete track.resampler;
571        track.resampler = NULL;
572        track.sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
573        // recreate the resampler with updated format, channels, saved sampleRate.
574        track.setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
575    }
576    return true;
577}
578
579status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackName)
580{
581    // Only remix (upmix or downmix) if the track and mixer/device channel masks
582    // are not the same and not handled internally, as mono -> stereo currently is.
583    if (pTrack->channelMask != pTrack->mMixerChannelMask
584            && !(pTrack->channelMask == AUDIO_CHANNEL_OUT_MONO
585                    && pTrack->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
586        return prepareTrackForDownmix(pTrack, trackName);
587    }
588    // no remix necessary
589    unprepareTrackForDownmix(pTrack, trackName);
590    return NO_ERROR;
591}
592
593void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) {
594    ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
595
596    if (pTrack->downmixerBufferProvider != NULL) {
597        // this track had previously been configured with a downmixer, delete it
598        ALOGV(" deleting old downmixer");
599        delete pTrack->downmixerBufferProvider;
600        pTrack->downmixerBufferProvider = NULL;
601        reconfigureBufferProviders(pTrack);
602    } else {
603        ALOGV(" nothing to do, no downmixer to delete");
604    }
605}
606
607status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName)
608{
609    ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask);
610
611    // discard the previous downmixer if there was one
612    unprepareTrackForDownmix(pTrack, trackName);
613    if (DownmixerBufferProvider::isMultichannelCapable()) {
614        DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(pTrack->channelMask,
615                pTrack->mMixerChannelMask,
616                AUDIO_FORMAT_PCM_16_BIT /* TODO: use pTrack->mMixerInFormat, now only PCM 16 */,
617                pTrack->sampleRate, pTrack->sessionId, kCopyBufferFrameCount);
618
619        if (pDbp->isValid()) { // if constructor completed properly
620            pTrack->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
621            pTrack->downmixerBufferProvider = pDbp;
622            reconfigureBufferProviders(pTrack);
623            return NO_ERROR;
624        }
625        delete pDbp;
626    }
627
628    // Effect downmixer does not accept the channel conversion.  Let's use our remixer.
629    RemixBufferProvider* pRbp = new RemixBufferProvider(pTrack->channelMask,
630            pTrack->mMixerChannelMask, pTrack->mMixerInFormat, kCopyBufferFrameCount);
631    // Remix always finds a conversion whereas Downmixer effect above may fail.
632    pTrack->downmixerBufferProvider = pRbp;
633    reconfigureBufferProviders(pTrack);
634    return NO_ERROR;
635}
636
637void AudioMixer::unprepareTrackForReformat(track_t* pTrack, int trackName __unused) {
638    ALOGV("AudioMixer::unprepareTrackForReformat(%d)", trackName);
639    if (pTrack->mReformatBufferProvider != NULL) {
640        delete pTrack->mReformatBufferProvider;
641        pTrack->mReformatBufferProvider = NULL;
642        reconfigureBufferProviders(pTrack);
643    }
644}
645
646status_t AudioMixer::prepareTrackForReformat(track_t* pTrack, int trackName)
647{
648    ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
649    // discard the previous reformatter if there was one
650    unprepareTrackForReformat(pTrack, trackName);
651    // only configure reformatter if needed
652    if (pTrack->mFormat != pTrack->mMixerInFormat) {
653        pTrack->mReformatBufferProvider = new ReformatBufferProvider(
654                audio_channel_count_from_out_mask(pTrack->channelMask),
655                pTrack->mFormat, pTrack->mMixerInFormat,
656                kCopyBufferFrameCount);
657        reconfigureBufferProviders(pTrack);
658    }
659    return NO_ERROR;
660}
661
662void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
663{
664    pTrack->bufferProvider = pTrack->mInputBufferProvider;
665    if (pTrack->mReformatBufferProvider) {
666        pTrack->mReformatBufferProvider->setBufferProvider(pTrack->bufferProvider);
667        pTrack->bufferProvider = pTrack->mReformatBufferProvider;
668    }
669    if (pTrack->downmixerBufferProvider) {
670        pTrack->downmixerBufferProvider->setBufferProvider(pTrack->bufferProvider);
671        pTrack->bufferProvider = pTrack->downmixerBufferProvider;
672    }
673}
674
675void AudioMixer::deleteTrackName(int name)
676{
677    ALOGV("AudioMixer::deleteTrackName(%d)", name);
678    name -= TRACK0;
679    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
680    ALOGV("deleteTrackName(%d)", name);
681    track_t& track(mState.tracks[ name ]);
682    if (track.enabled) {
683        track.enabled = false;
684        invalidateState(1<<name);
685    }
686    // delete the resampler
687    delete track.resampler;
688    track.resampler = NULL;
689    // delete the downmixer
690    unprepareTrackForDownmix(&mState.tracks[name], name);
691    // delete the reformatter
692    unprepareTrackForReformat(&mState.tracks[name], name);
693
694    mTrackNames &= ~(1<<name);
695}
696
697void AudioMixer::enable(int name)
698{
699    name -= TRACK0;
700    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
701    track_t& track = mState.tracks[name];
702
703    if (!track.enabled) {
704        track.enabled = true;
705        ALOGV("enable(%d)", name);
706        invalidateState(1 << name);
707    }
708}
709
710void AudioMixer::disable(int name)
711{
712    name -= TRACK0;
713    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
714    track_t& track = mState.tracks[name];
715
716    if (track.enabled) {
717        track.enabled = false;
718        ALOGV("disable(%d)", name);
719        invalidateState(1 << name);
720    }
721}
722
723/* Sets the volume ramp variables for the AudioMixer.
724 *
725 * The volume ramp variables are used to transition from the previous
726 * volume to the set volume.  ramp controls the duration of the transition.
727 * Its value is typically one state framecount period, but may also be 0,
728 * meaning "immediate."
729 *
730 * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
731 * even if there is a nonzero floating point increment (in that case, the volume
732 * change is immediate).  This restriction should be changed when the legacy mixer
733 * is removed (see #2).
734 * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
735 * when no longer needed.
736 *
737 * @param newVolume set volume target in floating point [0.0, 1.0].
738 * @param ramp number of frames to increment over. if ramp is 0, the volume
739 * should be set immediately.  Currently ramp should not exceed 65535 (frames).
740 * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
741 * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
742 * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
743 * @param pSetVolume pointer to the float target volume, set on return.
744 * @param pPrevVolume pointer to the float previous volume, set on return.
745 * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
746 * @return true if the volume has changed, false if volume is same.
747 */
748static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
749        int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
750        float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
751    if (newVolume == *pSetVolume) {
752        return false;
753    }
754    /* set the floating point volume variables */
755    if (ramp != 0) {
756        *pVolumeInc = (newVolume - *pSetVolume) / ramp;
757        *pPrevVolume = *pSetVolume;
758    } else {
759        *pVolumeInc = 0;
760        *pPrevVolume = newVolume;
761    }
762    *pSetVolume = newVolume;
763
764    /* set the legacy integer volume variables */
765    int32_t intVolume = newVolume * AudioMixer::UNITY_GAIN_INT;
766    if (intVolume > AudioMixer::UNITY_GAIN_INT) {
767        intVolume = AudioMixer::UNITY_GAIN_INT;
768    } else if (intVolume < 0) {
769        ALOGE("negative volume %.7g", newVolume);
770        intVolume = 0; // should never happen, but for safety check.
771    }
772    if (intVolume == *pIntSetVolume) {
773        *pIntVolumeInc = 0;
774        /* TODO: integer/float workaround: ignore floating volume ramp */
775        *pVolumeInc = 0;
776        *pPrevVolume = newVolume;
777        return true;
778    }
779    if (ramp != 0) {
780        *pIntVolumeInc = ((intVolume - *pIntSetVolume) << 16) / ramp;
781        *pIntPrevVolume = (*pIntVolumeInc == 0 ? intVolume : *pIntSetVolume) << 16;
782    } else {
783        *pIntVolumeInc = 0;
784        *pIntPrevVolume = intVolume << 16;
785    }
786    *pIntSetVolume = intVolume;
787    return true;
788}
789
790void AudioMixer::setParameter(int name, int target, int param, void *value)
791{
792    name -= TRACK0;
793    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
794    track_t& track = mState.tracks[name];
795
796    int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
797    int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
798
799    switch (target) {
800
801    case TRACK:
802        switch (param) {
803        case CHANNEL_MASK: {
804            const audio_channel_mask_t trackChannelMask =
805                static_cast<audio_channel_mask_t>(valueInt);
806            if (setChannelMasks(name, trackChannelMask, track.mMixerChannelMask)) {
807                ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
808                invalidateState(1 << name);
809            }
810            } break;
811        case MAIN_BUFFER:
812            if (track.mainBuffer != valueBuf) {
813                track.mainBuffer = valueBuf;
814                ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
815                invalidateState(1 << name);
816            }
817            break;
818        case AUX_BUFFER:
819            if (track.auxBuffer != valueBuf) {
820                track.auxBuffer = valueBuf;
821                ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
822                invalidateState(1 << name);
823            }
824            break;
825        case FORMAT: {
826            audio_format_t format = static_cast<audio_format_t>(valueInt);
827            if (track.mFormat != format) {
828                ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
829                track.mFormat = format;
830                ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
831                prepareTrackForReformat(&track, name);
832                invalidateState(1 << name);
833            }
834            } break;
835        // FIXME do we want to support setting the downmix type from AudioFlinger?
836        //         for a specific track? or per mixer?
837        /* case DOWNMIX_TYPE:
838            break          */
839        case MIXER_FORMAT: {
840            audio_format_t format = static_cast<audio_format_t>(valueInt);
841            if (track.mMixerFormat != format) {
842                track.mMixerFormat = format;
843                ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
844            }
845            } break;
846        case MIXER_CHANNEL_MASK: {
847            const audio_channel_mask_t mixerChannelMask =
848                    static_cast<audio_channel_mask_t>(valueInt);
849            if (setChannelMasks(name, track.channelMask, mixerChannelMask)) {
850                ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
851                invalidateState(1 << name);
852            }
853            } break;
854        default:
855            LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
856        }
857        break;
858
859    case RESAMPLE:
860        switch (param) {
861        case SAMPLE_RATE:
862            ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
863            if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
864                ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
865                        uint32_t(valueInt));
866                invalidateState(1 << name);
867            }
868            break;
869        case RESET:
870            track.resetResampler();
871            invalidateState(1 << name);
872            break;
873        case REMOVE:
874            delete track.resampler;
875            track.resampler = NULL;
876            track.sampleRate = mSampleRate;
877            invalidateState(1 << name);
878            break;
879        default:
880            LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
881        }
882        break;
883
884    case RAMP_VOLUME:
885    case VOLUME:
886        switch (param) {
887        case AUXLEVEL:
888            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
889                    target == RAMP_VOLUME ? mState.frameCount : 0,
890                    &track.auxLevel, &track.prevAuxLevel, &track.auxInc,
891                    &track.mAuxLevel, &track.mPrevAuxLevel, &track.mAuxInc)) {
892                ALOGV("setParameter(%s, AUXLEVEL: %04x)",
893                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track.auxLevel);
894                invalidateState(1 << name);
895            }
896            break;
897        default:
898            if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
899                if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
900                        target == RAMP_VOLUME ? mState.frameCount : 0,
901                        &track.volume[param - VOLUME0], &track.prevVolume[param - VOLUME0],
902                        &track.volumeInc[param - VOLUME0],
903                        &track.mVolume[param - VOLUME0], &track.mPrevVolume[param - VOLUME0],
904                        &track.mVolumeInc[param - VOLUME0])) {
905                    ALOGV("setParameter(%s, VOLUME%d: %04x)",
906                            target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
907                                    track.volume[param - VOLUME0]);
908                    invalidateState(1 << name);
909                }
910            } else {
911                LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
912            }
913        }
914        break;
915
916    default:
917        LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
918    }
919}
920
921bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
922{
923    if (trackSampleRate != devSampleRate || resampler != NULL) {
924        if (sampleRate != trackSampleRate) {
925            sampleRate = trackSampleRate;
926            if (resampler == NULL) {
927                ALOGV("Creating resampler from track %d Hz to device %d Hz",
928                        trackSampleRate, devSampleRate);
929                AudioResampler::src_quality quality;
930                // force lowest quality level resampler if use case isn't music or video
931                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
932                // quality level based on the initial ratio, but that could change later.
933                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
934                if (!((trackSampleRate == 44100 && devSampleRate == 48000) ||
935                      (trackSampleRate == 48000 && devSampleRate == 44100))) {
936                    quality = AudioResampler::DYN_LOW_QUALITY;
937                } else {
938                    quality = AudioResampler::DEFAULT_QUALITY;
939                }
940
941                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
942                // but if none exists, it is the channel count (1 for mono).
943                const int resamplerChannelCount = downmixerBufferProvider != NULL
944                        ? mMixerChannelCount : channelCount;
945                ALOGVV("Creating resampler:"
946                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
947                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
948                resampler = AudioResampler::create(
949                        mMixerInFormat,
950                        resamplerChannelCount,
951                        devSampleRate, quality);
952                resampler->setLocalTimeFreq(sLocalTimeFreq);
953            }
954            return true;
955        }
956    }
957    return false;
958}
959
960/* Checks to see if the volume ramp has completed and clears the increment
961 * variables appropriately.
962 *
963 * FIXME: There is code to handle int/float ramp variable switchover should it not
964 * complete within a mixer buffer processing call, but it is preferred to avoid switchover
965 * due to precision issues.  The switchover code is included for legacy code purposes
966 * and can be removed once the integer volume is removed.
967 *
968 * It is not sufficient to clear only the volumeInc integer variable because
969 * if one channel requires ramping, all channels are ramped.
970 *
971 * There is a bit of duplicated code here, but it keeps backward compatibility.
972 */
973inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat)
974{
975    if (useFloat) {
976        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
977            if (mVolumeInc[i] != 0 && fabs(mVolume[i] - mPrevVolume[i]) <= fabs(mVolumeInc[i])) {
978                volumeInc[i] = 0;
979                prevVolume[i] = volume[i] << 16;
980                mVolumeInc[i] = 0.;
981                mPrevVolume[i] = mVolume[i];
982            } else {
983                //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
984                prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
985            }
986        }
987    } else {
988        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
989            if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
990                    ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
991                volumeInc[i] = 0;
992                prevVolume[i] = volume[i] << 16;
993                mVolumeInc[i] = 0.;
994                mPrevVolume[i] = mVolume[i];
995            } else {
996                //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
997                mPrevVolume[i]  = float_from_u4_28(prevVolume[i]);
998            }
999        }
1000    }
1001    /* TODO: aux is always integer regardless of output buffer type */
1002    if (aux) {
1003        if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
1004                ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
1005            auxInc = 0;
1006            prevAuxLevel = auxLevel << 16;
1007            mAuxInc = 0.;
1008            mPrevAuxLevel = mAuxLevel;
1009        } else {
1010            //ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);
1011        }
1012    }
1013}
1014
1015size_t AudioMixer::getUnreleasedFrames(int name) const
1016{
1017    name -= TRACK0;
1018    if (uint32_t(name) < MAX_NUM_TRACKS) {
1019        return mState.tracks[name].getUnreleasedFrames();
1020    }
1021    return 0;
1022}
1023
1024void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
1025{
1026    name -= TRACK0;
1027    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
1028
1029    if (mState.tracks[name].mInputBufferProvider == bufferProvider) {
1030        return; // don't reset any buffer providers if identical.
1031    }
1032    if (mState.tracks[name].mReformatBufferProvider != NULL) {
1033        mState.tracks[name].mReformatBufferProvider->reset();
1034    } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
1035    }
1036
1037    mState.tracks[name].mInputBufferProvider = bufferProvider;
1038    reconfigureBufferProviders(&mState.tracks[name]);
1039}
1040
1041
1042void AudioMixer::process(int64_t pts)
1043{
1044    mState.hook(&mState, pts);
1045}
1046
1047
1048void AudioMixer::process__validate(state_t* state, int64_t pts)
1049{
1050    ALOGW_IF(!state->needsChanged,
1051        "in process__validate() but nothing's invalid");
1052
1053    uint32_t changed = state->needsChanged;
1054    state->needsChanged = 0; // clear the validation flag
1055
1056    // recompute which tracks are enabled / disabled
1057    uint32_t enabled = 0;
1058    uint32_t disabled = 0;
1059    while (changed) {
1060        const int i = 31 - __builtin_clz(changed);
1061        const uint32_t mask = 1<<i;
1062        changed &= ~mask;
1063        track_t& t = state->tracks[i];
1064        (t.enabled ? enabled : disabled) |= mask;
1065    }
1066    state->enabledTracks &= ~disabled;
1067    state->enabledTracks |=  enabled;
1068
1069    // compute everything we need...
1070    int countActiveTracks = 0;
1071    // TODO: fix all16BitsStereNoResample logic to
1072    // either properly handle muted tracks (it should ignore them)
1073    // or remove altogether as an obsolete optimization.
1074    bool all16BitsStereoNoResample = true;
1075    bool resampling = false;
1076    bool volumeRamp = false;
1077    uint32_t en = state->enabledTracks;
1078    while (en) {
1079        const int i = 31 - __builtin_clz(en);
1080        en &= ~(1<<i);
1081
1082        countActiveTracks++;
1083        track_t& t = state->tracks[i];
1084        uint32_t n = 0;
1085        // FIXME can overflow (mask is only 3 bits)
1086        n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
1087        if (t.doesResample()) {
1088            n |= NEEDS_RESAMPLE;
1089        }
1090        if (t.auxLevel != 0 && t.auxBuffer != NULL) {
1091            n |= NEEDS_AUX;
1092        }
1093
1094        if (t.volumeInc[0]|t.volumeInc[1]) {
1095            volumeRamp = true;
1096        } else if (!t.doesResample() && t.volumeRL == 0) {
1097            n |= NEEDS_MUTE;
1098        }
1099        t.needs = n;
1100
1101        if (n & NEEDS_MUTE) {
1102            t.hook = track__nop;
1103        } else {
1104            if (n & NEEDS_AUX) {
1105                all16BitsStereoNoResample = false;
1106            }
1107            if (n & NEEDS_RESAMPLE) {
1108                all16BitsStereoNoResample = false;
1109                resampling = true;
1110                t.hook = getTrackHook(TRACKTYPE_RESAMPLE, t.mMixerChannelCount,
1111                        t.mMixerInFormat, t.mMixerFormat);
1112                ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
1113                        "Track %d needs downmix + resample", i);
1114            } else {
1115                if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
1116                    t.hook = getTrackHook(
1117                            t.mMixerChannelCount == 2 // TODO: MONO_HACK.
1118                                ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
1119                            t.mMixerChannelCount,
1120                            t.mMixerInFormat, t.mMixerFormat);
1121                    all16BitsStereoNoResample = false;
1122                }
1123                if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
1124                    t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, t.mMixerChannelCount,
1125                            t.mMixerInFormat, t.mMixerFormat);
1126                    ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
1127                            "Track %d needs downmix", i);
1128                }
1129            }
1130        }
1131    }
1132
1133    // select the processing hooks
1134    state->hook = process__nop;
1135    if (countActiveTracks > 0) {
1136        if (resampling) {
1137            if (!state->outputTemp) {
1138                state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
1139            }
1140            if (!state->resampleTemp) {
1141                state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
1142            }
1143            state->hook = process__genericResampling;
1144        } else {
1145            if (state->outputTemp) {
1146                delete [] state->outputTemp;
1147                state->outputTemp = NULL;
1148            }
1149            if (state->resampleTemp) {
1150                delete [] state->resampleTemp;
1151                state->resampleTemp = NULL;
1152            }
1153            state->hook = process__genericNoResampling;
1154            if (all16BitsStereoNoResample && !volumeRamp) {
1155                if (countActiveTracks == 1) {
1156                    const int i = 31 - __builtin_clz(state->enabledTracks);
1157                    track_t& t = state->tracks[i];
1158                    if ((t.needs & NEEDS_MUTE) == 0) {
1159                        // The check prevents a muted track from acquiring a process hook.
1160                        //
1161                        // This is dangerous if the track is MONO as that requires
1162                        // special case handling due to implicit channel duplication.
1163                        // Stereo or Multichannel should actually be fine here.
1164                        state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
1165                                t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
1166                    }
1167                }
1168            }
1169        }
1170    }
1171
1172    ALOGV("mixer configuration change: %d activeTracks (%08x) "
1173        "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
1174        countActiveTracks, state->enabledTracks,
1175        all16BitsStereoNoResample, resampling, volumeRamp);
1176
1177   state->hook(state, pts);
1178
1179    // Now that the volume ramp has been done, set optimal state and
1180    // track hooks for subsequent mixer process
1181    if (countActiveTracks > 0) {
1182        bool allMuted = true;
1183        uint32_t en = state->enabledTracks;
1184        while (en) {
1185            const int i = 31 - __builtin_clz(en);
1186            en &= ~(1<<i);
1187            track_t& t = state->tracks[i];
1188            if (!t.doesResample() && t.volumeRL == 0) {
1189                t.needs |= NEEDS_MUTE;
1190                t.hook = track__nop;
1191            } else {
1192                allMuted = false;
1193            }
1194        }
1195        if (allMuted) {
1196            state->hook = process__nop;
1197        } else if (all16BitsStereoNoResample) {
1198            if (countActiveTracks == 1) {
1199                const int i = 31 - __builtin_clz(state->enabledTracks);
1200                track_t& t = state->tracks[i];
1201                // Muted single tracks handled by allMuted above.
1202                state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
1203                        t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
1204            }
1205        }
1206    }
1207}
1208
1209
1210void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
1211        int32_t* temp, int32_t* aux)
1212{
1213    ALOGVV("track__genericResample\n");
1214    t->resampler->setSampleRate(t->sampleRate);
1215
1216    // ramp gain - resample to temp buffer and scale/mix in 2nd step
1217    if (aux != NULL) {
1218        // always resample with unity gain when sending to auxiliary buffer to be able
1219        // to apply send level after resampling
1220        t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
1221        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(int32_t));
1222        t->resampler->resample(temp, outFrameCount, t->bufferProvider);
1223        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
1224            volumeRampStereo(t, out, outFrameCount, temp, aux);
1225        } else {
1226            volumeStereo(t, out, outFrameCount, temp, aux);
1227        }
1228    } else {
1229        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
1230            t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
1231            memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
1232            t->resampler->resample(temp, outFrameCount, t->bufferProvider);
1233            volumeRampStereo(t, out, outFrameCount, temp, aux);
1234        }
1235
1236        // constant gain
1237        else {
1238            t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
1239            t->resampler->resample(out, outFrameCount, t->bufferProvider);
1240        }
1241    }
1242}
1243
1244void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
1245        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
1246{
1247}
1248
1249void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
1250        int32_t* aux)
1251{
1252    int32_t vl = t->prevVolume[0];
1253    int32_t vr = t->prevVolume[1];
1254    const int32_t vlInc = t->volumeInc[0];
1255    const int32_t vrInc = t->volumeInc[1];
1256
1257    //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
1258    //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
1259    //       (vl + vlInc*frameCount)/65536.0f, frameCount);
1260
1261    // ramp volume
1262    if (CC_UNLIKELY(aux != NULL)) {
1263        int32_t va = t->prevAuxLevel;
1264        const int32_t vaInc = t->auxInc;
1265        int32_t l;
1266        int32_t r;
1267
1268        do {
1269            l = (*temp++ >> 12);
1270            r = (*temp++ >> 12);
1271            *out++ += (vl >> 16) * l;
1272            *out++ += (vr >> 16) * r;
1273            *aux++ += (va >> 17) * (l + r);
1274            vl += vlInc;
1275            vr += vrInc;
1276            va += vaInc;
1277        } while (--frameCount);
1278        t->prevAuxLevel = va;
1279    } else {
1280        do {
1281            *out++ += (vl >> 16) * (*temp++ >> 12);
1282            *out++ += (vr >> 16) * (*temp++ >> 12);
1283            vl += vlInc;
1284            vr += vrInc;
1285        } while (--frameCount);
1286    }
1287    t->prevVolume[0] = vl;
1288    t->prevVolume[1] = vr;
1289    t->adjustVolumeRamp(aux != NULL);
1290}
1291
1292void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
1293        int32_t* aux)
1294{
1295    const int16_t vl = t->volume[0];
1296    const int16_t vr = t->volume[1];
1297
1298    if (CC_UNLIKELY(aux != NULL)) {
1299        const int16_t va = t->auxLevel;
1300        do {
1301            int16_t l = (int16_t)(*temp++ >> 12);
1302            int16_t r = (int16_t)(*temp++ >> 12);
1303            out[0] = mulAdd(l, vl, out[0]);
1304            int16_t a = (int16_t)(((int32_t)l + r) >> 1);
1305            out[1] = mulAdd(r, vr, out[1]);
1306            out += 2;
1307            aux[0] = mulAdd(a, va, aux[0]);
1308            aux++;
1309        } while (--frameCount);
1310    } else {
1311        do {
1312            int16_t l = (int16_t)(*temp++ >> 12);
1313            int16_t r = (int16_t)(*temp++ >> 12);
1314            out[0] = mulAdd(l, vl, out[0]);
1315            out[1] = mulAdd(r, vr, out[1]);
1316            out += 2;
1317        } while (--frameCount);
1318    }
1319}
1320
1321void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
1322        int32_t* temp __unused, int32_t* aux)
1323{
1324    ALOGVV("track__16BitsStereo\n");
1325    const int16_t *in = static_cast<const int16_t *>(t->in);
1326
1327    if (CC_UNLIKELY(aux != NULL)) {
1328        int32_t l;
1329        int32_t r;
1330        // ramp gain
1331        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
1332            int32_t vl = t->prevVolume[0];
1333            int32_t vr = t->prevVolume[1];
1334            int32_t va = t->prevAuxLevel;
1335            const int32_t vlInc = t->volumeInc[0];
1336            const int32_t vrInc = t->volumeInc[1];
1337            const int32_t vaInc = t->auxInc;
1338            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
1339            //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
1340            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
1341
1342            do {
1343                l = (int32_t)*in++;
1344                r = (int32_t)*in++;
1345                *out++ += (vl >> 16) * l;
1346                *out++ += (vr >> 16) * r;
1347                *aux++ += (va >> 17) * (l + r);
1348                vl += vlInc;
1349                vr += vrInc;
1350                va += vaInc;
1351            } while (--frameCount);
1352
1353            t->prevVolume[0] = vl;
1354            t->prevVolume[1] = vr;
1355            t->prevAuxLevel = va;
1356            t->adjustVolumeRamp(true);
1357        }
1358
1359        // constant gain
1360        else {
1361            const uint32_t vrl = t->volumeRL;
1362            const int16_t va = (int16_t)t->auxLevel;
1363            do {
1364                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
1365                int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
1366                in += 2;
1367                out[0] = mulAddRL(1, rl, vrl, out[0]);
1368                out[1] = mulAddRL(0, rl, vrl, out[1]);
1369                out += 2;
1370                aux[0] = mulAdd(a, va, aux[0]);
1371                aux++;
1372            } while (--frameCount);
1373        }
1374    } else {
1375        // ramp gain
1376        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
1377            int32_t vl = t->prevVolume[0];
1378            int32_t vr = t->prevVolume[1];
1379            const int32_t vlInc = t->volumeInc[0];
1380            const int32_t vrInc = t->volumeInc[1];
1381
1382            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
1383            //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
1384            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
1385
1386            do {
1387                *out++ += (vl >> 16) * (int32_t) *in++;
1388                *out++ += (vr >> 16) * (int32_t) *in++;
1389                vl += vlInc;
1390                vr += vrInc;
1391            } while (--frameCount);
1392
1393            t->prevVolume[0] = vl;
1394            t->prevVolume[1] = vr;
1395            t->adjustVolumeRamp(false);
1396        }
1397
1398        // constant gain
1399        else {
1400            const uint32_t vrl = t->volumeRL;
1401            do {
1402                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
1403                in += 2;
1404                out[0] = mulAddRL(1, rl, vrl, out[0]);
1405                out[1] = mulAddRL(0, rl, vrl, out[1]);
1406                out += 2;
1407            } while (--frameCount);
1408        }
1409    }
1410    t->in = in;
1411}
1412
1413void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
1414        int32_t* temp __unused, int32_t* aux)
1415{
1416    ALOGVV("track__16BitsMono\n");
1417    const int16_t *in = static_cast<int16_t const *>(t->in);
1418
1419    if (CC_UNLIKELY(aux != NULL)) {
1420        // ramp gain
1421        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
1422            int32_t vl = t->prevVolume[0];
1423            int32_t vr = t->prevVolume[1];
1424            int32_t va = t->prevAuxLevel;
1425            const int32_t vlInc = t->volumeInc[0];
1426            const int32_t vrInc = t->volumeInc[1];
1427            const int32_t vaInc = t->auxInc;
1428
1429            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
1430            //         t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
1431            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
1432
1433            do {
1434                int32_t l = *in++;
1435                *out++ += (vl >> 16) * l;
1436                *out++ += (vr >> 16) * l;
1437                *aux++ += (va >> 16) * l;
1438                vl += vlInc;
1439                vr += vrInc;
1440                va += vaInc;
1441            } while (--frameCount);
1442
1443            t->prevVolume[0] = vl;
1444            t->prevVolume[1] = vr;
1445            t->prevAuxLevel = va;
1446            t->adjustVolumeRamp(true);
1447        }
1448        // constant gain
1449        else {
1450            const int16_t vl = t->volume[0];
1451            const int16_t vr = t->volume[1];
1452            const int16_t va = (int16_t)t->auxLevel;
1453            do {
1454                int16_t l = *in++;
1455                out[0] = mulAdd(l, vl, out[0]);
1456                out[1] = mulAdd(l, vr, out[1]);
1457                out += 2;
1458                aux[0] = mulAdd(l, va, aux[0]);
1459                aux++;
1460            } while (--frameCount);
1461        }
1462    } else {
1463        // ramp gain
1464        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
1465            int32_t vl = t->prevVolume[0];
1466            int32_t vr = t->prevVolume[1];
1467            const int32_t vlInc = t->volumeInc[0];
1468            const int32_t vrInc = t->volumeInc[1];
1469
1470            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
1471            //         t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
1472            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
1473
1474            do {
1475                int32_t l = *in++;
1476                *out++ += (vl >> 16) * l;
1477                *out++ += (vr >> 16) * l;
1478                vl += vlInc;
1479                vr += vrInc;
1480            } while (--frameCount);
1481
1482            t->prevVolume[0] = vl;
1483            t->prevVolume[1] = vr;
1484            t->adjustVolumeRamp(false);
1485        }
1486        // constant gain
1487        else {
1488            const int16_t vl = t->volume[0];
1489            const int16_t vr = t->volume[1];
1490            do {
1491                int16_t l = *in++;
1492                out[0] = mulAdd(l, vl, out[0]);
1493                out[1] = mulAdd(l, vr, out[1]);
1494                out += 2;
1495            } while (--frameCount);
1496        }
1497    }
1498    t->in = in;
1499}
1500
1501// no-op case
1502void AudioMixer::process__nop(state_t* state, int64_t pts)
1503{
1504    ALOGVV("process__nop\n");
1505    uint32_t e0 = state->enabledTracks;
1506    while (e0) {
1507        // process by group of tracks with same output buffer to
1508        // avoid multiple memset() on same buffer
1509        uint32_t e1 = e0, e2 = e0;
1510        int i = 31 - __builtin_clz(e1);
1511        {
1512            track_t& t1 = state->tracks[i];
1513            e2 &= ~(1<<i);
1514            while (e2) {
1515                i = 31 - __builtin_clz(e2);
1516                e2 &= ~(1<<i);
1517                track_t& t2 = state->tracks[i];
1518                if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
1519                    e1 &= ~(1<<i);
1520                }
1521            }
1522            e0 &= ~(e1);
1523
1524            memset(t1.mainBuffer, 0, state->frameCount * t1.mMixerChannelCount
1525                    * audio_bytes_per_sample(t1.mMixerFormat));
1526        }
1527
1528        while (e1) {
1529            i = 31 - __builtin_clz(e1);
1530            e1 &= ~(1<<i);
1531            {
1532                track_t& t3 = state->tracks[i];
1533                size_t outFrames = state->frameCount;
1534                while (outFrames) {
1535                    t3.buffer.frameCount = outFrames;
1536                    int64_t outputPTS = calculateOutputPTS(
1537                        t3, pts, state->frameCount - outFrames);
1538                    t3.bufferProvider->getNextBuffer(&t3.buffer, outputPTS);
1539                    if (t3.buffer.raw == NULL) break;
1540                    outFrames -= t3.buffer.frameCount;
1541                    t3.bufferProvider->releaseBuffer(&t3.buffer);
1542                }
1543            }
1544        }
1545    }
1546}
1547
1548// generic code without resampling
1549void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
1550{
1551    ALOGVV("process__genericNoResampling\n");
1552    int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
1553
1554    // acquire each track's buffer
1555    uint32_t enabledTracks = state->enabledTracks;
1556    uint32_t e0 = enabledTracks;
1557    while (e0) {
1558        const int i = 31 - __builtin_clz(e0);
1559        e0 &= ~(1<<i);
1560        track_t& t = state->tracks[i];
1561        t.buffer.frameCount = state->frameCount;
1562        t.bufferProvider->getNextBuffer(&t.buffer, pts);
1563        t.frameCount = t.buffer.frameCount;
1564        t.in = t.buffer.raw;
1565    }
1566
1567    e0 = enabledTracks;
1568    while (e0) {
1569        // process by group of tracks with same output buffer to
1570        // optimize cache use
1571        uint32_t e1 = e0, e2 = e0;
1572        int j = 31 - __builtin_clz(e1);
1573        track_t& t1 = state->tracks[j];
1574        e2 &= ~(1<<j);
1575        while (e2) {
1576            j = 31 - __builtin_clz(e2);
1577            e2 &= ~(1<<j);
1578            track_t& t2 = state->tracks[j];
1579            if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
1580                e1 &= ~(1<<j);
1581            }
1582        }
1583        e0 &= ~(e1);
1584        // this assumes output 16 bits stereo, no resampling
1585        int32_t *out = t1.mainBuffer;
1586        size_t numFrames = 0;
1587        do {
1588            memset(outTemp, 0, sizeof(outTemp));
1589            e2 = e1;
1590            while (e2) {
1591                const int i = 31 - __builtin_clz(e2);
1592                e2 &= ~(1<<i);
1593                track_t& t = state->tracks[i];
1594                size_t outFrames = BLOCKSIZE;
1595                int32_t *aux = NULL;
1596                if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
1597                    aux = t.auxBuffer + numFrames;
1598                }
1599                while (outFrames) {
1600                    // t.in == NULL can happen if the track was flushed just after having
1601                    // been enabled for mixing.
1602                   if (t.in == NULL) {
1603                        enabledTracks &= ~(1<<i);
1604                        e1 &= ~(1<<i);
1605                        break;
1606                    }
1607                    size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
1608                    if (inFrames > 0) {
1609                        t.hook(&t, outTemp + (BLOCKSIZE - outFrames) * t.mMixerChannelCount,
1610                                inFrames, state->resampleTemp, aux);
1611                        t.frameCount -= inFrames;
1612                        outFrames -= inFrames;
1613                        if (CC_UNLIKELY(aux != NULL)) {
1614                            aux += inFrames;
1615                        }
1616                    }
1617                    if (t.frameCount == 0 && outFrames) {
1618                        t.bufferProvider->releaseBuffer(&t.buffer);
1619                        t.buffer.frameCount = (state->frameCount - numFrames) -
1620                                (BLOCKSIZE - outFrames);
1621                        int64_t outputPTS = calculateOutputPTS(
1622                            t, pts, numFrames + (BLOCKSIZE - outFrames));
1623                        t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
1624                        t.in = t.buffer.raw;
1625                        if (t.in == NULL) {
1626                            enabledTracks &= ~(1<<i);
1627                            e1 &= ~(1<<i);
1628                            break;
1629                        }
1630                        t.frameCount = t.buffer.frameCount;
1631                    }
1632                }
1633            }
1634
1635            convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat,
1636                    BLOCKSIZE * t1.mMixerChannelCount);
1637            // TODO: fix ugly casting due to choice of out pointer type
1638            out = reinterpret_cast<int32_t*>((uint8_t*)out
1639                    + BLOCKSIZE * t1.mMixerChannelCount
1640                        * audio_bytes_per_sample(t1.mMixerFormat));
1641            numFrames += BLOCKSIZE;
1642        } while (numFrames < state->frameCount);
1643    }
1644
1645    // release each track's buffer
1646    e0 = enabledTracks;
1647    while (e0) {
1648        const int i = 31 - __builtin_clz(e0);
1649        e0 &= ~(1<<i);
1650        track_t& t = state->tracks[i];
1651        t.bufferProvider->releaseBuffer(&t.buffer);
1652    }
1653}
1654
1655
1656// generic code with resampling
1657void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
1658{
1659    ALOGVV("process__genericResampling\n");
1660    // this const just means that local variable outTemp doesn't change
1661    int32_t* const outTemp = state->outputTemp;
1662    size_t numFrames = state->frameCount;
1663
1664    uint32_t e0 = state->enabledTracks;
1665    while (e0) {
1666        // process by group of tracks with same output buffer
1667        // to optimize cache use
1668        uint32_t e1 = e0, e2 = e0;
1669        int j = 31 - __builtin_clz(e1);
1670        track_t& t1 = state->tracks[j];
1671        e2 &= ~(1<<j);
1672        while (e2) {
1673            j = 31 - __builtin_clz(e2);
1674            e2 &= ~(1<<j);
1675            track_t& t2 = state->tracks[j];
1676            if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
1677                e1 &= ~(1<<j);
1678            }
1679        }
1680        e0 &= ~(e1);
1681        int32_t *out = t1.mainBuffer;
1682        memset(outTemp, 0, sizeof(*outTemp) * t1.mMixerChannelCount * state->frameCount);
1683        while (e1) {
1684            const int i = 31 - __builtin_clz(e1);
1685            e1 &= ~(1<<i);
1686            track_t& t = state->tracks[i];
1687            int32_t *aux = NULL;
1688            if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
1689                aux = t.auxBuffer;
1690            }
1691
1692            // this is a little goofy, on the resampling case we don't
1693            // acquire/release the buffers because it's done by
1694            // the resampler.
1695            if (t.needs & NEEDS_RESAMPLE) {
1696                t.resampler->setPTS(pts);
1697                t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
1698            } else {
1699
1700                size_t outFrames = 0;
1701
1702                while (outFrames < numFrames) {
1703                    t.buffer.frameCount = numFrames - outFrames;
1704                    int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
1705                    t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
1706                    t.in = t.buffer.raw;
1707                    // t.in == NULL can happen if the track was flushed just after having
1708                    // been enabled for mixing.
1709                    if (t.in == NULL) break;
1710
1711                    if (CC_UNLIKELY(aux != NULL)) {
1712                        aux += outFrames;
1713                    }
1714                    t.hook(&t, outTemp + outFrames * t.mMixerChannelCount, t.buffer.frameCount,
1715                            state->resampleTemp, aux);
1716                    outFrames += t.buffer.frameCount;
1717                    t.bufferProvider->releaseBuffer(&t.buffer);
1718                }
1719            }
1720        }
1721        convertMixerFormat(out, t1.mMixerFormat,
1722                outTemp, t1.mMixerInFormat, numFrames * t1.mMixerChannelCount);
1723    }
1724}
1725
1726// one track, 16 bits stereo without resampling is the most common case
1727void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
1728                                                           int64_t pts)
1729{
1730    ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
1731    // This method is only called when state->enabledTracks has exactly
1732    // one bit set.  The asserts below would verify this, but are commented out
1733    // since the whole point of this method is to optimize performance.
1734    //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled");
1735    const int i = 31 - __builtin_clz(state->enabledTracks);
1736    //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
1737    const track_t& t = state->tracks[i];
1738
1739    AudioBufferProvider::Buffer& b(t.buffer);
1740
1741    int32_t* out = t.mainBuffer;
1742    float *fout = reinterpret_cast<float*>(out);
1743    size_t numFrames = state->frameCount;
1744
1745    const int16_t vl = t.volume[0];
1746    const int16_t vr = t.volume[1];
1747    const uint32_t vrl = t.volumeRL;
1748    while (numFrames) {
1749        b.frameCount = numFrames;
1750        int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
1751        t.bufferProvider->getNextBuffer(&b, outputPTS);
1752        const int16_t *in = b.i16;
1753
1754        // in == NULL can happen if the track was flushed just after having
1755        // been enabled for mixing.
1756        if (in == NULL || (((uintptr_t)in) & 3)) {
1757            memset(out, 0, numFrames
1758                    * t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat));
1759            ALOGE_IF((((uintptr_t)in) & 3),
1760                    "process__OneTrack16BitsStereoNoResampling: misaligned buffer"
1761                    " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
1762                    in, i, t.channelCount, t.needs, vrl, t.mVolume[0], t.mVolume[1]);
1763            return;
1764        }
1765        size_t outFrames = b.frameCount;
1766
1767        switch (t.mMixerFormat) {
1768        case AUDIO_FORMAT_PCM_FLOAT:
1769            do {
1770                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
1771                in += 2;
1772                int32_t l = mulRL(1, rl, vrl);
1773                int32_t r = mulRL(0, rl, vrl);
1774                *fout++ = float_from_q4_27(l);
1775                *fout++ = float_from_q4_27(r);
1776                // Note: In case of later int16_t sink output,
1777                // conversion and clamping is done by memcpy_to_i16_from_float().
1778            } while (--outFrames);
1779            break;
1780        case AUDIO_FORMAT_PCM_16_BIT:
1781            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
1782                // volume is boosted, so we might need to clamp even though
1783                // we process only one track.
1784                do {
1785                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
1786                    in += 2;
1787                    int32_t l = mulRL(1, rl, vrl) >> 12;
1788                    int32_t r = mulRL(0, rl, vrl) >> 12;
1789                    // clamping...
1790                    l = clamp16(l);
1791                    r = clamp16(r);
1792                    *out++ = (r<<16) | (l & 0xFFFF);
1793                } while (--outFrames);
1794            } else {
1795                do {
1796                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
1797                    in += 2;
1798                    int32_t l = mulRL(1, rl, vrl) >> 12;
1799                    int32_t r = mulRL(0, rl, vrl) >> 12;
1800                    *out++ = (r<<16) | (l & 0xFFFF);
1801                } while (--outFrames);
1802            }
1803            break;
1804        default:
1805            LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat);
1806        }
1807        numFrames -= b.frameCount;
1808        t.bufferProvider->releaseBuffer(&b);
1809    }
1810}
1811
1812int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
1813                                       int outputFrameIndex)
1814{
1815    if (AudioBufferProvider::kInvalidPTS == basePTS) {
1816        return AudioBufferProvider::kInvalidPTS;
1817    }
1818
1819    return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
1820}
1821
1822/*static*/ uint64_t AudioMixer::sLocalTimeFreq;
1823/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
1824
1825/*static*/ void AudioMixer::sInitRoutine()
1826{
1827    LocalClock lc;
1828    sLocalTimeFreq = lc.getLocalFreq(); // for the resampler
1829
1830    DownmixerBufferProvider::init(); // for the downmixer
1831}
1832
1833/* TODO: consider whether this level of optimization is necessary.
1834 * Perhaps just stick with a single for loop.
1835 */
1836
1837// Needs to derive a compile time constant (constexpr).  Could be targeted to go
1838// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
1839#define MIXTYPE_MONOVOL(mixtype) (mixtype == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
1840        mixtype == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : mixtype)
1841
1842/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
1843 * TO: int32_t (Q4.27) or float
1844 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
1845 * TA: int32_t (Q4.27)
1846 */
1847template <int MIXTYPE,
1848        typename TO, typename TI, typename TV, typename TA, typename TAV>
1849static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
1850        const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
1851{
1852    switch (channels) {
1853    case 1:
1854        volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
1855        break;
1856    case 2:
1857        volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
1858        break;
1859    case 3:
1860        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
1861                frameCount, in, aux, vol, volinc, vola, volainc);
1862        break;
1863    case 4:
1864        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
1865                frameCount, in, aux, vol, volinc, vola, volainc);
1866        break;
1867    case 5:
1868        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
1869                frameCount, in, aux, vol, volinc, vola, volainc);
1870        break;
1871    case 6:
1872        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
1873                frameCount, in, aux, vol, volinc, vola, volainc);
1874        break;
1875    case 7:
1876        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
1877                frameCount, in, aux, vol, volinc, vola, volainc);
1878        break;
1879    case 8:
1880        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
1881                frameCount, in, aux, vol, volinc, vola, volainc);
1882        break;
1883    }
1884}
1885
1886/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
1887 * TO: int32_t (Q4.27) or float
1888 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
1889 * TA: int32_t (Q4.27)
1890 */
1891template <int MIXTYPE,
1892        typename TO, typename TI, typename TV, typename TA, typename TAV>
1893static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
1894        const TI* in, TA* aux, const TV *vol, TAV vola)
1895{
1896    switch (channels) {
1897    case 1:
1898        volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
1899        break;
1900    case 2:
1901        volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
1902        break;
1903    case 3:
1904        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
1905        break;
1906    case 4:
1907        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
1908        break;
1909    case 5:
1910        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
1911        break;
1912    case 6:
1913        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
1914        break;
1915    case 7:
1916        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
1917        break;
1918    case 8:
1919        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
1920        break;
1921    }
1922}
1923
1924/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
1925 * USEFLOATVOL (set to true if float volume is used)
1926 * ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
1927 * TO: int32_t (Q4.27) or float
1928 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
1929 * TA: int32_t (Q4.27)
1930 */
1931template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
1932    typename TO, typename TI, typename TA>
1933void AudioMixer::volumeMix(TO *out, size_t outFrames,
1934        const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t)
1935{
1936    if (USEFLOATVOL) {
1937        if (ramp) {
1938            volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
1939                    t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);
1940            if (ADJUSTVOL) {
1941                t->adjustVolumeRamp(aux != NULL, true);
1942            }
1943        } else {
1944            volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
1945                    t->mVolume, t->auxLevel);
1946        }
1947    } else {
1948        if (ramp) {
1949            volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
1950                    t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
1951            if (ADJUSTVOL) {
1952                t->adjustVolumeRamp(aux != NULL);
1953            }
1954        } else {
1955            volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
1956                    t->volume, t->auxLevel);
1957        }
1958    }
1959}
1960
1961/* This process hook is called when there is a single track without
1962 * aux buffer, volume ramp, or resampling.
1963 * TODO: Update the hook selection: this can properly handle aux and ramp.
1964 *
1965 * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
1966 * TO: int32_t (Q4.27) or float
1967 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
1968 * TA: int32_t (Q4.27)
1969 */
1970template <int MIXTYPE, typename TO, typename TI, typename TA>
1971void AudioMixer::process_NoResampleOneTrack(state_t* state, int64_t pts)
1972{
1973    ALOGVV("process_NoResampleOneTrack\n");
1974    // CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
1975    const int i = 31 - __builtin_clz(state->enabledTracks);
1976    ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
1977    track_t *t = &state->tracks[i];
1978    const uint32_t channels = t->mMixerChannelCount;
1979    TO* out = reinterpret_cast<TO*>(t->mainBuffer);
1980    TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
1981    const bool ramp = t->needsRamp();
1982
1983    for (size_t numFrames = state->frameCount; numFrames; ) {
1984        AudioBufferProvider::Buffer& b(t->buffer);
1985        // get input buffer
1986        b.frameCount = numFrames;
1987        const int64_t outputPTS = calculateOutputPTS(*t, pts, state->frameCount - numFrames);
1988        t->bufferProvider->getNextBuffer(&b, outputPTS);
1989        const TI *in = reinterpret_cast<TI*>(b.raw);
1990
1991        // in == NULL can happen if the track was flushed just after having
1992        // been enabled for mixing.
1993        if (in == NULL || (((uintptr_t)in) & 3)) {
1994            memset(out, 0, numFrames
1995                    * channels * audio_bytes_per_sample(t->mMixerFormat));
1996            ALOGE_IF((((uintptr_t)in) & 3), "process_NoResampleOneTrack: bus error: "
1997                    "buffer %p track %p, channels %d, needs %#x",
1998                    in, t, t->channelCount, t->needs);
1999            return;
2000        }
2001
2002        const size_t outFrames = b.frameCount;
2003        volumeMix<MIXTYPE, is_same<TI, float>::value, false> (
2004                out, outFrames, in, aux, ramp, t);
2005
2006        out += outFrames * channels;
2007        if (aux != NULL) {
2008            aux += channels;
2009        }
2010        numFrames -= b.frameCount;
2011
2012        // release buffer
2013        t->bufferProvider->releaseBuffer(&b);
2014    }
2015    if (ramp) {
2016        t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
2017    }
2018}
2019
2020/* This track hook is called to do resampling then mixing,
2021 * pulling from the track's upstream AudioBufferProvider.
2022 *
2023 * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
2024 * TO: int32_t (Q4.27) or float
2025 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
2026 * TA: int32_t (Q4.27)
2027 */
2028template <int MIXTYPE, typename TO, typename TI, typename TA>
2029void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
2030{
2031    ALOGVV("track__Resample\n");
2032    t->resampler->setSampleRate(t->sampleRate);
2033    const bool ramp = t->needsRamp();
2034    if (ramp || aux != NULL) {
2035        // if ramp:        resample with unity gain to temp buffer and scale/mix in 2nd step.
2036        // if aux != NULL: resample with unity gain to temp buffer then apply send level.
2037
2038        t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
2039        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));
2040        t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
2041
2042        volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
2043                out, outFrameCount, temp, aux, ramp, t);
2044
2045    } else { // constant volume gain
2046        t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
2047        t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider);
2048    }
2049}
2050
2051/* This track hook is called to mix a track, when no resampling is required.
2052 * The input buffer should be present in t->in.
2053 *
2054 * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
2055 * TO: int32_t (Q4.27) or float
2056 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
2057 * TA: int32_t (Q4.27)
2058 */
2059template <int MIXTYPE, typename TO, typename TI, typename TA>
2060void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
2061        TO* temp __unused, TA* aux)
2062{
2063    ALOGVV("track__NoResample\n");
2064    const TI *in = static_cast<const TI *>(t->in);
2065
2066    volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
2067            out, frameCount, in, aux, t->needsRamp(), t);
2068
2069    // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
2070    // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
2071    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * t->mMixerChannelCount;
2072    t->in = in;
2073}
2074
2075/* The Mixer engine generates either int32_t (Q4_27) or float data.
2076 * We use this function to convert the engine buffers
2077 * to the desired mixer output format, either int16_t (Q.15) or float.
2078 */
2079void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
2080        void *in, audio_format_t mixerInFormat, size_t sampleCount)
2081{
2082    switch (mixerInFormat) {
2083    case AUDIO_FORMAT_PCM_FLOAT:
2084        switch (mixerOutFormat) {
2085        case AUDIO_FORMAT_PCM_FLOAT:
2086            memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
2087            break;
2088        case AUDIO_FORMAT_PCM_16_BIT:
2089            memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
2090            break;
2091        default:
2092            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
2093            break;
2094        }
2095        break;
2096    case AUDIO_FORMAT_PCM_16_BIT:
2097        switch (mixerOutFormat) {
2098        case AUDIO_FORMAT_PCM_FLOAT:
2099            memcpy_to_float_from_q4_27((float*)out, (int32_t*)in, sampleCount);
2100            break;
2101        case AUDIO_FORMAT_PCM_16_BIT:
2102            // two int16_t are produced per iteration
2103            ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1);
2104            break;
2105        default:
2106            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
2107            break;
2108        }
2109        break;
2110    default:
2111        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
2112        break;
2113    }
2114}
2115
2116/* Returns the proper track hook to use for mixing the track into the output buffer.
2117 */
2118AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount,
2119        audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
2120{
2121    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
2122        switch (trackType) {
2123        case TRACKTYPE_NOP:
2124            return track__nop;
2125        case TRACKTYPE_RESAMPLE:
2126            return track__genericResample;
2127        case TRACKTYPE_NORESAMPLEMONO:
2128            return track__16BitsMono;
2129        case TRACKTYPE_NORESAMPLE:
2130            return track__16BitsStereo;
2131        default:
2132            LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
2133            break;
2134        }
2135    }
2136    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
2137    switch (trackType) {
2138    case TRACKTYPE_NOP:
2139        return track__nop;
2140    case TRACKTYPE_RESAMPLE:
2141        switch (mixerInFormat) {
2142        case AUDIO_FORMAT_PCM_FLOAT:
2143            return (AudioMixer::hook_t)
2144                    track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>;
2145        case AUDIO_FORMAT_PCM_16_BIT:
2146            return (AudioMixer::hook_t)\
2147                    track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
2148        default:
2149            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
2150            break;
2151        }
2152        break;
2153    case TRACKTYPE_NORESAMPLEMONO:
2154        switch (mixerInFormat) {
2155        case AUDIO_FORMAT_PCM_FLOAT:
2156            return (AudioMixer::hook_t)
2157                    track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;
2158        case AUDIO_FORMAT_PCM_16_BIT:
2159            return (AudioMixer::hook_t)
2160                    track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;
2161        default:
2162            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
2163            break;
2164        }
2165        break;
2166    case TRACKTYPE_NORESAMPLE:
2167        switch (mixerInFormat) {
2168        case AUDIO_FORMAT_PCM_FLOAT:
2169            return (AudioMixer::hook_t)
2170                    track__NoResample<MIXTYPE_MULTI, float, float, int32_t>;
2171        case AUDIO_FORMAT_PCM_16_BIT:
2172            return (AudioMixer::hook_t)
2173                    track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
2174        default:
2175            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
2176            break;
2177        }
2178        break;
2179    default:
2180        LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
2181        break;
2182    }
2183    return NULL;
2184}
2185
2186/* Returns the proper process hook for mixing tracks. Currently works only for
2187 * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
2188 *
2189 * TODO: Due to the special mixing considerations of duplicating to
2190 * a stereo output track, the input track cannot be MONO.  This should be
2191 * prevented by the caller.
2192 */
2193AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t channelCount,
2194        audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
2195{
2196    if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
2197        LOG_ALWAYS_FATAL("bad processType: %d", processType);
2198        return NULL;
2199    }
2200    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
2201        return process__OneTrack16BitsStereoNoResampling;
2202    }
2203    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
2204    switch (mixerInFormat) {
2205    case AUDIO_FORMAT_PCM_FLOAT:
2206        switch (mixerOutFormat) {
2207        case AUDIO_FORMAT_PCM_FLOAT:
2208            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
2209                    float /*TO*/, float /*TI*/, int32_t /*TA*/>;
2210        case AUDIO_FORMAT_PCM_16_BIT:
2211            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
2212                    int16_t, float, int32_t>;
2213        default:
2214            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
2215            break;
2216        }
2217        break;
2218    case AUDIO_FORMAT_PCM_16_BIT:
2219        switch (mixerOutFormat) {
2220        case AUDIO_FORMAT_PCM_FLOAT:
2221            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
2222                    float, int16_t, int32_t>;
2223        case AUDIO_FORMAT_PCM_16_BIT:
2224            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
2225                    int16_t, int16_t, int32_t>;
2226        default:
2227            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
2228            break;
2229        }
2230        break;
2231    default:
2232        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
2233        break;
2234    }
2235    return NULL;
2236}
2237
2238// ----------------------------------------------------------------------------
2239}; // namespace android
2240