SoftAAC2.cpp revision 89334a07cb379c003f626471e4358d7cef99b00a 
1/*
2 * Copyright (C) 2012 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 *      http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17//#define LOG_NDEBUG 0
18#define LOG_TAG "SoftAAC2"
19#include <utils/Log.h>
20
21#include "SoftAAC2.h"
22#include <OMX_AudioExt.h>
23#include <OMX_IndexExt.h>
24
25#include <cutils/properties.h>
26#include <media/stagefright/foundation/ADebug.h>
27#include <media/stagefright/foundation/hexdump.h>
28#include <media/stagefright/MediaErrors.h>
29#include <utils/misc.h>
30
31#include <math.h>
32
33#define FILEREAD_MAX_LAYERS 2
34
35#define DRC_DEFAULT_MOBILE_REF_LEVEL 64  /* 64*-0.25dB = -16 dB below full scale for mobile conf */
36#define DRC_DEFAULT_MOBILE_DRC_CUT   127 /* maximum compression of dynamic range for mobile conf */
37#define DRC_DEFAULT_MOBILE_DRC_BOOST 127 /* maximum compression of dynamic range for mobile conf */
38#define DRC_DEFAULT_MOBILE_DRC_HEAVY 1   /* switch for heavy compression for mobile conf */
39#define DRC_DEFAULT_MOBILE_ENC_LEVEL -1 /* encoder target level; -1 => the value is unknown, otherwise dB step value (e.g. 64 for -16 dB) */
40#define MAX_CHANNEL_COUNT            8  /* maximum number of audio channels that can be decoded */
41// names of properties that can be used to override the default DRC settings
42#define PROP_DRC_OVERRIDE_REF_LEVEL  "aac_drc_reference_level"
43#define PROP_DRC_OVERRIDE_CUT        "aac_drc_cut"
44#define PROP_DRC_OVERRIDE_BOOST      "aac_drc_boost"
45#define PROP_DRC_OVERRIDE_HEAVY      "aac_drc_heavy"
46#define PROP_DRC_OVERRIDE_ENC_LEVEL "aac_drc_enc_target_level"
47
48namespace android {
49
50template<class T>
51static void InitOMXParams(T *params) {
52    params->nSize = sizeof(T);
53    params->nVersion.s.nVersionMajor = 1;
54    params->nVersion.s.nVersionMinor = 0;
55    params->nVersion.s.nRevision = 0;
56    params->nVersion.s.nStep = 0;
57}
58
59static const OMX_U32 kSupportedProfiles[] = {
60    OMX_AUDIO_AACObjectLC,
61    OMX_AUDIO_AACObjectHE,
62    OMX_AUDIO_AACObjectHE_PS,
63    OMX_AUDIO_AACObjectLD,
64    OMX_AUDIO_AACObjectELD,
65};
66
67SoftAAC2::SoftAAC2(
68        const char *name,
69        const OMX_CALLBACKTYPE *callbacks,
70        OMX_PTR appData,
71        OMX_COMPONENTTYPE **component)
72    : SimpleSoftOMXComponent(name, callbacks, appData, component),
73      mAACDecoder(NULL),
74      mStreamInfo(NULL),
75      mIsADTS(false),
76      mInputBufferCount(0),
77      mOutputBufferCount(0),
78      mSignalledError(false),
79      mLastInHeader(NULL),
80      mOutputPortSettingsChange(NONE) {
81    initPorts();
82    CHECK_EQ(initDecoder(), (status_t)OK);
83}
84
85SoftAAC2::~SoftAAC2() {
86    aacDecoder_Close(mAACDecoder);
87    delete[] mOutputDelayRingBuffer;
88}
89
90void SoftAAC2::initPorts() {
91    OMX_PARAM_PORTDEFINITIONTYPE def;
92    InitOMXParams(&def);
93
94    def.nPortIndex = 0;
95    def.eDir = OMX_DirInput;
96    def.nBufferCountMin = kNumInputBuffers;
97    def.nBufferCountActual = def.nBufferCountMin;
98    def.nBufferSize = 8192;
99    def.bEnabled = OMX_TRUE;
100    def.bPopulated = OMX_FALSE;
101    def.eDomain = OMX_PortDomainAudio;
102    def.bBuffersContiguous = OMX_FALSE;
103    def.nBufferAlignment = 1;
104
105    def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
106    def.format.audio.pNativeRender = NULL;
107    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
108    def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
109
110    addPort(def);
111
112    def.nPortIndex = 1;
113    def.eDir = OMX_DirOutput;
114    def.nBufferCountMin = kNumOutputBuffers;
115    def.nBufferCountActual = def.nBufferCountMin;
116    def.nBufferSize = 4096 * MAX_CHANNEL_COUNT;
117    def.bEnabled = OMX_TRUE;
118    def.bPopulated = OMX_FALSE;
119    def.eDomain = OMX_PortDomainAudio;
120    def.bBuffersContiguous = OMX_FALSE;
121    def.nBufferAlignment = 2;
122
123    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
124    def.format.audio.pNativeRender = NULL;
125    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
126    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
127
128    addPort(def);
129}
130
131status_t SoftAAC2::initDecoder() {
132    ALOGV("initDecoder()");
133    status_t status = UNKNOWN_ERROR;
134    mAACDecoder = aacDecoder_Open(TT_MP4_ADIF, /* num layers */ 1);
135    if (mAACDecoder != NULL) {
136        mStreamInfo = aacDecoder_GetStreamInfo(mAACDecoder);
137        if (mStreamInfo != NULL) {
138            status = OK;
139        }
140    }
141
142    mEndOfInput = false;
143    mEndOfOutput = false;
144    mOutputDelayCompensated = 0;
145    mOutputDelayRingBufferSize = 2048 * MAX_CHANNEL_COUNT * kNumDelayBlocksMax;
146    mOutputDelayRingBuffer = new short[mOutputDelayRingBufferSize];
147    mOutputDelayRingBufferWritePos = 0;
148    mOutputDelayRingBufferReadPos = 0;
149    mOutputDelayRingBufferFilled = 0;
150
151    if (mAACDecoder == NULL) {
152        ALOGE("AAC decoder is null. TODO: Can not call aacDecoder_SetParam in the following code");
153    }
154
155    //aacDecoder_SetParam(mAACDecoder, AAC_PCM_LIMITER_ENABLE, 0);
156
157    //init DRC wrapper
158    mDrcWrap.setDecoderHandle(mAACDecoder);
159    mDrcWrap.submitStreamData(mStreamInfo);
160
161    // for streams that contain metadata, use the mobile profile DRC settings unless overridden by platform properties
162    // TODO: change the DRC settings depending on audio output device type (HDMI, loadspeaker, headphone)
163    char value[PROPERTY_VALUE_MAX];
164    //  DRC_PRES_MODE_WRAP_DESIRED_TARGET
165    if (property_get(PROP_DRC_OVERRIDE_REF_LEVEL, value, NULL)) {
166        unsigned refLevel = atoi(value);
167        ALOGV("AAC decoder using desired DRC target reference level of %d instead of %d", refLevel,
168                DRC_DEFAULT_MOBILE_REF_LEVEL);
169        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, refLevel);
170    } else {
171        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, DRC_DEFAULT_MOBILE_REF_LEVEL);
172    }
173    //  DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR
174    if (property_get(PROP_DRC_OVERRIDE_CUT, value, NULL)) {
175        unsigned cut = atoi(value);
176        ALOGV("AAC decoder using desired DRC attenuation factor of %d instead of %d", cut,
177                DRC_DEFAULT_MOBILE_DRC_CUT);
178        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, cut);
179    } else {
180        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, DRC_DEFAULT_MOBILE_DRC_CUT);
181    }
182    //  DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR
183    if (property_get(PROP_DRC_OVERRIDE_BOOST, value, NULL)) {
184        unsigned boost = atoi(value);
185        ALOGV("AAC decoder using desired DRC boost factor of %d instead of %d", boost,
186                DRC_DEFAULT_MOBILE_DRC_BOOST);
187        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, boost);
188    } else {
189        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, DRC_DEFAULT_MOBILE_DRC_BOOST);
190    }
191    //  DRC_PRES_MODE_WRAP_DESIRED_HEAVY
192    if (property_get(PROP_DRC_OVERRIDE_HEAVY, value, NULL)) {
193        unsigned heavy = atoi(value);
194        ALOGV("AAC decoder using desried DRC heavy compression switch of %d instead of %d", heavy,
195                DRC_DEFAULT_MOBILE_DRC_HEAVY);
196        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, heavy);
197    } else {
198        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, DRC_DEFAULT_MOBILE_DRC_HEAVY);
199    }
200    // DRC_PRES_MODE_WRAP_ENCODER_TARGET
201    if (property_get(PROP_DRC_OVERRIDE_ENC_LEVEL, value, NULL)) {
202        unsigned encoderRefLevel = atoi(value);
203        ALOGV("AAC decoder using encoder-side DRC reference level of %d instead of %d",
204                encoderRefLevel, DRC_DEFAULT_MOBILE_ENC_LEVEL);
205        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, encoderRefLevel);
206    } else {
207        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, DRC_DEFAULT_MOBILE_ENC_LEVEL);
208    }
209
210    // By default, the decoder creates a 5.1 channel downmix signal.
211    // For seven and eight channel input streams, enable 6.1 and 7.1 channel output
212    aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1);
213
214    return status;
215}
216
217OMX_ERRORTYPE SoftAAC2::internalGetParameter(
218        OMX_INDEXTYPE index, OMX_PTR params) {
219    switch ((OMX_U32) index) {
220        case OMX_IndexParamAudioAac:
221        {
222            OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
223                (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
224
225            if (!isValidOMXParam(aacParams)) {
226                return OMX_ErrorBadParameter;
227            }
228
229            if (aacParams->nPortIndex != 0) {
230                return OMX_ErrorUndefined;
231            }
232
233            aacParams->nBitRate = 0;
234            aacParams->nAudioBandWidth = 0;
235            aacParams->nAACtools = 0;
236            aacParams->nAACERtools = 0;
237            aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
238
239            aacParams->eAACStreamFormat =
240                mIsADTS
241                    ? OMX_AUDIO_AACStreamFormatMP4ADTS
242                    : OMX_AUDIO_AACStreamFormatMP4FF;
243
244            aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
245
246            if (!isConfigured()) {
247                aacParams->nChannels = 1;
248                aacParams->nSampleRate = 44100;
249                aacParams->nFrameLength = 0;
250            } else {
251                aacParams->nChannels = mStreamInfo->numChannels;
252                aacParams->nSampleRate = mStreamInfo->sampleRate;
253                aacParams->nFrameLength = mStreamInfo->frameSize;
254            }
255
256            return OMX_ErrorNone;
257        }
258
259        case OMX_IndexParamAudioPcm:
260        {
261            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
262                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
263
264            if (!isValidOMXParam(pcmParams)) {
265                return OMX_ErrorBadParameter;
266            }
267
268            if (pcmParams->nPortIndex != 1) {
269                return OMX_ErrorUndefined;
270            }
271
272            pcmParams->eNumData = OMX_NumericalDataSigned;
273            pcmParams->eEndian = OMX_EndianBig;
274            pcmParams->bInterleaved = OMX_TRUE;
275            pcmParams->nBitPerSample = 16;
276            pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
277            pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
278            pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
279            pcmParams->eChannelMapping[2] = OMX_AUDIO_ChannelCF;
280            pcmParams->eChannelMapping[3] = OMX_AUDIO_ChannelLFE;
281            pcmParams->eChannelMapping[4] = OMX_AUDIO_ChannelLS;
282            pcmParams->eChannelMapping[5] = OMX_AUDIO_ChannelRS;
283
284            if (!isConfigured()) {
285                pcmParams->nChannels = 1;
286                pcmParams->nSamplingRate = 44100;
287            } else {
288                pcmParams->nChannels = mStreamInfo->numChannels;
289                pcmParams->nSamplingRate = mStreamInfo->sampleRate;
290            }
291
292            return OMX_ErrorNone;
293        }
294
295        case OMX_IndexParamAudioProfileQuerySupported:
296        {
297            OMX_AUDIO_PARAM_ANDROID_PROFILETYPE *profileParams =
298                (OMX_AUDIO_PARAM_ANDROID_PROFILETYPE *)params;
299
300            if (!isValidOMXParam(profileParams)) {
301                return OMX_ErrorBadParameter;
302            }
303
304            if (profileParams->nPortIndex != 0) {
305                return OMX_ErrorUndefined;
306            }
307
308            if (profileParams->nProfileIndex >= NELEM(kSupportedProfiles)) {
309                return OMX_ErrorNoMore;
310            }
311
312            profileParams->eProfile =
313                kSupportedProfiles[profileParams->nProfileIndex];
314
315            return OMX_ErrorNone;
316        }
317
318        default:
319            return SimpleSoftOMXComponent::internalGetParameter(index, params);
320    }
321}
322
323OMX_ERRORTYPE SoftAAC2::internalSetParameter(
324        OMX_INDEXTYPE index, const OMX_PTR params) {
325    switch ((int)index) {
326        case OMX_IndexParamStandardComponentRole:
327        {
328            const OMX_PARAM_COMPONENTROLETYPE *roleParams =
329                (const OMX_PARAM_COMPONENTROLETYPE *)params;
330
331            if (!isValidOMXParam(roleParams)) {
332                return OMX_ErrorBadParameter;
333            }
334
335            if (strncmp((const char *)roleParams->cRole,
336                        "audio_decoder.aac",
337                        OMX_MAX_STRINGNAME_SIZE - 1)) {
338                return OMX_ErrorUndefined;
339            }
340
341            return OMX_ErrorNone;
342        }
343
344        case OMX_IndexParamAudioAac:
345        {
346            const OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
347                (const OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
348
349            if (!isValidOMXParam(aacParams)) {
350                return OMX_ErrorBadParameter;
351            }
352
353            if (aacParams->nPortIndex != 0) {
354                return OMX_ErrorUndefined;
355            }
356
357            if (aacParams->eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4FF) {
358                mIsADTS = false;
359            } else if (aacParams->eAACStreamFormat
360                        == OMX_AUDIO_AACStreamFormatMP4ADTS) {
361                mIsADTS = true;
362            } else {
363                return OMX_ErrorUndefined;
364            }
365
366            return OMX_ErrorNone;
367        }
368
369        case OMX_IndexParamAudioAndroidAacPresentation:
370        {
371            const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *aacPresParams =
372                    (const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *)params;
373
374            if (!isValidOMXParam(aacPresParams)) {
375                return OMX_ErrorBadParameter;
376            }
377
378            // for the following parameters of the OMX_AUDIO_PARAM_AACPROFILETYPE structure,
379            // a value of -1 implies the parameter is not set by the application:
380            //   nMaxOutputChannels     -1 by default
381            //   nDrcCut                uses default platform properties, see initDecoder()
382            //   nDrcBoost                idem
383            //   nHeavyCompression        idem
384            //   nTargetReferenceLevel    idem
385            //   nEncodedTargetLevel      idem
386            if (aacPresParams->nMaxOutputChannels >= 0) {
387                int max;
388                if (aacPresParams->nMaxOutputChannels >= 8) { max = 8; }
389                else if (aacPresParams->nMaxOutputChannels >= 6) { max = 6; }
390                else if (aacPresParams->nMaxOutputChannels >= 2) { max = 2; }
391                else {
392                    // -1 or 0: disable downmix,  1: mono
393                    max = aacPresParams->nMaxOutputChannels;
394                }
395                ALOGV("set nMaxOutputChannels=%d", max);
396                aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, max);
397            }
398            bool updateDrcWrapper = false;
399            if (aacPresParams->nDrcBoost >= 0) {
400                ALOGV("set nDrcBoost=%d", aacPresParams->nDrcBoost);
401                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR,
402                        aacPresParams->nDrcBoost);
403                updateDrcWrapper = true;
404            }
405            if (aacPresParams->nDrcCut >= 0) {
406                ALOGV("set nDrcCut=%d", aacPresParams->nDrcCut);
407                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, aacPresParams->nDrcCut);
408                updateDrcWrapper = true;
409            }
410            if (aacPresParams->nHeavyCompression >= 0) {
411                ALOGV("set nHeavyCompression=%d", aacPresParams->nHeavyCompression);
412                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY,
413                        aacPresParams->nHeavyCompression);
414                updateDrcWrapper = true;
415            }
416            if (aacPresParams->nTargetReferenceLevel >= 0) {
417                ALOGV("set nTargetReferenceLevel=%d", aacPresParams->nTargetReferenceLevel);
418                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET,
419                        aacPresParams->nTargetReferenceLevel);
420                updateDrcWrapper = true;
421            }
422            if (aacPresParams->nEncodedTargetLevel >= 0) {
423                ALOGV("set nEncodedTargetLevel=%d", aacPresParams->nEncodedTargetLevel);
424                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET,
425                        aacPresParams->nEncodedTargetLevel);
426                updateDrcWrapper = true;
427            }
428            if (aacPresParams->nPCMLimiterEnable >= 0) {
429                aacDecoder_SetParam(mAACDecoder, AAC_PCM_LIMITER_ENABLE,
430                        (aacPresParams->nPCMLimiterEnable != 0));
431            }
432            if (updateDrcWrapper) {
433                mDrcWrap.update();
434            }
435
436            return OMX_ErrorNone;
437        }
438
439        case OMX_IndexParamAudioPcm:
440        {
441            const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
442                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
443
444            if (!isValidOMXParam(pcmParams)) {
445                return OMX_ErrorBadParameter;
446            }
447
448            if (pcmParams->nPortIndex != 1) {
449                return OMX_ErrorUndefined;
450            }
451
452            return OMX_ErrorNone;
453        }
454
455        default:
456            return SimpleSoftOMXComponent::internalSetParameter(index, params);
457    }
458}
459
460bool SoftAAC2::isConfigured() const {
461    return mInputBufferCount > 0;
462}
463
464bool SoftAAC2::outputDelayRingBufferPutSamples(INT_PCM *samples, int32_t numSamples) {
465    if (numSamples == 0) {
466        return true;
467    }
468    if (outputDelayRingBufferSpaceLeft() < numSamples) {
469        ALOGE("RING BUFFER WOULD OVERFLOW");
470        return false;
471    }
472    if (mOutputDelayRingBufferWritePos + numSamples <= mOutputDelayRingBufferSize
473            && (mOutputDelayRingBufferReadPos <= mOutputDelayRingBufferWritePos
474                    || mOutputDelayRingBufferReadPos > mOutputDelayRingBufferWritePos + numSamples)) {
475        // faster memcopy loop without checks, if the preconditions allow this
476        for (int32_t i = 0; i < numSamples; i++) {
477            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos++] = samples[i];
478        }
479
480        if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
481            mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
482        }
483    } else {
484        ALOGV("slow SoftAAC2::outputDelayRingBufferPutSamples()");
485
486        for (int32_t i = 0; i < numSamples; i++) {
487            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos] = samples[i];
488            mOutputDelayRingBufferWritePos++;
489            if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
490                mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
491            }
492        }
493    }
494    mOutputDelayRingBufferFilled += numSamples;
495    return true;
496}
497
498int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numSamples) {
499
500    if (numSamples > mOutputDelayRingBufferFilled) {
501        ALOGE("RING BUFFER WOULD UNDERRUN");
502        return -1;
503    }
504
505    if (mOutputDelayRingBufferReadPos + numSamples <= mOutputDelayRingBufferSize
506            && (mOutputDelayRingBufferWritePos < mOutputDelayRingBufferReadPos
507                    || mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferReadPos + numSamples)) {
508        // faster memcopy loop without checks, if the preconditions allow this
509        if (samples != 0) {
510            for (int32_t i = 0; i < numSamples; i++) {
511                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos++];
512            }
513        } else {
514            mOutputDelayRingBufferReadPos += numSamples;
515        }
516        if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
517            mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
518        }
519    } else {
520        ALOGV("slow SoftAAC2::outputDelayRingBufferGetSamples()");
521
522        for (int32_t i = 0; i < numSamples; i++) {
523            if (samples != 0) {
524                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos];
525            }
526            mOutputDelayRingBufferReadPos++;
527            if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
528                mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
529            }
530        }
531    }
532    mOutputDelayRingBufferFilled -= numSamples;
533    return numSamples;
534}
535
536int32_t SoftAAC2::outputDelayRingBufferSamplesAvailable() {
537    return mOutputDelayRingBufferFilled;
538}
539
540int32_t SoftAAC2::outputDelayRingBufferSpaceLeft() {
541    return mOutputDelayRingBufferSize - outputDelayRingBufferSamplesAvailable();
542}
543
544
545void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
546    if (mSignalledError || mOutputPortSettingsChange != NONE) {
547        return;
548    }
549
550    UCHAR* inBuffer[FILEREAD_MAX_LAYERS];
551    UINT inBufferLength[FILEREAD_MAX_LAYERS] = {0};
552    UINT bytesValid[FILEREAD_MAX_LAYERS] = {0};
553
554    List<BufferInfo *> &inQueue = getPortQueue(0);
555    List<BufferInfo *> &outQueue = getPortQueue(1);
556
557    while ((!inQueue.empty() || mEndOfInput) && !outQueue.empty()) {
558        if (!inQueue.empty()) {
559            INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
560            BufferInfo *inInfo = *inQueue.begin();
561            OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
562
563            mEndOfInput = (inHeader->nFlags & OMX_BUFFERFLAG_EOS) != 0;
564
565            if (mInputBufferCount == 0 && !(inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
566                ALOGE("first buffer should have OMX_BUFFERFLAG_CODECCONFIG set");
567                inHeader->nFlags |= OMX_BUFFERFLAG_CODECCONFIG;
568            }
569            if ((inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG) != 0) {
570                BufferInfo *inInfo = *inQueue.begin();
571                OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
572
573                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
574                inBufferLength[0] = inHeader->nFilledLen;
575
576                AAC_DECODER_ERROR decoderErr =
577                    aacDecoder_ConfigRaw(mAACDecoder,
578                                         inBuffer,
579                                         inBufferLength);
580
581                if (decoderErr != AAC_DEC_OK) {
582                    ALOGW("aacDecoder_ConfigRaw decoderErr = 0x%4.4x", decoderErr);
583                    mSignalledError = true;
584                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
585                    return;
586                }
587
588                mInputBufferCount++;
589                mOutputBufferCount++; // fake increase of outputBufferCount to keep the counters aligned
590
591                inInfo->mOwnedByUs = false;
592                inQueue.erase(inQueue.begin());
593                mLastInHeader = NULL;
594                inInfo = NULL;
595                notifyEmptyBufferDone(inHeader);
596                inHeader = NULL;
597
598                // Only send out port settings changed event if both sample rate
599                // and numChannels are valid.
600                if (mStreamInfo->sampleRate && mStreamInfo->numChannels) {
601                    ALOGI("Initially configuring decoder: %d Hz, %d channels",
602                        mStreamInfo->sampleRate,
603                        mStreamInfo->numChannels);
604
605                    notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
606                    mOutputPortSettingsChange = AWAITING_DISABLED;
607                }
608                return;
609            }
610
611            if (inHeader->nFilledLen == 0) {
612                inInfo->mOwnedByUs = false;
613                inQueue.erase(inQueue.begin());
614                mLastInHeader = NULL;
615                inInfo = NULL;
616                notifyEmptyBufferDone(inHeader);
617                inHeader = NULL;
618                continue;
619            }
620
621            if (mIsADTS) {
622                size_t adtsHeaderSize = 0;
623                // skip 30 bits, aac_frame_length follows.
624                // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
625
626                const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
627
628                bool signalError = false;
629                if (inHeader->nFilledLen < 7) {
630                    ALOGE("Audio data too short to contain even the ADTS header. "
631                            "Got %d bytes.", inHeader->nFilledLen);
632                    hexdump(adtsHeader, inHeader->nFilledLen);
633                    signalError = true;
634                } else {
635                    bool protectionAbsent = (adtsHeader[1] & 1);
636
637                    unsigned aac_frame_length =
638                        ((adtsHeader[3] & 3) << 11)
639                        | (adtsHeader[4] << 3)
640                        | (adtsHeader[5] >> 5);
641
642                    if (inHeader->nFilledLen < aac_frame_length) {
643                        ALOGE("Not enough audio data for the complete frame. "
644                                "Got %d bytes, frame size according to the ADTS "
645                                "header is %u bytes.",
646                                inHeader->nFilledLen, aac_frame_length);
647                        hexdump(adtsHeader, inHeader->nFilledLen);
648                        signalError = true;
649                    } else {
650                        adtsHeaderSize = (protectionAbsent ? 7 : 9);
651
652                        inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
653                        inBufferLength[0] = aac_frame_length - adtsHeaderSize;
654
655                        inHeader->nOffset += adtsHeaderSize;
656                        inHeader->nFilledLen -= adtsHeaderSize;
657                    }
658                }
659
660                if (signalError) {
661                    mSignalledError = true;
662                    notify(OMX_EventError, OMX_ErrorStreamCorrupt, ERROR_MALFORMED, NULL);
663                    return;
664                }
665
666                // insert buffer size and time stamp
667                mBufferSizes.add(inBufferLength[0]);
668                if (mLastInHeader != inHeader) {
669                    mBufferTimestamps.add(inHeader->nTimeStamp);
670                    mLastInHeader = inHeader;
671                } else {
672                    int64_t currentTime = mBufferTimestamps.top();
673                    currentTime += mStreamInfo->aacSamplesPerFrame *
674                            1000000ll / mStreamInfo->aacSampleRate;
675                    mBufferTimestamps.add(currentTime);
676                }
677            } else {
678                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
679                inBufferLength[0] = inHeader->nFilledLen;
680                mLastInHeader = inHeader;
681                mBufferTimestamps.add(inHeader->nTimeStamp);
682                mBufferSizes.add(inHeader->nFilledLen);
683            }
684
685            // Fill and decode
686            bytesValid[0] = inBufferLength[0];
687
688            INT prevSampleRate = mStreamInfo->sampleRate;
689            INT prevNumChannels = mStreamInfo->numChannels;
690
691            aacDecoder_Fill(mAACDecoder,
692                            inBuffer,
693                            inBufferLength,
694                            bytesValid);
695
696            // run DRC check
697            mDrcWrap.submitStreamData(mStreamInfo);
698            mDrcWrap.update();
699
700            UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
701            inHeader->nFilledLen -= inBufferUsedLength;
702            inHeader->nOffset += inBufferUsedLength;
703
704            AAC_DECODER_ERROR decoderErr;
705            int numLoops = 0;
706            do {
707                if (outputDelayRingBufferSpaceLeft() <
708                        (mStreamInfo->frameSize * mStreamInfo->numChannels)) {
709                    ALOGV("skipping decode: not enough space left in ringbuffer");
710                    break;
711                }
712
713                int numConsumed = mStreamInfo->numTotalBytes;
714                decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
715                                           tmpOutBuffer,
716                                           2048 * MAX_CHANNEL_COUNT,
717                                           0 /* flags */);
718
719                numConsumed = mStreamInfo->numTotalBytes - numConsumed;
720                numLoops++;
721
722                if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
723                    break;
724                }
725                mDecodedSizes.add(numConsumed);
726
727                if (decoderErr != AAC_DEC_OK) {
728                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
729                }
730
731                if (bytesValid[0] != 0) {
732                    ALOGE("bytesValid[0] != 0 should never happen");
733                    mSignalledError = true;
734                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
735                    return;
736                }
737
738                size_t numOutBytes =
739                    mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
740
741                if (decoderErr == AAC_DEC_OK) {
742                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
743                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
744                        mSignalledError = true;
745                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
746                        return;
747                    }
748                } else {
749                    ALOGW("AAC decoder returned error 0x%4.4x, substituting silence", decoderErr);
750
751                    memset(tmpOutBuffer, 0, numOutBytes); // TODO: check for overflow
752
753                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
754                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
755                        mSignalledError = true;
756                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
757                        return;
758                    }
759
760                    // Discard input buffer.
761                    if (inHeader) {
762                        inHeader->nFilledLen = 0;
763                    }
764
765                    aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
766
767                    // After an error, replace the last entry in mBufferSizes with the sum of the
768                    // last <numLoops> entries from mDecodedSizes to resynchronize the in/out lists.
769                    mBufferSizes.pop();
770                    int n = 0;
771                    for (int i = 0; i < numLoops; i++) {
772                        n += mDecodedSizes.itemAt(mDecodedSizes.size() - numLoops + i);
773                    }
774                    mBufferSizes.add(n);
775
776                    // fall through
777                }
778
779                /*
780                 * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
781                 * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
782                 * rate system and the sampling rate in the final output is actually
783                 * doubled compared with the core AAC decoder sampling rate.
784                 *
785                 * Explicit signalling is done by explicitly defining SBR audio object
786                 * type in the bitstream. Implicit signalling is done by embedding
787                 * SBR content in AAC extension payload specific to SBR, and hence
788                 * requires an AAC decoder to perform pre-checks on actual audio frames.
789                 *
790                 * Thus, we could not say for sure whether a stream is
791                 * AAC+/eAAC+ until the first data frame is decoded.
792                 */
793                if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
794                    if ((mInputBufferCount > 2) && (mOutputBufferCount <= 1)) {
795                        ALOGW("Invalid AAC stream");
796                        mSignalledError = true;
797                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
798                        return;
799                    }
800                } else if ((mStreamInfo->sampleRate != prevSampleRate) ||
801                           (mStreamInfo->numChannels != prevNumChannels)) {
802                    ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
803                          prevSampleRate, mStreamInfo->sampleRate,
804                          prevNumChannels, mStreamInfo->numChannels);
805
806                    notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
807                    mOutputPortSettingsChange = AWAITING_DISABLED;
808
809                    if (inHeader && inHeader->nFilledLen == 0) {
810                        inInfo->mOwnedByUs = false;
811                        mInputBufferCount++;
812                        inQueue.erase(inQueue.begin());
813                        mLastInHeader = NULL;
814                        inInfo = NULL;
815                        notifyEmptyBufferDone(inHeader);
816                        inHeader = NULL;
817                    }
818                    return;
819                }
820                if (inHeader && inHeader->nFilledLen == 0) {
821                    inInfo->mOwnedByUs = false;
822                    mInputBufferCount++;
823                    inQueue.erase(inQueue.begin());
824                    mLastInHeader = NULL;
825                    inInfo = NULL;
826                    notifyEmptyBufferDone(inHeader);
827                    inHeader = NULL;
828                } else {
829                    ALOGV("inHeader->nFilledLen = %d", inHeader ? inHeader->nFilledLen : 0);
830                }
831            } while (decoderErr == AAC_DEC_OK);
832        }
833
834        int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
835
836        if (!mEndOfInput && mOutputDelayCompensated < outputDelay) {
837            // discard outputDelay at the beginning
838            int32_t toCompensate = outputDelay - mOutputDelayCompensated;
839            int32_t discard = outputDelayRingBufferSamplesAvailable();
840            if (discard > toCompensate) {
841                discard = toCompensate;
842            }
843            int32_t discarded = outputDelayRingBufferGetSamples(0, discard);
844            mOutputDelayCompensated += discarded;
845            continue;
846        }
847
848        if (mEndOfInput) {
849            while (mOutputDelayCompensated > 0) {
850                // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
851                INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
852
853                // run DRC check
854                mDrcWrap.submitStreamData(mStreamInfo);
855                mDrcWrap.update();
856
857                AAC_DECODER_ERROR decoderErr =
858                    aacDecoder_DecodeFrame(mAACDecoder,
859                                           tmpOutBuffer,
860                                           2048 * MAX_CHANNEL_COUNT,
861                                           AACDEC_FLUSH);
862                if (decoderErr != AAC_DEC_OK) {
863                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
864                }
865
866                int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
867                if (tmpOutBufferSamples > mOutputDelayCompensated) {
868                    tmpOutBufferSamples = mOutputDelayCompensated;
869                }
870                outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
871                mOutputDelayCompensated -= tmpOutBufferSamples;
872            }
873        }
874
875        while (!outQueue.empty()
876                && outputDelayRingBufferSamplesAvailable()
877                        >= mStreamInfo->frameSize * mStreamInfo->numChannels) {
878            BufferInfo *outInfo = *outQueue.begin();
879            OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
880
881            if (outHeader->nOffset != 0) {
882                ALOGE("outHeader->nOffset != 0 is not handled");
883                mSignalledError = true;
884                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
885                return;
886            }
887
888            INT_PCM *outBuffer =
889                    reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);
890            int samplesize = mStreamInfo->numChannels * sizeof(int16_t);
891            if (outHeader->nOffset
892                    + mStreamInfo->frameSize * samplesize
893                    > outHeader->nAllocLen) {
894                ALOGE("buffer overflow");
895                mSignalledError = true;
896                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
897                return;
898
899            }
900
901            int available = outputDelayRingBufferSamplesAvailable();
902            int numSamples = outHeader->nAllocLen / sizeof(int16_t);
903            if (numSamples > available) {
904                numSamples = available;
905            }
906            int64_t currentTime = 0;
907            if (available) {
908
909                int numFrames = numSamples / (mStreamInfo->frameSize * mStreamInfo->numChannels);
910                numSamples = numFrames * (mStreamInfo->frameSize * mStreamInfo->numChannels);
911
912                ALOGV("%d samples available (%d), or %d frames",
913                        numSamples, available, numFrames);
914                int64_t *nextTimeStamp = &mBufferTimestamps.editItemAt(0);
915                currentTime = *nextTimeStamp;
916                int32_t *currentBufLeft = &mBufferSizes.editItemAt(0);
917                for (int i = 0; i < numFrames; i++) {
918                    int32_t decodedSize = mDecodedSizes.itemAt(0);
919                    mDecodedSizes.removeAt(0);
920                    ALOGV("decoded %d of %d", decodedSize, *currentBufLeft);
921                    if (*currentBufLeft > decodedSize) {
922                        // adjust/interpolate next time stamp
923                        *currentBufLeft -= decodedSize;
924                        *nextTimeStamp += mStreamInfo->aacSamplesPerFrame *
925                                1000000ll / mStreamInfo->aacSampleRate;
926                        ALOGV("adjusted nextTimeStamp/size to %lld/%d",
927                                (long long) *nextTimeStamp, *currentBufLeft);
928                    } else {
929                        // move to next timestamp in list
930                        if (mBufferTimestamps.size() > 0) {
931                            mBufferTimestamps.removeAt(0);
932                            nextTimeStamp = &mBufferTimestamps.editItemAt(0);
933                            mBufferSizes.removeAt(0);
934                            currentBufLeft = &mBufferSizes.editItemAt(0);
935                            ALOGV("moved to next time/size: %lld/%d",
936                                    (long long) *nextTimeStamp, *currentBufLeft);
937                        }
938                        // try to limit output buffer size to match input buffers
939                        // (e.g when an input buffer contained 4 "sub" frames, output
940                        // at most 4 decoded units in the corresponding output buffer)
941                        // This is optional. Remove the next three lines to fill the output
942                        // buffer with as many units as available.
943                        numFrames = i + 1;
944                        numSamples = numFrames * mStreamInfo->frameSize * mStreamInfo->numChannels;
945                        break;
946                    }
947                }
948
949                ALOGV("getting %d from ringbuffer", numSamples);
950                int32_t ns = outputDelayRingBufferGetSamples(outBuffer, numSamples);
951                if (ns != numSamples) {
952                    ALOGE("not a complete frame of samples available");
953                    mSignalledError = true;
954                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
955                    return;
956                }
957            }
958
959            outHeader->nFilledLen = numSamples * sizeof(int16_t);
960
961            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
962                outHeader->nFlags = OMX_BUFFERFLAG_EOS;
963                mEndOfOutput = true;
964            } else {
965                outHeader->nFlags = 0;
966            }
967
968            outHeader->nTimeStamp = currentTime;
969
970            mOutputBufferCount++;
971            outInfo->mOwnedByUs = false;
972            outQueue.erase(outQueue.begin());
973            outInfo = NULL;
974            ALOGV("out timestamp %lld / %d", outHeader->nTimeStamp, outHeader->nFilledLen);
975            notifyFillBufferDone(outHeader);
976            outHeader = NULL;
977        }
978
979        if (mEndOfInput) {
980            int ringBufAvail = outputDelayRingBufferSamplesAvailable();
981            if (!outQueue.empty()
982                    && ringBufAvail < mStreamInfo->frameSize * mStreamInfo->numChannels) {
983                if (!mEndOfOutput) {
984                    // send partial or empty block signaling EOS
985                    mEndOfOutput = true;
986                    BufferInfo *outInfo = *outQueue.begin();
987                    OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
988
989                    INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer
990                            + outHeader->nOffset);
991                    int32_t ns = outputDelayRingBufferGetSamples(outBuffer, ringBufAvail);
992                    if (ns < 0) {
993                        ns = 0;
994                    }
995                    outHeader->nFilledLen = ns;
996                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
997
998                    outHeader->nTimeStamp = mBufferTimestamps.itemAt(0);
999                    mBufferTimestamps.clear();
1000                    mBufferSizes.clear();
1001                    mDecodedSizes.clear();
1002
1003                    mOutputBufferCount++;
1004                    outInfo->mOwnedByUs = false;
1005                    outQueue.erase(outQueue.begin());
1006                    outInfo = NULL;
1007                    notifyFillBufferDone(outHeader);
1008                    outHeader = NULL;
1009                }
1010                break; // if outQueue not empty but no more output
1011            }
1012        }
1013    }
1014}
1015
1016void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
1017    if (portIndex == 0) {
1018        // Make sure that the next buffer output does not still
1019        // depend on fragments from the last one decoded.
1020        // drain all existing data
1021        drainDecoder();
1022        mBufferTimestamps.clear();
1023        mBufferSizes.clear();
1024        mDecodedSizes.clear();
1025        mLastInHeader = NULL;
1026        mEndOfInput = false;
1027    } else {
1028        int avail;
1029        while ((avail = outputDelayRingBufferSamplesAvailable()) > 0) {
1030            if (avail > mStreamInfo->frameSize * mStreamInfo->numChannels) {
1031                avail = mStreamInfo->frameSize * mStreamInfo->numChannels;
1032            }
1033            int32_t ns = outputDelayRingBufferGetSamples(0, avail);
1034            if (ns != avail) {
1035                ALOGW("not a complete frame of samples available");
1036                break;
1037            }
1038            mOutputBufferCount++;
1039        }
1040        mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
1041        mEndOfOutput = false;
1042    }
1043}
1044
1045void SoftAAC2::drainDecoder() {
1046    // flush decoder until outputDelay is compensated
1047    while (mOutputDelayCompensated > 0) {
1048        // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
1049        INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
1050
1051        // run DRC check
1052        mDrcWrap.submitStreamData(mStreamInfo);
1053        mDrcWrap.update();
1054
1055        AAC_DECODER_ERROR decoderErr =
1056            aacDecoder_DecodeFrame(mAACDecoder,
1057                                   tmpOutBuffer,
1058                                   2048 * MAX_CHANNEL_COUNT,
1059                                   AACDEC_FLUSH);
1060        if (decoderErr != AAC_DEC_OK) {
1061            ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
1062        }
1063
1064        int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
1065        if (tmpOutBufferSamples > mOutputDelayCompensated) {
1066            tmpOutBufferSamples = mOutputDelayCompensated;
1067        }
1068        outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
1069
1070        mOutputDelayCompensated -= tmpOutBufferSamples;
1071    }
1072}
1073
1074void SoftAAC2::onReset() {
1075    drainDecoder();
1076    // reset the "configured" state
1077    mInputBufferCount = 0;
1078    mOutputBufferCount = 0;
1079    mOutputDelayCompensated = 0;
1080    mOutputDelayRingBufferWritePos = 0;
1081    mOutputDelayRingBufferReadPos = 0;
1082    mOutputDelayRingBufferFilled = 0;
1083    mEndOfInput = false;
1084    mEndOfOutput = false;
1085    mBufferTimestamps.clear();
1086    mBufferSizes.clear();
1087    mDecodedSizes.clear();
1088    mLastInHeader = NULL;
1089
1090    // To make the codec behave the same before and after a reset, we need to invalidate the
1091    // streaminfo struct. This does that:
1092    mStreamInfo->sampleRate = 0; // TODO: mStreamInfo is read only
1093
1094    mSignalledError = false;
1095    mOutputPortSettingsChange = NONE;
1096}
1097
1098void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
1099    if (portIndex != 1) {
1100        return;
1101    }
1102
1103    switch (mOutputPortSettingsChange) {
1104        case NONE:
1105            break;
1106
1107        case AWAITING_DISABLED:
1108        {
1109            CHECK(!enabled);
1110            mOutputPortSettingsChange = AWAITING_ENABLED;
1111            break;
1112        }
1113
1114        default:
1115        {
1116            CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
1117            CHECK(enabled);
1118            mOutputPortSettingsChange = NONE;
1119            break;
1120        }
1121    }
1122}
1123
1124}  // namespace android
1125
1126android::SoftOMXComponent *createSoftOMXComponent(
1127        const char *name, const OMX_CALLBACKTYPE *callbacks,
1128        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
1129    return new android::SoftAAC2(name, callbacks, appData, component);
1130}
1131