SoftAAC2.cpp revision fa20a1db4be377a004efd756887f8b212e31d670 
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
30#include <math.h>
31
32#define FILEREAD_MAX_LAYERS 2
33
34#define DRC_DEFAULT_MOBILE_REF_LEVEL 64  /* 64*-0.25dB = -16 dB below full scale for mobile conf */
35#define DRC_DEFAULT_MOBILE_DRC_CUT   127 /* maximum compression of dynamic range for mobile conf */
36#define DRC_DEFAULT_MOBILE_DRC_BOOST 127 /* maximum compression of dynamic range for mobile conf */
37#define DRC_DEFAULT_MOBILE_DRC_HEAVY 1   /* switch for heavy compression for mobile conf */
38#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) */
39#define MAX_CHANNEL_COUNT            8  /* maximum number of audio channels that can be decoded */
40// names of properties that can be used to override the default DRC settings
41#define PROP_DRC_OVERRIDE_REF_LEVEL  "aac_drc_reference_level"
42#define PROP_DRC_OVERRIDE_CUT        "aac_drc_cut"
43#define PROP_DRC_OVERRIDE_BOOST      "aac_drc_boost"
44#define PROP_DRC_OVERRIDE_HEAVY      "aac_drc_heavy"
45#define PROP_DRC_OVERRIDE_ENC_LEVEL "aac_drc_enc_target_level"
46
47namespace android {
48
49template<class T>
50static void InitOMXParams(T *params) {
51    params->nSize = sizeof(T);
52    params->nVersion.s.nVersionMajor = 1;
53    params->nVersion.s.nVersionMinor = 0;
54    params->nVersion.s.nRevision = 0;
55    params->nVersion.s.nStep = 0;
56}
57
58SoftAAC2::SoftAAC2(
59        const char *name,
60        const OMX_CALLBACKTYPE *callbacks,
61        OMX_PTR appData,
62        OMX_COMPONENTTYPE **component)
63    : SimpleSoftOMXComponent(name, callbacks, appData, component),
64      mAACDecoder(NULL),
65      mStreamInfo(NULL),
66      mIsADTS(false),
67      mInputBufferCount(0),
68      mOutputBufferCount(0),
69      mSignalledError(false),
70      mLastInHeader(NULL),
71      mOutputPortSettingsChange(NONE) {
72    initPorts();
73    CHECK_EQ(initDecoder(), (status_t)OK);
74}
75
76SoftAAC2::~SoftAAC2() {
77    aacDecoder_Close(mAACDecoder);
78    delete mOutputDelayRingBuffer;
79}
80
81void SoftAAC2::initPorts() {
82    OMX_PARAM_PORTDEFINITIONTYPE def;
83    InitOMXParams(&def);
84
85    def.nPortIndex = 0;
86    def.eDir = OMX_DirInput;
87    def.nBufferCountMin = kNumInputBuffers;
88    def.nBufferCountActual = def.nBufferCountMin;
89    def.nBufferSize = 8192;
90    def.bEnabled = OMX_TRUE;
91    def.bPopulated = OMX_FALSE;
92    def.eDomain = OMX_PortDomainAudio;
93    def.bBuffersContiguous = OMX_FALSE;
94    def.nBufferAlignment = 1;
95
96    def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
97    def.format.audio.pNativeRender = NULL;
98    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
99    def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
100
101    addPort(def);
102
103    def.nPortIndex = 1;
104    def.eDir = OMX_DirOutput;
105    def.nBufferCountMin = kNumOutputBuffers;
106    def.nBufferCountActual = def.nBufferCountMin;
107    def.nBufferSize = 4096 * MAX_CHANNEL_COUNT;
108    def.bEnabled = OMX_TRUE;
109    def.bPopulated = OMX_FALSE;
110    def.eDomain = OMX_PortDomainAudio;
111    def.bBuffersContiguous = OMX_FALSE;
112    def.nBufferAlignment = 2;
113
114    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
115    def.format.audio.pNativeRender = NULL;
116    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
117    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
118
119    addPort(def);
120}
121
122status_t SoftAAC2::initDecoder() {
123    ALOGV("initDecoder()");
124    status_t status = UNKNOWN_ERROR;
125    mAACDecoder = aacDecoder_Open(TT_MP4_ADIF, /* num layers */ 1);
126    if (mAACDecoder != NULL) {
127        mStreamInfo = aacDecoder_GetStreamInfo(mAACDecoder);
128        if (mStreamInfo != NULL) {
129            status = OK;
130        }
131    }
132
133    mEndOfInput = false;
134    mEndOfOutput = false;
135    mOutputDelayCompensated = 0;
136    mOutputDelayRingBufferSize = 2048 * MAX_CHANNEL_COUNT * kNumDelayBlocksMax;
137    mOutputDelayRingBuffer = new short[mOutputDelayRingBufferSize];
138    mOutputDelayRingBufferWritePos = 0;
139    mOutputDelayRingBufferReadPos = 0;
140
141    if (mAACDecoder == NULL) {
142        ALOGE("AAC decoder is null. TODO: Can not call aacDecoder_SetParam in the following code");
143    }
144
145    //aacDecoder_SetParam(mAACDecoder, AAC_PCM_LIMITER_ENABLE, 0);
146
147    //init DRC wrapper
148    mDrcWrap.setDecoderHandle(mAACDecoder);
149    mDrcWrap.submitStreamData(mStreamInfo);
150
151    // for streams that contain metadata, use the mobile profile DRC settings unless overridden by platform properties
152    // TODO: change the DRC settings depending on audio output device type (HDMI, loadspeaker, headphone)
153    char value[PROPERTY_VALUE_MAX];
154    //  DRC_PRES_MODE_WRAP_DESIRED_TARGET
155    if (property_get(PROP_DRC_OVERRIDE_REF_LEVEL, value, NULL)) {
156        unsigned refLevel = atoi(value);
157        ALOGV("AAC decoder using desired DRC target reference level of %d instead of %d", refLevel,
158                DRC_DEFAULT_MOBILE_REF_LEVEL);
159        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, refLevel);
160    } else {
161        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, DRC_DEFAULT_MOBILE_REF_LEVEL);
162    }
163    //  DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR
164    if (property_get(PROP_DRC_OVERRIDE_CUT, value, NULL)) {
165        unsigned cut = atoi(value);
166        ALOGV("AAC decoder using desired DRC attenuation factor of %d instead of %d", cut,
167                DRC_DEFAULT_MOBILE_DRC_CUT);
168        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, cut);
169    } else {
170        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, DRC_DEFAULT_MOBILE_DRC_CUT);
171    }
172    //  DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR
173    if (property_get(PROP_DRC_OVERRIDE_BOOST, value, NULL)) {
174        unsigned boost = atoi(value);
175        ALOGV("AAC decoder using desired DRC boost factor of %d instead of %d", boost,
176                DRC_DEFAULT_MOBILE_DRC_BOOST);
177        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, boost);
178    } else {
179        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, DRC_DEFAULT_MOBILE_DRC_BOOST);
180    }
181    //  DRC_PRES_MODE_WRAP_DESIRED_HEAVY
182    if (property_get(PROP_DRC_OVERRIDE_HEAVY, value, NULL)) {
183        unsigned heavy = atoi(value);
184        ALOGV("AAC decoder using desried DRC heavy compression switch of %d instead of %d", heavy,
185                DRC_DEFAULT_MOBILE_DRC_HEAVY);
186        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, heavy);
187    } else {
188        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, DRC_DEFAULT_MOBILE_DRC_HEAVY);
189    }
190    // DRC_PRES_MODE_WRAP_ENCODER_TARGET
191    if (property_get(PROP_DRC_OVERRIDE_ENC_LEVEL, value, NULL)) {
192        unsigned encoderRefLevel = atoi(value);
193        ALOGV("AAC decoder using encoder-side DRC reference level of %d instead of %d",
194                encoderRefLevel, DRC_DEFAULT_MOBILE_ENC_LEVEL);
195        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, encoderRefLevel);
196    } else {
197        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, DRC_DEFAULT_MOBILE_ENC_LEVEL);
198    }
199
200    return status;
201}
202
203OMX_ERRORTYPE SoftAAC2::internalGetParameter(
204        OMX_INDEXTYPE index, OMX_PTR params) {
205    switch (index) {
206        case OMX_IndexParamAudioAac:
207        {
208            OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
209                (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
210
211            if (aacParams->nPortIndex != 0) {
212                return OMX_ErrorUndefined;
213            }
214
215            aacParams->nBitRate = 0;
216            aacParams->nAudioBandWidth = 0;
217            aacParams->nAACtools = 0;
218            aacParams->nAACERtools = 0;
219            aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
220
221            aacParams->eAACStreamFormat =
222                mIsADTS
223                    ? OMX_AUDIO_AACStreamFormatMP4ADTS
224                    : OMX_AUDIO_AACStreamFormatMP4FF;
225
226            aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
227
228            if (!isConfigured()) {
229                aacParams->nChannels = 1;
230                aacParams->nSampleRate = 44100;
231                aacParams->nFrameLength = 0;
232            } else {
233                aacParams->nChannels = mStreamInfo->numChannels;
234                aacParams->nSampleRate = mStreamInfo->sampleRate;
235                aacParams->nFrameLength = mStreamInfo->frameSize;
236            }
237
238            return OMX_ErrorNone;
239        }
240
241        case OMX_IndexParamAudioPcm:
242        {
243            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
244                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
245
246            if (pcmParams->nPortIndex != 1) {
247                return OMX_ErrorUndefined;
248            }
249
250            pcmParams->eNumData = OMX_NumericalDataSigned;
251            pcmParams->eEndian = OMX_EndianBig;
252            pcmParams->bInterleaved = OMX_TRUE;
253            pcmParams->nBitPerSample = 16;
254            pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
255            pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
256            pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
257            pcmParams->eChannelMapping[2] = OMX_AUDIO_ChannelCF;
258            pcmParams->eChannelMapping[3] = OMX_AUDIO_ChannelLFE;
259            pcmParams->eChannelMapping[4] = OMX_AUDIO_ChannelLS;
260            pcmParams->eChannelMapping[5] = OMX_AUDIO_ChannelRS;
261
262            if (!isConfigured()) {
263                pcmParams->nChannels = 1;
264                pcmParams->nSamplingRate = 44100;
265            } else {
266                pcmParams->nChannels = mStreamInfo->numChannels;
267                pcmParams->nSamplingRate = mStreamInfo->sampleRate;
268            }
269
270            return OMX_ErrorNone;
271        }
272
273        default:
274            return SimpleSoftOMXComponent::internalGetParameter(index, params);
275    }
276}
277
278OMX_ERRORTYPE SoftAAC2::internalSetParameter(
279        OMX_INDEXTYPE index, const OMX_PTR params) {
280    switch ((int)index) {
281        case OMX_IndexParamStandardComponentRole:
282        {
283            const OMX_PARAM_COMPONENTROLETYPE *roleParams =
284                (const OMX_PARAM_COMPONENTROLETYPE *)params;
285
286            if (strncmp((const char *)roleParams->cRole,
287                        "audio_decoder.aac",
288                        OMX_MAX_STRINGNAME_SIZE - 1)) {
289                return OMX_ErrorUndefined;
290            }
291
292            return OMX_ErrorNone;
293        }
294
295        case OMX_IndexParamAudioAac:
296        {
297            const OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
298                (const OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
299
300            if (aacParams->nPortIndex != 0) {
301                return OMX_ErrorUndefined;
302            }
303
304            if (aacParams->eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4FF) {
305                mIsADTS = false;
306            } else if (aacParams->eAACStreamFormat
307                        == OMX_AUDIO_AACStreamFormatMP4ADTS) {
308                mIsADTS = true;
309            } else {
310                return OMX_ErrorUndefined;
311            }
312
313            return OMX_ErrorNone;
314        }
315
316        case OMX_IndexParamAudioAndroidAacPresentation:
317        {
318            const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *aacPresParams =
319                    (const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *)params;
320            // for the following parameters of the OMX_AUDIO_PARAM_AACPROFILETYPE structure,
321            // a value of -1 implies the parameter is not set by the application:
322            //   nMaxOutputChannels     uses default platform properties, see configureDownmix()
323            //   nDrcCut                uses default platform properties, see initDecoder()
324            //   nDrcBoost                idem
325            //   nHeavyCompression        idem
326            //   nTargetReferenceLevel    idem
327            //   nEncodedTargetLevel      idem
328            if (aacPresParams->nMaxOutputChannels >= 0) {
329                int max;
330                if (aacPresParams->nMaxOutputChannels >= 8) { max = 8; }
331                else if (aacPresParams->nMaxOutputChannels >= 6) { max = 6; }
332                else if (aacPresParams->nMaxOutputChannels >= 2) { max = 2; }
333                else {
334                    // -1 or 0: disable downmix,  1: mono
335                    max = aacPresParams->nMaxOutputChannels;
336                }
337                ALOGV("set nMaxOutputChannels=%d", max);
338                aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, max);
339            }
340            bool updateDrcWrapper = false;
341            if (aacPresParams->nDrcBoost >= 0) {
342                ALOGV("set nDrcBoost=%d", aacPresParams->nDrcBoost);
343                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR,
344                        aacPresParams->nDrcBoost);
345                updateDrcWrapper = true;
346            }
347            if (aacPresParams->nDrcCut >= 0) {
348                ALOGV("set nDrcCut=%d", aacPresParams->nDrcCut);
349                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, aacPresParams->nDrcCut);
350                updateDrcWrapper = true;
351            }
352            if (aacPresParams->nHeavyCompression >= 0) {
353                ALOGV("set nHeavyCompression=%d", aacPresParams->nHeavyCompression);
354                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY,
355                        aacPresParams->nHeavyCompression);
356                updateDrcWrapper = true;
357            }
358            if (aacPresParams->nTargetReferenceLevel >= 0) {
359                ALOGV("set nTargetReferenceLevel=%d", aacPresParams->nTargetReferenceLevel);
360                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET,
361                        aacPresParams->nTargetReferenceLevel);
362                updateDrcWrapper = true;
363            }
364            if (aacPresParams->nEncodedTargetLevel >= 0) {
365                ALOGV("set nEncodedTargetLevel=%d", aacPresParams->nEncodedTargetLevel);
366                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET,
367                        aacPresParams->nEncodedTargetLevel);
368                updateDrcWrapper = true;
369            }
370            if (updateDrcWrapper) {
371                mDrcWrap.update();
372            }
373
374            return OMX_ErrorNone;
375        }
376
377        case OMX_IndexParamAudioPcm:
378        {
379            const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
380                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
381
382            if (pcmParams->nPortIndex != 1) {
383                return OMX_ErrorUndefined;
384            }
385
386            return OMX_ErrorNone;
387        }
388
389        default:
390            return SimpleSoftOMXComponent::internalSetParameter(index, params);
391    }
392}
393
394bool SoftAAC2::isConfigured() const {
395    return mInputBufferCount > 0;
396}
397
398void SoftAAC2::configureDownmix() const {
399    char value[PROPERTY_VALUE_MAX];
400    if (!(property_get("media.aac_51_output_enabled", value, NULL)
401            && (!strcmp(value, "1") || !strcasecmp(value, "true")))) {
402        ALOGI("limiting to stereo output");
403        aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
404        // By default, the decoder creates a 5.1 channel downmix signal
405        // for seven and eight channel input streams. To enable 6.1 and 7.1 channel output
406        // use aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1)
407    }
408}
409
410bool SoftAAC2::outputDelayRingBufferPutSamples(INT_PCM *samples, int32_t numSamples) {
411    if (mOutputDelayRingBufferWritePos + numSamples <= mOutputDelayRingBufferSize
412            && (mOutputDelayRingBufferReadPos <= mOutputDelayRingBufferWritePos
413                    || mOutputDelayRingBufferReadPos > mOutputDelayRingBufferWritePos + numSamples)) {
414        // faster memcopy loop without checks, if the preconditions allow this
415        for (int32_t i = 0; i < numSamples; i++) {
416            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos++] = samples[i];
417        }
418
419        if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
420            mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
421        }
422        if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
423            ALOGE("RING BUFFER OVERFLOW");
424            return false;
425        }
426    } else {
427        ALOGV("slow SoftAAC2::outputDelayRingBufferPutSamples()");
428
429        for (int32_t i = 0; i < numSamples; i++) {
430            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos] = samples[i];
431            mOutputDelayRingBufferWritePos++;
432            if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
433                mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
434            }
435            if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
436                ALOGE("RING BUFFER OVERFLOW");
437                return false;
438            }
439        }
440    }
441    return true;
442}
443
444int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numSamples) {
445    if (mOutputDelayRingBufferReadPos + numSamples <= mOutputDelayRingBufferSize
446            && (mOutputDelayRingBufferWritePos < mOutputDelayRingBufferReadPos
447                    || mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferReadPos + numSamples)) {
448        // faster memcopy loop without checks, if the preconditions allow this
449        if (samples != 0) {
450            for (int32_t i = 0; i < numSamples; i++) {
451                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos++];
452            }
453        } else {
454            mOutputDelayRingBufferReadPos += numSamples;
455        }
456        if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
457            mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
458        }
459    } else {
460        ALOGV("slow SoftAAC2::outputDelayRingBufferGetSamples()");
461
462        for (int32_t i = 0; i < numSamples; i++) {
463            if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
464                ALOGE("RING BUFFER UNDERRUN");
465                return -1;
466            }
467            if (samples != 0) {
468                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos];
469            }
470            mOutputDelayRingBufferReadPos++;
471            if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
472                mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
473            }
474        }
475    }
476    return numSamples;
477}
478
479int32_t SoftAAC2::outputDelayRingBufferSamplesAvailable() {
480    int32_t available = mOutputDelayRingBufferWritePos - mOutputDelayRingBufferReadPos;
481    if (available < 0) {
482        available += mOutputDelayRingBufferSize;
483    }
484    if (available < 0) {
485        ALOGE("FATAL RING BUFFER ERROR");
486        return 0;
487    }
488    return available;
489}
490
491int32_t SoftAAC2::outputDelayRingBufferSamplesLeft() {
492    return mOutputDelayRingBufferSize - outputDelayRingBufferSamplesAvailable();
493}
494
495
496void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
497    if (mSignalledError || mOutputPortSettingsChange != NONE) {
498        return;
499    }
500
501    UCHAR* inBuffer[FILEREAD_MAX_LAYERS];
502    UINT inBufferLength[FILEREAD_MAX_LAYERS] = {0};
503    UINT bytesValid[FILEREAD_MAX_LAYERS] = {0};
504
505    List<BufferInfo *> &inQueue = getPortQueue(0);
506    List<BufferInfo *> &outQueue = getPortQueue(1);
507
508    while ((!inQueue.empty() || mEndOfInput) && !outQueue.empty()) {
509        if (!inQueue.empty()) {
510            INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
511            BufferInfo *inInfo = *inQueue.begin();
512            OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
513
514            mEndOfInput = (inHeader->nFlags & OMX_BUFFERFLAG_EOS) != 0;
515            if ((inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG) != 0) {
516                BufferInfo *inInfo = *inQueue.begin();
517                OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
518
519                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
520                inBufferLength[0] = inHeader->nFilledLen;
521
522                AAC_DECODER_ERROR decoderErr =
523                    aacDecoder_ConfigRaw(mAACDecoder,
524                                         inBuffer,
525                                         inBufferLength);
526
527                if (decoderErr != AAC_DEC_OK) {
528                    ALOGW("aacDecoder_ConfigRaw decoderErr = 0x%4.4x", decoderErr);
529                    mSignalledError = true;
530                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
531                    return;
532                }
533
534                mInputBufferCount++;
535                mOutputBufferCount++; // fake increase of outputBufferCount to keep the counters aligned
536
537                inInfo->mOwnedByUs = false;
538                inQueue.erase(inQueue.begin());
539                mLastInHeader = NULL;
540                inInfo = NULL;
541                notifyEmptyBufferDone(inHeader);
542                inHeader = NULL;
543
544                configureDownmix();
545                // Only send out port settings changed event if both sample rate
546                // and numChannels are valid.
547                if (mStreamInfo->sampleRate && mStreamInfo->numChannels) {
548                    ALOGI("Initially configuring decoder: %d Hz, %d channels",
549                        mStreamInfo->sampleRate,
550                        mStreamInfo->numChannels);
551
552                    notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
553                    mOutputPortSettingsChange = AWAITING_DISABLED;
554                }
555                return;
556            }
557
558            if (inHeader->nFilledLen == 0) {
559                inInfo->mOwnedByUs = false;
560                inQueue.erase(inQueue.begin());
561                mLastInHeader = NULL;
562                inInfo = NULL;
563                notifyEmptyBufferDone(inHeader);
564                inHeader = NULL;
565                continue;
566            }
567
568            if (mIsADTS) {
569                size_t adtsHeaderSize = 0;
570                // skip 30 bits, aac_frame_length follows.
571                // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
572
573                const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
574
575                bool signalError = false;
576                if (inHeader->nFilledLen < 7) {
577                    ALOGE("Audio data too short to contain even the ADTS header. "
578                            "Got %d bytes.", inHeader->nFilledLen);
579                    hexdump(adtsHeader, inHeader->nFilledLen);
580                    signalError = true;
581                } else {
582                    bool protectionAbsent = (adtsHeader[1] & 1);
583
584                    unsigned aac_frame_length =
585                        ((adtsHeader[3] & 3) << 11)
586                        | (adtsHeader[4] << 3)
587                        | (adtsHeader[5] >> 5);
588
589                    if (inHeader->nFilledLen < aac_frame_length) {
590                        ALOGE("Not enough audio data for the complete frame. "
591                                "Got %d bytes, frame size according to the ADTS "
592                                "header is %u bytes.",
593                                inHeader->nFilledLen, aac_frame_length);
594                        hexdump(adtsHeader, inHeader->nFilledLen);
595                        signalError = true;
596                    } else {
597                        adtsHeaderSize = (protectionAbsent ? 7 : 9);
598
599                        inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
600                        inBufferLength[0] = aac_frame_length - adtsHeaderSize;
601
602                        inHeader->nOffset += adtsHeaderSize;
603                        inHeader->nFilledLen -= adtsHeaderSize;
604                    }
605                }
606
607                if (signalError) {
608                    mSignalledError = true;
609                    notify(OMX_EventError, OMX_ErrorStreamCorrupt, ERROR_MALFORMED, NULL);
610                    return;
611                }
612
613                // insert buffer size and time stamp
614                mBufferSizes.add(inBufferLength[0]);
615                if (mLastInHeader != inHeader) {
616                    mBufferTimestamps.add(inHeader->nTimeStamp);
617                    mLastInHeader = inHeader;
618                } else {
619                    int64_t currentTime = mBufferTimestamps.top();
620                    currentTime += mStreamInfo->aacSamplesPerFrame *
621                            1000000ll / mStreamInfo->sampleRate;
622                    mBufferTimestamps.add(currentTime);
623                }
624            } else {
625                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
626                inBufferLength[0] = inHeader->nFilledLen;
627                mLastInHeader = inHeader;
628                mBufferTimestamps.add(inHeader->nTimeStamp);
629                mBufferSizes.add(inHeader->nFilledLen);
630            }
631
632            // Fill and decode
633            bytesValid[0] = inBufferLength[0];
634
635            INT prevSampleRate = mStreamInfo->sampleRate;
636            INT prevNumChannels = mStreamInfo->numChannels;
637
638            aacDecoder_Fill(mAACDecoder,
639                            inBuffer,
640                            inBufferLength,
641                            bytesValid);
642
643            // run DRC check
644            mDrcWrap.submitStreamData(mStreamInfo);
645            mDrcWrap.update();
646
647            UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
648            inHeader->nFilledLen -= inBufferUsedLength;
649            inHeader->nOffset += inBufferUsedLength;
650
651            AAC_DECODER_ERROR decoderErr;
652            do {
653                int numconsumed = mStreamInfo->numTotalBytes + mStreamInfo->numBadBytes;
654                decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
655                                           tmpOutBuffer,
656                                           2048 * MAX_CHANNEL_COUNT,
657                                           0 /* flags */);
658
659                numconsumed = (mStreamInfo->numTotalBytes + mStreamInfo->numBadBytes) - numconsumed;
660                if (numconsumed != 0) {
661                    mDecodedSizes.add(numconsumed);
662                }
663
664                if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
665                    break;
666                }
667
668                if (decoderErr != AAC_DEC_OK) {
669                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
670                }
671
672                if (bytesValid[0] != 0) {
673                    ALOGE("bytesValid[0] != 0 should never happen");
674                    mSignalledError = true;
675                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
676                    return;
677                }
678
679                size_t numOutBytes =
680                    mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
681
682                if (decoderErr == AAC_DEC_OK) {
683                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
684                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
685                        mSignalledError = true;
686                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
687                        return;
688                    }
689                } else {
690                    ALOGW("AAC decoder returned error 0x%4.4x, substituting silence", decoderErr);
691
692                    memset(tmpOutBuffer, 0, numOutBytes); // TODO: check for overflow
693
694                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
695                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
696                        mSignalledError = true;
697                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
698                        return;
699                    }
700
701                    // Discard input buffer.
702                    inHeader->nFilledLen = 0;
703
704                    aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
705
706                    // fall through
707                }
708
709                /*
710                 * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
711                 * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
712                 * rate system and the sampling rate in the final output is actually
713                 * doubled compared with the core AAC decoder sampling rate.
714                 *
715                 * Explicit signalling is done by explicitly defining SBR audio object
716                 * type in the bitstream. Implicit signalling is done by embedding
717                 * SBR content in AAC extension payload specific to SBR, and hence
718                 * requires an AAC decoder to perform pre-checks on actual audio frames.
719                 *
720                 * Thus, we could not say for sure whether a stream is
721                 * AAC+/eAAC+ until the first data frame is decoded.
722                 */
723                if (mInputBufferCount <= 2 || mOutputBufferCount > 1) { // TODO: <= 1
724                    if (mStreamInfo->sampleRate != prevSampleRate ||
725                        mStreamInfo->numChannels != prevNumChannels) {
726                        ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
727                              prevSampleRate, mStreamInfo->sampleRate,
728                              prevNumChannels, mStreamInfo->numChannels);
729
730                        notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
731                        mOutputPortSettingsChange = AWAITING_DISABLED;
732
733                        if (inHeader->nFilledLen == 0) {
734                            inInfo->mOwnedByUs = false;
735                            mInputBufferCount++;
736                            inQueue.erase(inQueue.begin());
737                            mLastInHeader = NULL;
738                            inInfo = NULL;
739                            notifyEmptyBufferDone(inHeader);
740                            inHeader = NULL;
741                        }
742                        return;
743                    }
744                } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
745                    ALOGW("Invalid AAC stream");
746                    mSignalledError = true;
747                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
748                    return;
749                }
750                if (inHeader && inHeader->nFilledLen == 0) {
751                    inInfo->mOwnedByUs = false;
752                    mInputBufferCount++;
753                    inQueue.erase(inQueue.begin());
754                    mLastInHeader = NULL;
755                    inInfo = NULL;
756                    notifyEmptyBufferDone(inHeader);
757                    inHeader = NULL;
758                } else {
759                    ALOGV("inHeader->nFilledLen = %d", inHeader ? inHeader->nFilledLen : 0);
760                }
761            } while (decoderErr == AAC_DEC_OK);
762        }
763
764        int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
765
766        if (!mEndOfInput && mOutputDelayCompensated < outputDelay) {
767            // discard outputDelay at the beginning
768            int32_t toCompensate = outputDelay - mOutputDelayCompensated;
769            int32_t discard = outputDelayRingBufferSamplesAvailable();
770            if (discard > toCompensate) {
771                discard = toCompensate;
772            }
773            int32_t discarded = outputDelayRingBufferGetSamples(0, discard);
774            mOutputDelayCompensated += discarded;
775            continue;
776        }
777
778        if (mEndOfInput) {
779            while (mOutputDelayCompensated > 0) {
780                // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
781                INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
782
783                 // run DRC check
784                 mDrcWrap.submitStreamData(mStreamInfo);
785                 mDrcWrap.update();
786
787                AAC_DECODER_ERROR decoderErr =
788                    aacDecoder_DecodeFrame(mAACDecoder,
789                                           tmpOutBuffer,
790                                           2048 * MAX_CHANNEL_COUNT,
791                                           AACDEC_FLUSH);
792                if (decoderErr != AAC_DEC_OK) {
793                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
794                }
795
796                int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
797                if (tmpOutBufferSamples > mOutputDelayCompensated) {
798                    tmpOutBufferSamples = mOutputDelayCompensated;
799                }
800                outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
801                mOutputDelayCompensated -= tmpOutBufferSamples;
802            }
803        }
804
805        while (!outQueue.empty()
806                && outputDelayRingBufferSamplesAvailable()
807                        >= mStreamInfo->frameSize * mStreamInfo->numChannels) {
808            BufferInfo *outInfo = *outQueue.begin();
809            OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
810
811            if (outHeader->nOffset != 0) {
812                ALOGE("outHeader->nOffset != 0 is not handled");
813                mSignalledError = true;
814                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
815                return;
816            }
817
818            INT_PCM *outBuffer =
819                    reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);
820            int samplesize = mStreamInfo->numChannels * sizeof(int16_t);
821            if (outHeader->nOffset
822                    + mStreamInfo->frameSize * samplesize
823                    > outHeader->nAllocLen) {
824                ALOGE("buffer overflow");
825                mSignalledError = true;
826                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
827                return;
828
829            }
830
831            int available = outputDelayRingBufferSamplesAvailable();
832            int numSamples = outHeader->nAllocLen / samplesize;
833            if (numSamples > available) {
834                numSamples = available;
835            }
836            int64_t currentTime = 0;
837            if (available) {
838
839                int numFrames = numSamples / (mStreamInfo->frameSize * mStreamInfo->numChannels);
840                numSamples = numFrames * (mStreamInfo->frameSize * mStreamInfo->numChannels);
841
842                ALOGV("%d samples available (%d), or %d frames",
843                        numSamples, available, numFrames);
844                int64_t *nextTimeStamp = &mBufferTimestamps.editItemAt(0);
845                currentTime = *nextTimeStamp;
846                int32_t *currentBufLeft = &mBufferSizes.editItemAt(0);
847                for (int i = 0; i < numFrames; i++) {
848                    int32_t decodedSize = mDecodedSizes.itemAt(0);
849                    mDecodedSizes.removeAt(0);
850                    ALOGV("decoded %d of %d", decodedSize, *currentBufLeft);
851                    if (*currentBufLeft > decodedSize) {
852                        // adjust/interpolate next time stamp
853                        *currentBufLeft -= decodedSize;
854                        *nextTimeStamp += mStreamInfo->aacSamplesPerFrame *
855                                1000000ll / mStreamInfo->sampleRate;
856                        ALOGV("adjusted nextTimeStamp/size to %lld/%d",
857                                *nextTimeStamp, *currentBufLeft);
858                    } else {
859                        // move to next timestamp in list
860                        if (mBufferTimestamps.size() > 0) {
861                            mBufferTimestamps.removeAt(0);
862                            nextTimeStamp = &mBufferTimestamps.editItemAt(0);
863                            mBufferSizes.removeAt(0);
864                            currentBufLeft = &mBufferSizes.editItemAt(0);
865                            ALOGV("moved to next time/size: %lld/%d",
866                                    *nextTimeStamp, *currentBufLeft);
867                        }
868                        // try to limit output buffer size to match input buffers
869                        // (e.g when an input buffer contained 4 "sub" frames, output
870                        // at most 4 decoded units in the corresponding output buffer)
871                        // This is optional. Remove the next three lines to fill the output
872                        // buffer with as many units as available.
873                        numFrames = i + 1;
874                        numSamples = numFrames * mStreamInfo->frameSize * mStreamInfo->numChannels;
875                        break;
876                    }
877                }
878
879                ALOGV("getting %d from ringbuffer", numSamples);
880                int32_t ns = outputDelayRingBufferGetSamples(outBuffer, numSamples);
881                if (ns != numSamples) {
882                    ALOGE("not a complete frame of samples available");
883                    mSignalledError = true;
884                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
885                    return;
886                }
887            }
888
889            outHeader->nFilledLen = numSamples * sizeof(int16_t);
890
891            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
892                outHeader->nFlags = OMX_BUFFERFLAG_EOS;
893                mEndOfOutput = true;
894            } else {
895                outHeader->nFlags = 0;
896            }
897
898            outHeader->nTimeStamp = currentTime;
899
900            mOutputBufferCount++;
901            outInfo->mOwnedByUs = false;
902            outQueue.erase(outQueue.begin());
903            outInfo = NULL;
904            ALOGV("out timestamp %lld / %d", outHeader->nTimeStamp, outHeader->nFilledLen);
905            notifyFillBufferDone(outHeader);
906            outHeader = NULL;
907        }
908
909        if (mEndOfInput) {
910            if (outputDelayRingBufferSamplesAvailable() > 0
911                    && outputDelayRingBufferSamplesAvailable()
912                            < mStreamInfo->frameSize * mStreamInfo->numChannels) {
913                ALOGE("not a complete frame of samples available");
914                mSignalledError = true;
915                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
916                return;
917            }
918
919            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
920                if (!mEndOfOutput) {
921                    // send empty block signaling EOS
922                    mEndOfOutput = true;
923                    BufferInfo *outInfo = *outQueue.begin();
924                    OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
925
926                    if (outHeader->nOffset != 0) {
927                        ALOGE("outHeader->nOffset != 0 is not handled");
928                        mSignalledError = true;
929                        notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
930                        return;
931                    }
932
933                    INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer
934                            + outHeader->nOffset);
935                    int32_t ns = 0;
936                    outHeader->nFilledLen = 0;
937                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
938
939                    outHeader->nTimeStamp = mBufferTimestamps.itemAt(0);
940                    mBufferTimestamps.clear();
941                    mBufferSizes.clear();
942                    mDecodedSizes.clear();
943
944                    mOutputBufferCount++;
945                    outInfo->mOwnedByUs = false;
946                    outQueue.erase(outQueue.begin());
947                    outInfo = NULL;
948                    notifyFillBufferDone(outHeader);
949                    outHeader = NULL;
950                }
951                break; // if outQueue not empty but no more output
952            }
953        }
954    }
955}
956
957void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
958    if (portIndex == 0) {
959        // Make sure that the next buffer output does not still
960        // depend on fragments from the last one decoded.
961        // drain all existing data
962        drainDecoder();
963        mBufferTimestamps.clear();
964        mBufferSizes.clear();
965        mDecodedSizes.clear();
966        mLastInHeader = NULL;
967    } else {
968        while (outputDelayRingBufferSamplesAvailable() > 0) {
969            int32_t ns = outputDelayRingBufferGetSamples(0,
970                    mStreamInfo->frameSize * mStreamInfo->numChannels);
971            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
972                ALOGE("not a complete frame of samples available");
973            }
974            mOutputBufferCount++;
975        }
976        mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
977    }
978}
979
980void SoftAAC2::drainDecoder() {
981    int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
982
983    // flush decoder until outputDelay is compensated
984    while (mOutputDelayCompensated > 0) {
985        // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
986        INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
987
988        // run DRC check
989        mDrcWrap.submitStreamData(mStreamInfo);
990        mDrcWrap.update();
991
992        AAC_DECODER_ERROR decoderErr =
993            aacDecoder_DecodeFrame(mAACDecoder,
994                                   tmpOutBuffer,
995                                   2048 * MAX_CHANNEL_COUNT,
996                                   AACDEC_FLUSH);
997        if (decoderErr != AAC_DEC_OK) {
998            ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
999        }
1000
1001        int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
1002        if (tmpOutBufferSamples > mOutputDelayCompensated) {
1003            tmpOutBufferSamples = mOutputDelayCompensated;
1004        }
1005        outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
1006
1007        mOutputDelayCompensated -= tmpOutBufferSamples;
1008    }
1009}
1010
1011void SoftAAC2::onReset() {
1012    drainDecoder();
1013    // reset the "configured" state
1014    mInputBufferCount = 0;
1015    mOutputBufferCount = 0;
1016    mOutputDelayCompensated = 0;
1017    mOutputDelayRingBufferWritePos = 0;
1018    mOutputDelayRingBufferReadPos = 0;
1019    mEndOfInput = false;
1020    mEndOfOutput = false;
1021    mBufferTimestamps.clear();
1022    mBufferSizes.clear();
1023    mDecodedSizes.clear();
1024    mLastInHeader = NULL;
1025
1026    // To make the codec behave the same before and after a reset, we need to invalidate the
1027    // streaminfo struct. This does that:
1028    mStreamInfo->sampleRate = 0; // TODO: mStreamInfo is read only
1029
1030    mSignalledError = false;
1031    mOutputPortSettingsChange = NONE;
1032}
1033
1034void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
1035    if (portIndex != 1) {
1036        return;
1037    }
1038
1039    switch (mOutputPortSettingsChange) {
1040        case NONE:
1041            break;
1042
1043        case AWAITING_DISABLED:
1044        {
1045            CHECK(!enabled);
1046            mOutputPortSettingsChange = AWAITING_ENABLED;
1047            break;
1048        }
1049
1050        default:
1051        {
1052            CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
1053            CHECK(enabled);
1054            mOutputPortSettingsChange = NONE;
1055            break;
1056        }
1057    }
1058}
1059
1060}  // namespace android
1061
1062android::SoftOMXComponent *createSoftOMXComponent(
1063        const char *name, const OMX_CALLBACKTYPE *callbacks,
1064        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
1065    return new android::SoftAAC2(name, callbacks, appData, component);
1066}
1067