SoftAAC2.cpp revision 8484830a6b488b41da0e32acacf2e6b68060d9d0 
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_TAG "SoftAAC2"
18//#define LOG_NDEBUG 0
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      mCurrentInputTime(0),
72      mOutputPortSettingsChange(NONE) {
73    initPorts();
74    CHECK_EQ(initDecoder(), (status_t)OK);
75}
76
77SoftAAC2::~SoftAAC2() {
78    aacDecoder_Close(mAACDecoder);
79    delete mOutputDelayRingBuffer;
80}
81
82void SoftAAC2::initPorts() {
83    OMX_PARAM_PORTDEFINITIONTYPE def;
84    InitOMXParams(&def);
85
86    def.nPortIndex = 0;
87    def.eDir = OMX_DirInput;
88    def.nBufferCountMin = kNumInputBuffers;
89    def.nBufferCountActual = def.nBufferCountMin;
90    def.nBufferSize = 8192;
91    def.bEnabled = OMX_TRUE;
92    def.bPopulated = OMX_FALSE;
93    def.eDomain = OMX_PortDomainAudio;
94    def.bBuffersContiguous = OMX_FALSE;
95    def.nBufferAlignment = 1;
96
97    def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
98    def.format.audio.pNativeRender = NULL;
99    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
100    def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
101
102    addPort(def);
103
104    def.nPortIndex = 1;
105    def.eDir = OMX_DirOutput;
106    def.nBufferCountMin = kNumOutputBuffers;
107    def.nBufferCountActual = def.nBufferCountMin;
108    def.nBufferSize = 4096 * MAX_CHANNEL_COUNT;
109    def.bEnabled = OMX_TRUE;
110    def.bPopulated = OMX_FALSE;
111    def.eDomain = OMX_PortDomainAudio;
112    def.bBuffersContiguous = OMX_FALSE;
113    def.nBufferAlignment = 2;
114
115    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
116    def.format.audio.pNativeRender = NULL;
117    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
118    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
119
120    addPort(def);
121}
122
123status_t SoftAAC2::initDecoder() {
124    ALOGV("initDecoder()");
125    status_t status = UNKNOWN_ERROR;
126    mAACDecoder = aacDecoder_Open(TT_MP4_ADIF, /* num layers */ 1);
127    if (mAACDecoder != NULL) {
128        mStreamInfo = aacDecoder_GetStreamInfo(mAACDecoder);
129        if (mStreamInfo != NULL) {
130            status = OK;
131        }
132    }
133
134    mEndOfInput = false;
135    mEndOfOutput = false;
136    mOutputDelayCompensated = 0;
137    mOutputDelayRingBufferSize = 2048 * MAX_CHANNEL_COUNT * kNumDelayBlocksMax;
138    mOutputDelayRingBuffer = new short[mOutputDelayRingBufferSize];
139    mOutputDelayRingBufferWritePos = 0;
140    mOutputDelayRingBufferReadPos = 0;
141
142    if (mAACDecoder == NULL) {
143        ALOGE("AAC decoder is null. TODO: Can not call aacDecoder_SetParam in the following code");
144    }
145
146    //aacDecoder_SetParam(mAACDecoder, AAC_PCM_LIMITER_ENABLE, 0);
147
148    //init DRC wrapper
149    mDrcWrap.setDecoderHandle(mAACDecoder);
150    mDrcWrap.submitStreamData(mStreamInfo);
151
152    // for streams that contain metadata, use the mobile profile DRC settings unless overridden by platform properties
153    // TODO: change the DRC settings depending on audio output device type (HDMI, loadspeaker, headphone)
154    char value[PROPERTY_VALUE_MAX];
155    //  DRC_PRES_MODE_WRAP_DESIRED_TARGET
156    if (property_get(PROP_DRC_OVERRIDE_REF_LEVEL, value, NULL)) {
157        unsigned refLevel = atoi(value);
158        ALOGV("AAC decoder using desired DRC target reference level of %d instead of %d", refLevel,
159                DRC_DEFAULT_MOBILE_REF_LEVEL);
160        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, refLevel);
161    } else {
162        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, DRC_DEFAULT_MOBILE_REF_LEVEL);
163    }
164    //  DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR
165    if (property_get(PROP_DRC_OVERRIDE_CUT, value, NULL)) {
166        unsigned cut = atoi(value);
167        ALOGV("AAC decoder using desired DRC attenuation factor of %d instead of %d", cut,
168                DRC_DEFAULT_MOBILE_DRC_CUT);
169        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, cut);
170    } else {
171        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, DRC_DEFAULT_MOBILE_DRC_CUT);
172    }
173    //  DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR
174    if (property_get(PROP_DRC_OVERRIDE_BOOST, value, NULL)) {
175        unsigned boost = atoi(value);
176        ALOGV("AAC decoder using desired DRC boost factor of %d instead of %d", boost,
177                DRC_DEFAULT_MOBILE_DRC_BOOST);
178        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, boost);
179    } else {
180        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, DRC_DEFAULT_MOBILE_DRC_BOOST);
181    }
182    //  DRC_PRES_MODE_WRAP_DESIRED_HEAVY
183    if (property_get(PROP_DRC_OVERRIDE_HEAVY, value, NULL)) {
184        unsigned heavy = atoi(value);
185        ALOGV("AAC decoder using desried DRC heavy compression switch of %d instead of %d", heavy,
186                DRC_DEFAULT_MOBILE_DRC_HEAVY);
187        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, heavy);
188    } else {
189        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, DRC_DEFAULT_MOBILE_DRC_HEAVY);
190    }
191    // DRC_PRES_MODE_WRAP_ENCODER_TARGET
192    if (property_get(PROP_DRC_OVERRIDE_ENC_LEVEL, value, NULL)) {
193        unsigned encoderRefLevel = atoi(value);
194        ALOGV("AAC decoder using encoder-side DRC reference level of %d instead of %d",
195                encoderRefLevel, DRC_DEFAULT_MOBILE_ENC_LEVEL);
196        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, encoderRefLevel);
197    } else {
198        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, DRC_DEFAULT_MOBILE_ENC_LEVEL);
199    }
200
201    return status;
202}
203
204OMX_ERRORTYPE SoftAAC2::internalGetParameter(
205        OMX_INDEXTYPE index, OMX_PTR params) {
206    switch (index) {
207        case OMX_IndexParamAudioAac:
208        {
209            OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
210                (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
211
212            if (aacParams->nPortIndex != 0) {
213                return OMX_ErrorUndefined;
214            }
215
216            aacParams->nBitRate = 0;
217            aacParams->nAudioBandWidth = 0;
218            aacParams->nAACtools = 0;
219            aacParams->nAACERtools = 0;
220            aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
221
222            aacParams->eAACStreamFormat =
223                mIsADTS
224                    ? OMX_AUDIO_AACStreamFormatMP4ADTS
225                    : OMX_AUDIO_AACStreamFormatMP4FF;
226
227            aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
228
229            if (!isConfigured()) {
230                aacParams->nChannels = 1;
231                aacParams->nSampleRate = 44100;
232                aacParams->nFrameLength = 0;
233            } else {
234                aacParams->nChannels = mStreamInfo->numChannels;
235                aacParams->nSampleRate = mStreamInfo->sampleRate;
236                aacParams->nFrameLength = mStreamInfo->frameSize;
237            }
238
239            return OMX_ErrorNone;
240        }
241
242        case OMX_IndexParamAudioPcm:
243        {
244            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
245                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
246
247            if (pcmParams->nPortIndex != 1) {
248                return OMX_ErrorUndefined;
249            }
250
251            pcmParams->eNumData = OMX_NumericalDataSigned;
252            pcmParams->eEndian = OMX_EndianBig;
253            pcmParams->bInterleaved = OMX_TRUE;
254            pcmParams->nBitPerSample = 16;
255            pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
256            pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
257            pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
258            pcmParams->eChannelMapping[2] = OMX_AUDIO_ChannelCF;
259            pcmParams->eChannelMapping[3] = OMX_AUDIO_ChannelLFE;
260            pcmParams->eChannelMapping[4] = OMX_AUDIO_ChannelLS;
261            pcmParams->eChannelMapping[5] = OMX_AUDIO_ChannelRS;
262
263            if (!isConfigured()) {
264                pcmParams->nChannels = 1;
265                pcmParams->nSamplingRate = 44100;
266            } else {
267                pcmParams->nChannels = mStreamInfo->numChannels;
268                pcmParams->nSamplingRate = mStreamInfo->sampleRate;
269            }
270
271            return OMX_ErrorNone;
272        }
273
274        default:
275            return SimpleSoftOMXComponent::internalGetParameter(index, params);
276    }
277}
278
279OMX_ERRORTYPE SoftAAC2::internalSetParameter(
280        OMX_INDEXTYPE index, const OMX_PTR params) {
281    switch ((int)index) {
282        case OMX_IndexParamStandardComponentRole:
283        {
284            const OMX_PARAM_COMPONENTROLETYPE *roleParams =
285                (const OMX_PARAM_COMPONENTROLETYPE *)params;
286
287            if (strncmp((const char *)roleParams->cRole,
288                        "audio_decoder.aac",
289                        OMX_MAX_STRINGNAME_SIZE - 1)) {
290                return OMX_ErrorUndefined;
291            }
292
293            return OMX_ErrorNone;
294        }
295
296        case OMX_IndexParamAudioAac:
297        {
298            const OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
299                (const OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
300
301            if (aacParams->nPortIndex != 0) {
302                return OMX_ErrorUndefined;
303            }
304
305            if (aacParams->eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4FF) {
306                mIsADTS = false;
307            } else if (aacParams->eAACStreamFormat
308                        == OMX_AUDIO_AACStreamFormatMP4ADTS) {
309                mIsADTS = true;
310            } else {
311                return OMX_ErrorUndefined;
312            }
313
314            return OMX_ErrorNone;
315        }
316
317        case OMX_IndexParamAudioAndroidAacPresentation:
318        {
319            const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *aacPresParams =
320                    (const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *)params;
321            // for the following parameters of the OMX_AUDIO_PARAM_AACPROFILETYPE structure,
322            // a value of -1 implies the parameter is not set by the application:
323            //   nMaxOutputChannels     uses default platform properties, see configureDownmix()
324            //   nDrcCut                uses default platform properties, see initDecoder()
325            //   nDrcBoost                idem
326            //   nHeavyCompression        idem
327            //   nTargetReferenceLevel    idem
328            //   nEncodedTargetLevel      idem
329            if (aacPresParams->nMaxOutputChannels >= 0) {
330                int max;
331                if (aacPresParams->nMaxOutputChannels >= 8) { max = 8; }
332                else if (aacPresParams->nMaxOutputChannels >= 6) { max = 6; }
333                else if (aacPresParams->nMaxOutputChannels >= 2) { max = 2; }
334                else {
335                    // -1 or 0: disable downmix,  1: mono
336                    max = aacPresParams->nMaxOutputChannels;
337                }
338                ALOGV("set nMaxOutputChannels=%d", max);
339                aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, max);
340            }
341            bool updateDrcWrapper = false;
342            if (aacPresParams->nDrcBoost >= 0) {
343                ALOGV("set nDrcBoost=%d", aacPresParams->nDrcBoost);
344                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR,
345                        aacPresParams->nDrcBoost);
346                updateDrcWrapper = true;
347            }
348            if (aacPresParams->nDrcCut >= 0) {
349                ALOGV("set nDrcCut=%d", aacPresParams->nDrcCut);
350                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, aacPresParams->nDrcCut);
351                updateDrcWrapper = true;
352            }
353            if (aacPresParams->nHeavyCompression >= 0) {
354                ALOGV("set nHeavyCompression=%d", aacPresParams->nHeavyCompression);
355                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY,
356                        aacPresParams->nHeavyCompression);
357                updateDrcWrapper = true;
358            }
359            if (aacPresParams->nTargetReferenceLevel >= 0) {
360                ALOGV("set nTargetReferenceLevel=%d", aacPresParams->nTargetReferenceLevel);
361                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET,
362                        aacPresParams->nTargetReferenceLevel);
363                updateDrcWrapper = true;
364            }
365            if (aacPresParams->nEncodedTargetLevel >= 0) {
366                ALOGV("set nEncodedTargetLevel=%d", aacPresParams->nEncodedTargetLevel);
367                mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET,
368                        aacPresParams->nEncodedTargetLevel);
369                updateDrcWrapper = true;
370            }
371            if (updateDrcWrapper) {
372                mDrcWrap.update();
373            }
374
375            return OMX_ErrorNone;
376        }
377
378        case OMX_IndexParamAudioPcm:
379        {
380            const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
381                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
382
383            if (pcmParams->nPortIndex != 1) {
384                return OMX_ErrorUndefined;
385            }
386
387            return OMX_ErrorNone;
388        }
389
390        default:
391            return SimpleSoftOMXComponent::internalSetParameter(index, params);
392    }
393}
394
395bool SoftAAC2::isConfigured() const {
396    return mInputBufferCount > 0;
397}
398
399void SoftAAC2::configureDownmix() const {
400    char value[PROPERTY_VALUE_MAX];
401    if (!(property_get("media.aac_51_output_enabled", value, NULL)
402            && (!strcmp(value, "1") || !strcasecmp(value, "true")))) {
403        ALOGI("limiting to stereo output");
404        aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
405        // By default, the decoder creates a 5.1 channel downmix signal
406        // for seven and eight channel input streams. To enable 6.1 and 7.1 channel output
407        // use aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1)
408    }
409}
410
411bool SoftAAC2::outputDelayRingBufferPutSamples(INT_PCM *samples, int32_t numSamples) {
412    if (mOutputDelayRingBufferWritePos + numSamples <= mOutputDelayRingBufferSize
413            && (mOutputDelayRingBufferReadPos <= mOutputDelayRingBufferWritePos
414                    || mOutputDelayRingBufferReadPos > mOutputDelayRingBufferWritePos + numSamples)) {
415        // faster memcopy loop without checks, if the preconditions allow this
416        for (int32_t i = 0; i < numSamples; i++) {
417            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos++] = samples[i];
418        }
419
420        if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
421            mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
422        }
423        if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
424            ALOGE("RING BUFFER OVERFLOW");
425            return false;
426        }
427    } else {
428        ALOGV("slow SoftAAC2::outputDelayRingBufferPutSamples()");
429
430        for (int32_t i = 0; i < numSamples; i++) {
431            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos] = samples[i];
432            mOutputDelayRingBufferWritePos++;
433            if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
434                mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
435            }
436            if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
437                ALOGE("RING BUFFER OVERFLOW");
438                return false;
439            }
440        }
441    }
442    return true;
443}
444
445int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numSamples) {
446    if (mOutputDelayRingBufferReadPos + numSamples <= mOutputDelayRingBufferSize
447            && (mOutputDelayRingBufferWritePos < mOutputDelayRingBufferReadPos
448                    || mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferReadPos + numSamples)) {
449        // faster memcopy loop without checks, if the preconditions allow this
450        if (samples != 0) {
451            for (int32_t i = 0; i < numSamples; i++) {
452                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos++];
453            }
454        } else {
455            mOutputDelayRingBufferReadPos += numSamples;
456        }
457        if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
458            mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
459        }
460    } else {
461        ALOGV("slow SoftAAC2::outputDelayRingBufferGetSamples()");
462
463        for (int32_t i = 0; i < numSamples; i++) {
464            if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
465                ALOGE("RING BUFFER UNDERRUN");
466                return -1;
467            }
468            if (samples != 0) {
469                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos];
470            }
471            mOutputDelayRingBufferReadPos++;
472            if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
473                mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
474            }
475        }
476    }
477    return numSamples;
478}
479
480int32_t SoftAAC2::outputDelayRingBufferSamplesAvailable() {
481    int32_t available = mOutputDelayRingBufferWritePos - mOutputDelayRingBufferReadPos;
482    if (available < 0) {
483        available += mOutputDelayRingBufferSize;
484    }
485    if (available < 0) {
486        ALOGE("FATAL RING BUFFER ERROR");
487        return 0;
488    }
489    return available;
490}
491
492int32_t SoftAAC2::outputDelayRingBufferSamplesLeft() {
493    return mOutputDelayRingBufferSize - outputDelayRingBufferSamplesAvailable();
494}
495
496
497void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
498    if (mSignalledError || mOutputPortSettingsChange != NONE) {
499        return;
500    }
501
502    UCHAR* inBuffer[FILEREAD_MAX_LAYERS];
503    UINT inBufferLength[FILEREAD_MAX_LAYERS] = {0};
504    UINT bytesValid[FILEREAD_MAX_LAYERS] = {0};
505
506    List<BufferInfo *> &inQueue = getPortQueue(0);
507    List<BufferInfo *> &outQueue = getPortQueue(1);
508
509    while ((!inQueue.empty() || mEndOfInput) && !outQueue.empty()) {
510        if (!inQueue.empty()) {
511            INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
512            BufferInfo *inInfo = *inQueue.begin();
513            OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
514
515            mEndOfInput = (inHeader->nFlags & OMX_BUFFERFLAG_EOS) != 0;
516            if ((inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG) != 0) {
517                BufferInfo *inInfo = *inQueue.begin();
518                OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
519
520                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
521                inBufferLength[0] = inHeader->nFilledLen;
522
523                AAC_DECODER_ERROR decoderErr =
524                    aacDecoder_ConfigRaw(mAACDecoder,
525                                         inBuffer,
526                                         inBufferLength);
527
528                if (decoderErr != AAC_DEC_OK) {
529                    ALOGW("aacDecoder_ConfigRaw decoderErr = 0x%4.4x", decoderErr);
530                    mSignalledError = true;
531                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
532                    return;
533                }
534
535                mInputBufferCount++;
536                mOutputBufferCount++; // fake increase of outputBufferCount to keep the counters aligned
537
538                inInfo->mOwnedByUs = false;
539                inQueue.erase(inQueue.begin());
540                mLastInHeader = NULL;
541                inInfo = NULL;
542                notifyEmptyBufferDone(inHeader);
543                inHeader = NULL;
544
545                configureDownmix();
546                // Only send out port settings changed event if both sample rate
547                // and numChannels are valid.
548                if (mStreamInfo->sampleRate && mStreamInfo->numChannels) {
549                    ALOGI("Initially configuring decoder: %d Hz, %d channels",
550                        mStreamInfo->sampleRate,
551                        mStreamInfo->numChannels);
552
553                    notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
554                    mOutputPortSettingsChange = AWAITING_DISABLED;
555                }
556                return;
557            }
558
559            if (inHeader->nFilledLen == 0) {
560                inInfo->mOwnedByUs = false;
561                inQueue.erase(inQueue.begin());
562                mLastInHeader = NULL;
563                inInfo = NULL;
564                notifyEmptyBufferDone(inHeader);
565                inHeader = NULL;
566                continue;
567            }
568
569            if (mIsADTS) {
570                size_t adtsHeaderSize = 0;
571                // skip 30 bits, aac_frame_length follows.
572                // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
573
574                const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
575
576                bool signalError = false;
577                if (inHeader->nFilledLen < 7) {
578                    ALOGE("Audio data too short to contain even the ADTS header. "
579                            "Got %d bytes.", inHeader->nFilledLen);
580                    hexdump(adtsHeader, inHeader->nFilledLen);
581                    signalError = true;
582                } else {
583                    bool protectionAbsent = (adtsHeader[1] & 1);
584
585                    unsigned aac_frame_length =
586                        ((adtsHeader[3] & 3) << 11)
587                        | (adtsHeader[4] << 3)
588                        | (adtsHeader[5] >> 5);
589
590                    if (inHeader->nFilledLen < aac_frame_length) {
591                        ALOGE("Not enough audio data for the complete frame. "
592                                "Got %d bytes, frame size according to the ADTS "
593                                "header is %u bytes.",
594                                inHeader->nFilledLen, aac_frame_length);
595                        hexdump(adtsHeader, inHeader->nFilledLen);
596                        signalError = true;
597                    } else {
598                        adtsHeaderSize = (protectionAbsent ? 7 : 9);
599
600                        inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
601                        inBufferLength[0] = aac_frame_length - adtsHeaderSize;
602
603                        inHeader->nOffset += adtsHeaderSize;
604                        inHeader->nFilledLen -= adtsHeaderSize;
605                    }
606                }
607
608                if (signalError) {
609                    mSignalledError = true;
610                    notify(OMX_EventError, OMX_ErrorStreamCorrupt, ERROR_MALFORMED, NULL);
611                    return;
612                }
613            } else {
614                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
615                inBufferLength[0] = inHeader->nFilledLen;
616            }
617
618            // Fill and decode
619            bytesValid[0] = inBufferLength[0];
620
621            INT prevSampleRate = mStreamInfo->sampleRate;
622            INT prevNumChannels = mStreamInfo->numChannels;
623
624            if (inHeader != mLastInHeader) {
625                mLastInHeader = inHeader;
626                mCurrentInputTime = inHeader->nTimeStamp;
627            } else {
628                if (mStreamInfo->sampleRate) {
629                    mCurrentInputTime += mStreamInfo->aacSamplesPerFrame *
630                            1000000ll / mStreamInfo->sampleRate;
631                } else {
632                    ALOGW("no sample rate yet");
633                }
634            }
635            mAnchorTimes.add(mCurrentInputTime);
636            aacDecoder_Fill(mAACDecoder,
637                            inBuffer,
638                            inBufferLength,
639                            bytesValid);
640
641             // run DRC check
642             mDrcWrap.submitStreamData(mStreamInfo);
643             mDrcWrap.update();
644
645            AAC_DECODER_ERROR decoderErr =
646                aacDecoder_DecodeFrame(mAACDecoder,
647                                       tmpOutBuffer,
648                                       2048 * MAX_CHANNEL_COUNT,
649                                       0 /* flags */);
650
651            if (decoderErr != AAC_DEC_OK) {
652                ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
653            }
654
655            if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
656                ALOGE("AAC_DEC_NOT_ENOUGH_BITS should never happen");
657                mSignalledError = true;
658                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
659                return;
660            }
661
662            if (bytesValid[0] != 0) {
663                ALOGE("bytesValid[0] != 0 should never happen");
664                mSignalledError = true;
665                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
666                return;
667            }
668
669            size_t numOutBytes =
670                mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
671
672            if (decoderErr == AAC_DEC_OK) {
673                if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
674                        mStreamInfo->frameSize * mStreamInfo->numChannels)) {
675                    mSignalledError = true;
676                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
677                    return;
678                }
679                UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
680                inHeader->nFilledLen -= inBufferUsedLength;
681                inHeader->nOffset += inBufferUsedLength;
682            } else {
683                ALOGW("AAC decoder returned error 0x%4.4x, substituting silence", decoderErr);
684
685                memset(tmpOutBuffer, 0, numOutBytes); // TODO: check for overflow
686
687                if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
688                        mStreamInfo->frameSize * mStreamInfo->numChannels)) {
689                    mSignalledError = true;
690                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
691                    return;
692                }
693
694                // Discard input buffer.
695                inHeader->nFilledLen = 0;
696
697                aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
698
699                // fall through
700            }
701
702            /*
703             * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
704             * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
705             * rate system and the sampling rate in the final output is actually
706             * doubled compared with the core AAC decoder sampling rate.
707             *
708             * Explicit signalling is done by explicitly defining SBR audio object
709             * type in the bitstream. Implicit signalling is done by embedding
710             * SBR content in AAC extension payload specific to SBR, and hence
711             * requires an AAC decoder to perform pre-checks on actual audio frames.
712             *
713             * Thus, we could not say for sure whether a stream is
714             * AAC+/eAAC+ until the first data frame is decoded.
715             */
716            if (mInputBufferCount <= 2 || mOutputBufferCount > 1) { // TODO: <= 1
717                if (mStreamInfo->sampleRate != prevSampleRate ||
718                    mStreamInfo->numChannels != prevNumChannels) {
719                    ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
720                          prevSampleRate, mStreamInfo->sampleRate,
721                          prevNumChannels, mStreamInfo->numChannels);
722
723                    notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
724                    mOutputPortSettingsChange = AWAITING_DISABLED;
725
726                    if (inHeader->nFilledLen == 0) {
727                        inInfo->mOwnedByUs = false;
728                        mInputBufferCount++;
729                        inQueue.erase(inQueue.begin());
730                        mLastInHeader = NULL;
731                        inInfo = NULL;
732                        notifyEmptyBufferDone(inHeader);
733                        inHeader = NULL;
734                    }
735                    return;
736                }
737            } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
738                ALOGW("Invalid AAC stream");
739                mSignalledError = true;
740                notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
741                return;
742            }
743            if (inHeader->nFilledLen == 0) {
744                inInfo->mOwnedByUs = false;
745                mInputBufferCount++;
746                inQueue.erase(inQueue.begin());
747                mLastInHeader = NULL;
748                inInfo = NULL;
749                notifyEmptyBufferDone(inHeader);
750                inHeader = NULL;
751            } else {
752                ALOGV("inHeader->nFilledLen = %d", inHeader->nFilledLen);
753            }
754        }
755
756        int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
757
758        if (!mEndOfInput && mOutputDelayCompensated < outputDelay) {
759            // discard outputDelay at the beginning
760            int32_t toCompensate = outputDelay - mOutputDelayCompensated;
761            int32_t discard = outputDelayRingBufferSamplesAvailable();
762            if (discard > toCompensate) {
763                discard = toCompensate;
764            }
765            int32_t discarded = outputDelayRingBufferGetSamples(0, discard);
766            mOutputDelayCompensated += discarded;
767            continue;
768        }
769
770        if (mEndOfInput) {
771            while (mOutputDelayCompensated > 0) {
772                // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
773                INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
774
775                 // run DRC check
776                 mDrcWrap.submitStreamData(mStreamInfo);
777                 mDrcWrap.update();
778
779                AAC_DECODER_ERROR decoderErr =
780                    aacDecoder_DecodeFrame(mAACDecoder,
781                                           tmpOutBuffer,
782                                           2048 * MAX_CHANNEL_COUNT,
783                                           AACDEC_FLUSH);
784                if (decoderErr != AAC_DEC_OK) {
785                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
786                }
787
788                int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
789                if (tmpOutBufferSamples > mOutputDelayCompensated) {
790                    tmpOutBufferSamples = mOutputDelayCompensated;
791                }
792                outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
793                mOutputDelayCompensated -= tmpOutBufferSamples;
794            }
795        }
796
797        while (!outQueue.empty()
798                && outputDelayRingBufferSamplesAvailable()
799                        >= mStreamInfo->frameSize * mStreamInfo->numChannels) {
800            BufferInfo *outInfo = *outQueue.begin();
801            OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
802
803            if (outHeader->nOffset != 0) {
804                ALOGE("outHeader->nOffset != 0 is not handled");
805                mSignalledError = true;
806                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
807                return;
808            }
809
810            INT_PCM *outBuffer =
811                    reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);
812            if (outHeader->nOffset
813                    + mStreamInfo->frameSize * mStreamInfo->numChannels * sizeof(int16_t)
814                    > outHeader->nAllocLen) {
815                ALOGE("buffer overflow");
816                mSignalledError = true;
817                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
818                return;
819
820            }
821            int32_t ns = outputDelayRingBufferGetSamples(outBuffer,
822                    mStreamInfo->frameSize * mStreamInfo->numChannels); // TODO: check for overflow
823            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
824                ALOGE("not a complete frame of samples available");
825                mSignalledError = true;
826                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
827                return;
828            }
829
830            outHeader->nFilledLen = mStreamInfo->frameSize * mStreamInfo->numChannels
831                    * sizeof(int16_t);
832            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
833                outHeader->nFlags = OMX_BUFFERFLAG_EOS;
834                mEndOfOutput = true;
835            } else {
836                outHeader->nFlags = 0;
837            }
838
839            outHeader->nTimeStamp = mAnchorTimes.isEmpty() ? 0 : mAnchorTimes.itemAt(0);
840            mAnchorTimes.removeAt(0);
841
842            mOutputBufferCount++;
843            outInfo->mOwnedByUs = false;
844            outQueue.erase(outQueue.begin());
845            outInfo = NULL;
846            notifyFillBufferDone(outHeader);
847            outHeader = NULL;
848        }
849
850        if (mEndOfInput) {
851            if (outputDelayRingBufferSamplesAvailable() > 0
852                    && outputDelayRingBufferSamplesAvailable()
853                            < mStreamInfo->frameSize * mStreamInfo->numChannels) {
854                ALOGE("not a complete frame of samples available");
855                mSignalledError = true;
856                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
857                return;
858            }
859
860            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
861                if (!mEndOfOutput) {
862                    // send empty block signaling EOS
863                    mEndOfOutput = true;
864                    BufferInfo *outInfo = *outQueue.begin();
865                    OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
866
867                    if (outHeader->nOffset != 0) {
868                        ALOGE("outHeader->nOffset != 0 is not handled");
869                        mSignalledError = true;
870                        notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
871                        return;
872                    }
873
874                    INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer
875                            + outHeader->nOffset);
876                    int32_t ns = 0;
877                    outHeader->nFilledLen = 0;
878                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
879
880                    outHeader->nTimeStamp = mAnchorTimes.itemAt(0);
881                    mAnchorTimes.removeAt(0);
882
883                    mOutputBufferCount++;
884                    outInfo->mOwnedByUs = false;
885                    outQueue.erase(outQueue.begin());
886                    outInfo = NULL;
887                    notifyFillBufferDone(outHeader);
888                    outHeader = NULL;
889                }
890                break; // if outQueue not empty but no more output
891            }
892        }
893    }
894}
895
896void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
897    if (portIndex == 0) {
898        // Make sure that the next buffer output does not still
899        // depend on fragments from the last one decoded.
900        // drain all existing data
901        drainDecoder();
902        mAnchorTimes.clear();
903        mLastInHeader = NULL;
904    } else {
905        while (outputDelayRingBufferSamplesAvailable() > 0) {
906            int32_t ns = outputDelayRingBufferGetSamples(0,
907                    mStreamInfo->frameSize * mStreamInfo->numChannels);
908            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
909                ALOGE("not a complete frame of samples available");
910            }
911            mOutputBufferCount++;
912        }
913        mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
914    }
915}
916
917void SoftAAC2::drainDecoder() {
918    int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
919
920    // flush decoder until outputDelay is compensated
921    while (mOutputDelayCompensated > 0) {
922        // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
923        INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
924
925        // run DRC check
926        mDrcWrap.submitStreamData(mStreamInfo);
927        mDrcWrap.update();
928
929        AAC_DECODER_ERROR decoderErr =
930            aacDecoder_DecodeFrame(mAACDecoder,
931                                   tmpOutBuffer,
932                                   2048 * MAX_CHANNEL_COUNT,
933                                   AACDEC_FLUSH);
934        if (decoderErr != AAC_DEC_OK) {
935            ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
936        }
937
938        int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
939        if (tmpOutBufferSamples > mOutputDelayCompensated) {
940            tmpOutBufferSamples = mOutputDelayCompensated;
941        }
942        outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
943
944        mOutputDelayCompensated -= tmpOutBufferSamples;
945    }
946}
947
948void SoftAAC2::onReset() {
949    drainDecoder();
950    // reset the "configured" state
951    mInputBufferCount = 0;
952    mOutputBufferCount = 0;
953    mOutputDelayCompensated = 0;
954    mOutputDelayRingBufferWritePos = 0;
955    mOutputDelayRingBufferReadPos = 0;
956    mEndOfInput = false;
957    mEndOfOutput = false;
958    mAnchorTimes.clear();
959    mLastInHeader = NULL;
960
961    // To make the codec behave the same before and after a reset, we need to invalidate the
962    // streaminfo struct. This does that:
963    mStreamInfo->sampleRate = 0; // TODO: mStreamInfo is read only
964
965    mSignalledError = false;
966    mOutputPortSettingsChange = NONE;
967}
968
969void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
970    if (portIndex != 1) {
971        return;
972    }
973
974    switch (mOutputPortSettingsChange) {
975        case NONE:
976            break;
977
978        case AWAITING_DISABLED:
979        {
980            CHECK(!enabled);
981            mOutputPortSettingsChange = AWAITING_ENABLED;
982            break;
983        }
984
985        default:
986        {
987            CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
988            CHECK(enabled);
989            mOutputPortSettingsChange = NONE;
990            break;
991        }
992    }
993}
994
995}  // namespace android
996
997android::SoftOMXComponent *createSoftOMXComponent(
998        const char *name, const OMX_CALLBACKTYPE *callbacks,
999        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
1000    return new android::SoftAAC2(name, callbacks, appData, component);
1001}
1002