SoftAAC2.cpp revision 5f42113f21c31802e044f0a73351eef35e32feaf 
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 (portIndex == 0 &&
517                    (inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG) != 0) {
518                BufferInfo *inInfo = *inQueue.begin();
519                OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
520
521                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
522                inBufferLength[0] = inHeader->nFilledLen;
523
524                AAC_DECODER_ERROR decoderErr =
525                    aacDecoder_ConfigRaw(mAACDecoder,
526                                         inBuffer,
527                                         inBufferLength);
528
529                if (decoderErr != AAC_DEC_OK) {
530                    ALOGW("aacDecoder_ConfigRaw decoderErr = 0x%4.4x", decoderErr);
531                    mSignalledError = true;
532                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
533                    return;
534                }
535
536                mInputBufferCount++;
537                mOutputBufferCount++; // fake increase of outputBufferCount to keep the counters aligned
538
539                inInfo->mOwnedByUs = false;
540                inQueue.erase(inQueue.begin());
541                mLastInHeader = NULL;
542                inInfo = NULL;
543                notifyEmptyBufferDone(inHeader);
544                inHeader = NULL;
545
546                configureDownmix();
547                // Only send out port settings changed event if both sample rate
548                // and numChannels are valid.
549                if (mStreamInfo->sampleRate && mStreamInfo->numChannels) {
550                    ALOGI("Initially configuring decoder: %d Hz, %d channels",
551                        mStreamInfo->sampleRate,
552                        mStreamInfo->numChannels);
553
554                    notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
555                    mOutputPortSettingsChange = AWAITING_DISABLED;
556                }
557                return;
558            }
559
560            if (inHeader->nFilledLen == 0) {
561                inInfo->mOwnedByUs = false;
562                inQueue.erase(inQueue.begin());
563                mLastInHeader = NULL;
564                inInfo = NULL;
565                notifyEmptyBufferDone(inHeader);
566                inHeader = NULL;
567                continue;
568            }
569
570            if (mIsADTS) {
571                size_t adtsHeaderSize = 0;
572                // skip 30 bits, aac_frame_length follows.
573                // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
574
575                const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
576
577                bool signalError = false;
578                if (inHeader->nFilledLen < 7) {
579                    ALOGE("Audio data too short to contain even the ADTS header. "
580                            "Got %d bytes.", inHeader->nFilledLen);
581                    hexdump(adtsHeader, inHeader->nFilledLen);
582                    signalError = true;
583                } else {
584                    bool protectionAbsent = (adtsHeader[1] & 1);
585
586                    unsigned aac_frame_length =
587                        ((adtsHeader[3] & 3) << 11)
588                        | (adtsHeader[4] << 3)
589                        | (adtsHeader[5] >> 5);
590
591                    if (inHeader->nFilledLen < aac_frame_length) {
592                        ALOGE("Not enough audio data for the complete frame. "
593                                "Got %d bytes, frame size according to the ADTS "
594                                "header is %u bytes.",
595                                inHeader->nFilledLen, aac_frame_length);
596                        hexdump(adtsHeader, inHeader->nFilledLen);
597                        signalError = true;
598                    } else {
599                        adtsHeaderSize = (protectionAbsent ? 7 : 9);
600
601                        inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
602                        inBufferLength[0] = aac_frame_length - adtsHeaderSize;
603
604                        inHeader->nOffset += adtsHeaderSize;
605                        inHeader->nFilledLen -= adtsHeaderSize;
606                    }
607                }
608
609                if (signalError) {
610                    mSignalledError = true;
611                    notify(OMX_EventError, OMX_ErrorStreamCorrupt, ERROR_MALFORMED, NULL);
612                    return;
613                }
614            } else {
615                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
616                inBufferLength[0] = inHeader->nFilledLen;
617            }
618
619            // Fill and decode
620            bytesValid[0] = inBufferLength[0];
621
622            INT prevSampleRate = mStreamInfo->sampleRate;
623            INT prevNumChannels = mStreamInfo->numChannels;
624
625            if (inHeader != mLastInHeader) {
626                mLastInHeader = inHeader;
627                mCurrentInputTime = inHeader->nTimeStamp;
628            } else {
629                if (mStreamInfo->sampleRate) {
630                    mCurrentInputTime += mStreamInfo->aacSamplesPerFrame *
631                            1000000ll / mStreamInfo->sampleRate;
632                } else {
633                    ALOGW("no sample rate yet");
634                }
635            }
636            mAnchorTimes.add(mCurrentInputTime);
637            aacDecoder_Fill(mAACDecoder,
638                            inBuffer,
639                            inBufferLength,
640                            bytesValid);
641
642             // run DRC check
643             mDrcWrap.submitStreamData(mStreamInfo);
644             mDrcWrap.update();
645
646            AAC_DECODER_ERROR decoderErr =
647                aacDecoder_DecodeFrame(mAACDecoder,
648                                       tmpOutBuffer,
649                                       2048 * MAX_CHANNEL_COUNT,
650                                       0 /* flags */);
651
652            if (decoderErr != AAC_DEC_OK) {
653                ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
654            }
655
656            if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
657                ALOGE("AAC_DEC_NOT_ENOUGH_BITS should never happen");
658                mSignalledError = true;
659                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
660                return;
661            }
662
663            if (bytesValid[0] != 0) {
664                ALOGE("bytesValid[0] != 0 should never happen");
665                mSignalledError = true;
666                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
667                return;
668            }
669
670            size_t numOutBytes =
671                mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
672
673            if (decoderErr == AAC_DEC_OK) {
674                if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
675                        mStreamInfo->frameSize * mStreamInfo->numChannels)) {
676                    mSignalledError = true;
677                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
678                    return;
679                }
680                UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
681                inHeader->nFilledLen -= inBufferUsedLength;
682                inHeader->nOffset += inBufferUsedLength;
683            } else {
684                ALOGW("AAC decoder returned error 0x%4.4x, substituting silence", decoderErr);
685
686                memset(tmpOutBuffer, 0, numOutBytes); // TODO: check for overflow
687
688                if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
689                        mStreamInfo->frameSize * mStreamInfo->numChannels)) {
690                    mSignalledError = true;
691                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
692                    return;
693                }
694
695                // Discard input buffer.
696                inHeader->nFilledLen = 0;
697
698                aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
699
700                // fall through
701            }
702
703            /*
704             * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
705             * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
706             * rate system and the sampling rate in the final output is actually
707             * doubled compared with the core AAC decoder sampling rate.
708             *
709             * Explicit signalling is done by explicitly defining SBR audio object
710             * type in the bitstream. Implicit signalling is done by embedding
711             * SBR content in AAC extension payload specific to SBR, and hence
712             * requires an AAC decoder to perform pre-checks on actual audio frames.
713             *
714             * Thus, we could not say for sure whether a stream is
715             * AAC+/eAAC+ until the first data frame is decoded.
716             */
717            if (mInputBufferCount <= 2 || mOutputBufferCount > 1) { // TODO: <= 1
718                if (mStreamInfo->sampleRate != prevSampleRate ||
719                    mStreamInfo->numChannels != prevNumChannels) {
720                    ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
721                          prevSampleRate, mStreamInfo->sampleRate,
722                          prevNumChannels, mStreamInfo->numChannels);
723
724                    notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
725                    mOutputPortSettingsChange = AWAITING_DISABLED;
726
727                    if (inHeader->nFilledLen == 0) {
728                        inInfo->mOwnedByUs = false;
729                        mInputBufferCount++;
730                        inQueue.erase(inQueue.begin());
731                        mLastInHeader = NULL;
732                        inInfo = NULL;
733                        notifyEmptyBufferDone(inHeader);
734                        inHeader = NULL;
735                    }
736                    return;
737                }
738            } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
739                ALOGW("Invalid AAC stream");
740                mSignalledError = true;
741                notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
742                return;
743            }
744            if (inHeader->nFilledLen == 0) {
745                inInfo->mOwnedByUs = false;
746                mInputBufferCount++;
747                inQueue.erase(inQueue.begin());
748                mLastInHeader = NULL;
749                inInfo = NULL;
750                notifyEmptyBufferDone(inHeader);
751                inHeader = NULL;
752            } else {
753                ALOGV("inHeader->nFilledLen = %d", inHeader->nFilledLen);
754            }
755        }
756
757        int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
758
759        if (!mEndOfInput && mOutputDelayCompensated < outputDelay) {
760            // discard outputDelay at the beginning
761            int32_t toCompensate = outputDelay - mOutputDelayCompensated;
762            int32_t discard = outputDelayRingBufferSamplesAvailable();
763            if (discard > toCompensate) {
764                discard = toCompensate;
765            }
766            int32_t discarded = outputDelayRingBufferGetSamples(0, discard);
767            mOutputDelayCompensated += discarded;
768            continue;
769        }
770
771        if (mEndOfInput) {
772            while (mOutputDelayCompensated > 0) {
773                // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
774                INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
775
776                 // run DRC check
777                 mDrcWrap.submitStreamData(mStreamInfo);
778                 mDrcWrap.update();
779
780                AAC_DECODER_ERROR decoderErr =
781                    aacDecoder_DecodeFrame(mAACDecoder,
782                                           tmpOutBuffer,
783                                           2048 * MAX_CHANNEL_COUNT,
784                                           AACDEC_FLUSH);
785                if (decoderErr != AAC_DEC_OK) {
786                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
787                }
788
789                int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
790                if (tmpOutBufferSamples > mOutputDelayCompensated) {
791                    tmpOutBufferSamples = mOutputDelayCompensated;
792                }
793                outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
794                mOutputDelayCompensated -= tmpOutBufferSamples;
795            }
796        }
797
798        while (!outQueue.empty()
799                && outputDelayRingBufferSamplesAvailable()
800                        >= mStreamInfo->frameSize * mStreamInfo->numChannels) {
801            BufferInfo *outInfo = *outQueue.begin();
802            OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
803
804            if (outHeader->nOffset != 0) {
805                ALOGE("outHeader->nOffset != 0 is not handled");
806                mSignalledError = true;
807                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
808                return;
809            }
810
811            INT_PCM *outBuffer =
812                    reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);
813            if (outHeader->nOffset
814                    + mStreamInfo->frameSize * mStreamInfo->numChannels * sizeof(int16_t)
815                    > outHeader->nAllocLen) {
816                ALOGE("buffer overflow");
817                mSignalledError = true;
818                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
819                return;
820
821            }
822            int32_t ns = outputDelayRingBufferGetSamples(outBuffer,
823                    mStreamInfo->frameSize * mStreamInfo->numChannels); // TODO: check for overflow
824            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
825                ALOGE("not a complete frame of samples available");
826                mSignalledError = true;
827                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
828                return;
829            }
830
831            outHeader->nFilledLen = mStreamInfo->frameSize * mStreamInfo->numChannels
832                    * sizeof(int16_t);
833            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
834                outHeader->nFlags = OMX_BUFFERFLAG_EOS;
835                mEndOfOutput = true;
836            } else {
837                outHeader->nFlags = 0;
838            }
839
840            outHeader->nTimeStamp = mAnchorTimes.isEmpty() ? 0 : mAnchorTimes.itemAt(0);
841            mAnchorTimes.removeAt(0);
842
843            mOutputBufferCount++;
844            outInfo->mOwnedByUs = false;
845            outQueue.erase(outQueue.begin());
846            outInfo = NULL;
847            notifyFillBufferDone(outHeader);
848            outHeader = NULL;
849        }
850
851        if (mEndOfInput) {
852            if (outputDelayRingBufferSamplesAvailable() > 0
853                    && outputDelayRingBufferSamplesAvailable()
854                            < mStreamInfo->frameSize * mStreamInfo->numChannels) {
855                ALOGE("not a complete frame of samples available");
856                mSignalledError = true;
857                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
858                return;
859            }
860
861            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
862                if (!mEndOfOutput) {
863                    // send empty block signaling EOS
864                    mEndOfOutput = true;
865                    BufferInfo *outInfo = *outQueue.begin();
866                    OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
867
868                    if (outHeader->nOffset != 0) {
869                        ALOGE("outHeader->nOffset != 0 is not handled");
870                        mSignalledError = true;
871                        notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
872                        return;
873                    }
874
875                    INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer
876                            + outHeader->nOffset);
877                    int32_t ns = 0;
878                    outHeader->nFilledLen = 0;
879                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
880
881                    outHeader->nTimeStamp = mAnchorTimes.itemAt(0);
882                    mAnchorTimes.removeAt(0);
883
884                    mOutputBufferCount++;
885                    outInfo->mOwnedByUs = false;
886                    outQueue.erase(outQueue.begin());
887                    outInfo = NULL;
888                    notifyFillBufferDone(outHeader);
889                    outHeader = NULL;
890                }
891                break; // if outQueue not empty but no more output
892            }
893        }
894    }
895}
896
897void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
898    if (portIndex == 0) {
899        // Make sure that the next buffer output does not still
900        // depend on fragments from the last one decoded.
901        // drain all existing data
902        drainDecoder();
903        mAnchorTimes.clear();
904        mLastInHeader = NULL;
905    } else {
906        while (outputDelayRingBufferSamplesAvailable() > 0) {
907            int32_t ns = outputDelayRingBufferGetSamples(0,
908                    mStreamInfo->frameSize * mStreamInfo->numChannels);
909            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
910                ALOGE("not a complete frame of samples available");
911            }
912            mOutputBufferCount++;
913        }
914        mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
915    }
916}
917
918void SoftAAC2::drainDecoder() {
919    int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
920
921    // flush decoder until outputDelay is compensated
922    while (mOutputDelayCompensated > 0) {
923        // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
924        INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
925
926        // run DRC check
927        mDrcWrap.submitStreamData(mStreamInfo);
928        mDrcWrap.update();
929
930        AAC_DECODER_ERROR decoderErr =
931            aacDecoder_DecodeFrame(mAACDecoder,
932                                   tmpOutBuffer,
933                                   2048 * MAX_CHANNEL_COUNT,
934                                   AACDEC_FLUSH);
935        if (decoderErr != AAC_DEC_OK) {
936            ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
937        }
938
939        int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
940        if (tmpOutBufferSamples > mOutputDelayCompensated) {
941            tmpOutBufferSamples = mOutputDelayCompensated;
942        }
943        outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
944
945        mOutputDelayCompensated -= tmpOutBufferSamples;
946    }
947}
948
949void SoftAAC2::onReset() {
950    drainDecoder();
951    // reset the "configured" state
952    mInputBufferCount = 0;
953    mOutputBufferCount = 0;
954    mOutputDelayCompensated = 0;
955    mOutputDelayRingBufferWritePos = 0;
956    mOutputDelayRingBufferReadPos = 0;
957    mEndOfInput = false;
958    mEndOfOutput = false;
959    mAnchorTimes.clear();
960    mLastInHeader = NULL;
961
962    // To make the codec behave the same before and after a reset, we need to invalidate the
963    // streaminfo struct. This does that:
964    mStreamInfo->sampleRate = 0; // TODO: mStreamInfo is read only
965
966    mSignalledError = false;
967    mOutputPortSettingsChange = NONE;
968}
969
970void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
971    if (portIndex != 1) {
972        return;
973    }
974
975    switch (mOutputPortSettingsChange) {
976        case NONE:
977            break;
978
979        case AWAITING_DISABLED:
980        {
981            CHECK(!enabled);
982            mOutputPortSettingsChange = AWAITING_ENABLED;
983            break;
984        }
985
986        default:
987        {
988            CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
989            CHECK(enabled);
990            mOutputPortSettingsChange = NONE;
991            break;
992        }
993    }
994}
995
996}  // namespace android
997
998android::SoftOMXComponent *createSoftOMXComponent(
999        const char *name, const OMX_CALLBACKTYPE *callbacks,
1000        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
1001    return new android::SoftAAC2(name, callbacks, appData, component);
1002}
1003