SoftAAC2.cpp revision ab7f4182d4d509733107622216db4dd128340185 
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    mOutputDelayRingBufferFilled = 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 (numSamples == 0) {
413        return true;
414    }
415    if (outputDelayRingBufferSpaceLeft() < numSamples) {
416        ALOGE("RING BUFFER WOULD OVERFLOW");
417        return false;
418    }
419    if (mOutputDelayRingBufferWritePos + numSamples <= mOutputDelayRingBufferSize
420            && (mOutputDelayRingBufferReadPos <= mOutputDelayRingBufferWritePos
421                    || mOutputDelayRingBufferReadPos > mOutputDelayRingBufferWritePos + numSamples)) {
422        // faster memcopy loop without checks, if the preconditions allow this
423        for (int32_t i = 0; i < numSamples; i++) {
424            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos++] = samples[i];
425        }
426
427        if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
428            mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
429        }
430    } else {
431        ALOGV("slow SoftAAC2::outputDelayRingBufferPutSamples()");
432
433        for (int32_t i = 0; i < numSamples; i++) {
434            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos] = samples[i];
435            mOutputDelayRingBufferWritePos++;
436            if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
437                mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
438            }
439        }
440    }
441    mOutputDelayRingBufferFilled += numSamples;
442    return true;
443}
444
445int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numSamples) {
446
447    if (numSamples > mOutputDelayRingBufferFilled) {
448        ALOGE("RING BUFFER WOULD UNDERRUN");
449        return -1;
450    }
451
452    if (mOutputDelayRingBufferReadPos + numSamples <= mOutputDelayRingBufferSize
453            && (mOutputDelayRingBufferWritePos < mOutputDelayRingBufferReadPos
454                    || mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferReadPos + numSamples)) {
455        // faster memcopy loop without checks, if the preconditions allow this
456        if (samples != 0) {
457            for (int32_t i = 0; i < numSamples; i++) {
458                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos++];
459            }
460        } else {
461            mOutputDelayRingBufferReadPos += numSamples;
462        }
463        if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
464            mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
465        }
466    } else {
467        ALOGV("slow SoftAAC2::outputDelayRingBufferGetSamples()");
468
469        for (int32_t i = 0; i < numSamples; i++) {
470            if (samples != 0) {
471                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos];
472            }
473            mOutputDelayRingBufferReadPos++;
474            if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
475                mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
476            }
477        }
478    }
479    mOutputDelayRingBufferFilled -= numSamples;
480    return numSamples;
481}
482
483int32_t SoftAAC2::outputDelayRingBufferSamplesAvailable() {
484    return mOutputDelayRingBufferFilled;
485}
486
487int32_t SoftAAC2::outputDelayRingBufferSpaceLeft() {
488    return mOutputDelayRingBufferSize - outputDelayRingBufferSamplesAvailable();
489}
490
491
492void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
493    if (mSignalledError || mOutputPortSettingsChange != NONE) {
494        return;
495    }
496
497    UCHAR* inBuffer[FILEREAD_MAX_LAYERS];
498    UINT inBufferLength[FILEREAD_MAX_LAYERS] = {0};
499    UINT bytesValid[FILEREAD_MAX_LAYERS] = {0};
500
501    List<BufferInfo *> &inQueue = getPortQueue(0);
502    List<BufferInfo *> &outQueue = getPortQueue(1);
503
504    while ((!inQueue.empty() || mEndOfInput) && !outQueue.empty()) {
505        if (!inQueue.empty()) {
506            INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
507            BufferInfo *inInfo = *inQueue.begin();
508            OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
509
510            mEndOfInput = (inHeader->nFlags & OMX_BUFFERFLAG_EOS) != 0;
511
512            if (mInputBufferCount == 0 && !(inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
513                ALOGE("first buffer should have OMX_BUFFERFLAG_CODECCONFIG set");
514                inHeader->nFlags |= OMX_BUFFERFLAG_CODECCONFIG;
515            }
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
614                // insert buffer size and time stamp
615                mBufferSizes.add(inBufferLength[0]);
616                if (mLastInHeader != inHeader) {
617                    mBufferTimestamps.add(inHeader->nTimeStamp);
618                    mLastInHeader = inHeader;
619                } else {
620                    int64_t currentTime = mBufferTimestamps.top();
621                    currentTime += mStreamInfo->aacSamplesPerFrame *
622                            1000000ll / mStreamInfo->sampleRate;
623                    mBufferTimestamps.add(currentTime);
624                }
625            } else {
626                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
627                inBufferLength[0] = inHeader->nFilledLen;
628                mLastInHeader = inHeader;
629                mBufferTimestamps.add(inHeader->nTimeStamp);
630                mBufferSizes.add(inHeader->nFilledLen);
631            }
632
633            // Fill and decode
634            bytesValid[0] = inBufferLength[0];
635
636            INT prevSampleRate = mStreamInfo->sampleRate;
637            INT prevNumChannels = mStreamInfo->numChannels;
638
639            aacDecoder_Fill(mAACDecoder,
640                            inBuffer,
641                            inBufferLength,
642                            bytesValid);
643
644            // run DRC check
645            mDrcWrap.submitStreamData(mStreamInfo);
646            mDrcWrap.update();
647
648            UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
649            inHeader->nFilledLen -= inBufferUsedLength;
650            inHeader->nOffset += inBufferUsedLength;
651
652            AAC_DECODER_ERROR decoderErr;
653            do {
654                if (outputDelayRingBufferSpaceLeft() <
655                        (mStreamInfo->frameSize * mStreamInfo->numChannels)) {
656                    ALOGV("skipping decode: not enough space left in ringbuffer");
657                    break;
658                }
659
660                int numconsumed = mStreamInfo->numTotalBytes + mStreamInfo->numBadBytes;
661                decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
662                                           tmpOutBuffer,
663                                           2048 * MAX_CHANNEL_COUNT,
664                                           0 /* flags */);
665
666                numconsumed = (mStreamInfo->numTotalBytes + mStreamInfo->numBadBytes) - numconsumed;
667                if (numconsumed != 0) {
668                    mDecodedSizes.add(numconsumed);
669                }
670
671                if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
672                    break;
673                }
674
675                if (decoderErr != AAC_DEC_OK) {
676                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
677                }
678
679                if (bytesValid[0] != 0) {
680                    ALOGE("bytesValid[0] != 0 should never happen");
681                    mSignalledError = true;
682                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
683                    return;
684                }
685
686                size_t numOutBytes =
687                    mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
688
689                if (decoderErr == AAC_DEC_OK) {
690                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
691                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
692                        mSignalledError = true;
693                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
694                        return;
695                    }
696                } else {
697                    ALOGW("AAC decoder returned error 0x%4.4x, substituting silence", decoderErr);
698
699                    memset(tmpOutBuffer, 0, numOutBytes); // TODO: check for overflow
700
701                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
702                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
703                        mSignalledError = true;
704                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
705                        return;
706                    }
707
708                    // Discard input buffer.
709                    if (inHeader) {
710                        inHeader->nFilledLen = 0;
711                    }
712
713                    aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
714
715                    // fall through
716                }
717
718                /*
719                 * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
720                 * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
721                 * rate system and the sampling rate in the final output is actually
722                 * doubled compared with the core AAC decoder sampling rate.
723                 *
724                 * Explicit signalling is done by explicitly defining SBR audio object
725                 * type in the bitstream. Implicit signalling is done by embedding
726                 * SBR content in AAC extension payload specific to SBR, and hence
727                 * requires an AAC decoder to perform pre-checks on actual audio frames.
728                 *
729                 * Thus, we could not say for sure whether a stream is
730                 * AAC+/eAAC+ until the first data frame is decoded.
731                 */
732                if (mInputBufferCount <= 2 || mOutputBufferCount > 1) { // TODO: <= 1
733                    if (mStreamInfo->sampleRate != prevSampleRate ||
734                        mStreamInfo->numChannels != prevNumChannels) {
735                        ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
736                              prevSampleRate, mStreamInfo->sampleRate,
737                              prevNumChannels, mStreamInfo->numChannels);
738
739                        notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
740                        mOutputPortSettingsChange = AWAITING_DISABLED;
741
742                        if (inHeader && inHeader->nFilledLen == 0) {
743                            inInfo->mOwnedByUs = false;
744                            mInputBufferCount++;
745                            inQueue.erase(inQueue.begin());
746                            mLastInHeader = NULL;
747                            inInfo = NULL;
748                            notifyEmptyBufferDone(inHeader);
749                            inHeader = NULL;
750                        }
751                        return;
752                    }
753                } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
754                    ALOGW("Invalid AAC stream");
755                    mSignalledError = true;
756                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
757                    return;
758                }
759                if (inHeader && inHeader->nFilledLen == 0) {
760                    inInfo->mOwnedByUs = false;
761                    mInputBufferCount++;
762                    inQueue.erase(inQueue.begin());
763                    mLastInHeader = NULL;
764                    inInfo = NULL;
765                    notifyEmptyBufferDone(inHeader);
766                    inHeader = NULL;
767                } else {
768                    ALOGV("inHeader->nFilledLen = %d", inHeader ? inHeader->nFilledLen : 0);
769                }
770            } while (decoderErr == AAC_DEC_OK);
771        }
772
773        int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
774
775        if (!mEndOfInput && mOutputDelayCompensated < outputDelay) {
776            // discard outputDelay at the beginning
777            int32_t toCompensate = outputDelay - mOutputDelayCompensated;
778            int32_t discard = outputDelayRingBufferSamplesAvailable();
779            if (discard > toCompensate) {
780                discard = toCompensate;
781            }
782            int32_t discarded = outputDelayRingBufferGetSamples(0, discard);
783            mOutputDelayCompensated += discarded;
784            continue;
785        }
786
787        if (mEndOfInput) {
788            while (mOutputDelayCompensated > 0) {
789                // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
790                INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
791
792                 // run DRC check
793                 mDrcWrap.submitStreamData(mStreamInfo);
794                 mDrcWrap.update();
795
796                AAC_DECODER_ERROR decoderErr =
797                    aacDecoder_DecodeFrame(mAACDecoder,
798                                           tmpOutBuffer,
799                                           2048 * MAX_CHANNEL_COUNT,
800                                           AACDEC_FLUSH);
801                if (decoderErr != AAC_DEC_OK) {
802                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
803                }
804
805                int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
806                if (tmpOutBufferSamples > mOutputDelayCompensated) {
807                    tmpOutBufferSamples = mOutputDelayCompensated;
808                }
809                outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
810                mOutputDelayCompensated -= tmpOutBufferSamples;
811            }
812        }
813
814        while (!outQueue.empty()
815                && outputDelayRingBufferSamplesAvailable()
816                        >= mStreamInfo->frameSize * mStreamInfo->numChannels) {
817            BufferInfo *outInfo = *outQueue.begin();
818            OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
819
820            if (outHeader->nOffset != 0) {
821                ALOGE("outHeader->nOffset != 0 is not handled");
822                mSignalledError = true;
823                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
824                return;
825            }
826
827            INT_PCM *outBuffer =
828                    reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);
829            int samplesize = mStreamInfo->numChannels * sizeof(int16_t);
830            if (outHeader->nOffset
831                    + mStreamInfo->frameSize * samplesize
832                    > outHeader->nAllocLen) {
833                ALOGE("buffer overflow");
834                mSignalledError = true;
835                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
836                return;
837
838            }
839
840            int available = outputDelayRingBufferSamplesAvailable();
841            int numSamples = outHeader->nAllocLen / sizeof(int16_t);
842            if (numSamples > available) {
843                numSamples = available;
844            }
845            int64_t currentTime = 0;
846            if (available) {
847
848                int numFrames = numSamples / (mStreamInfo->frameSize * mStreamInfo->numChannels);
849                numSamples = numFrames * (mStreamInfo->frameSize * mStreamInfo->numChannels);
850
851                ALOGV("%d samples available (%d), or %d frames",
852                        numSamples, available, numFrames);
853                int64_t *nextTimeStamp = &mBufferTimestamps.editItemAt(0);
854                currentTime = *nextTimeStamp;
855                int32_t *currentBufLeft = &mBufferSizes.editItemAt(0);
856                for (int i = 0; i < numFrames; i++) {
857                    int32_t decodedSize = mDecodedSizes.itemAt(0);
858                    mDecodedSizes.removeAt(0);
859                    ALOGV("decoded %d of %d", decodedSize, *currentBufLeft);
860                    if (*currentBufLeft > decodedSize) {
861                        // adjust/interpolate next time stamp
862                        *currentBufLeft -= decodedSize;
863                        *nextTimeStamp += mStreamInfo->aacSamplesPerFrame *
864                                1000000ll / mStreamInfo->sampleRate;
865                        ALOGV("adjusted nextTimeStamp/size to %lld/%d",
866                                *nextTimeStamp, *currentBufLeft);
867                    } else {
868                        // move to next timestamp in list
869                        if (mBufferTimestamps.size() > 0) {
870                            mBufferTimestamps.removeAt(0);
871                            nextTimeStamp = &mBufferTimestamps.editItemAt(0);
872                            mBufferSizes.removeAt(0);
873                            currentBufLeft = &mBufferSizes.editItemAt(0);
874                            ALOGV("moved to next time/size: %lld/%d",
875                                    *nextTimeStamp, *currentBufLeft);
876                        }
877                        // try to limit output buffer size to match input buffers
878                        // (e.g when an input buffer contained 4 "sub" frames, output
879                        // at most 4 decoded units in the corresponding output buffer)
880                        // This is optional. Remove the next three lines to fill the output
881                        // buffer with as many units as available.
882                        numFrames = i + 1;
883                        numSamples = numFrames * mStreamInfo->frameSize * mStreamInfo->numChannels;
884                        break;
885                    }
886                }
887
888                ALOGV("getting %d from ringbuffer", numSamples);
889                int32_t ns = outputDelayRingBufferGetSamples(outBuffer, numSamples);
890                if (ns != numSamples) {
891                    ALOGE("not a complete frame of samples available");
892                    mSignalledError = true;
893                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
894                    return;
895                }
896            }
897
898            outHeader->nFilledLen = numSamples * sizeof(int16_t);
899
900            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
901                outHeader->nFlags = OMX_BUFFERFLAG_EOS;
902                mEndOfOutput = true;
903            } else {
904                outHeader->nFlags = 0;
905            }
906
907            outHeader->nTimeStamp = currentTime;
908
909            mOutputBufferCount++;
910            outInfo->mOwnedByUs = false;
911            outQueue.erase(outQueue.begin());
912            outInfo = NULL;
913            ALOGV("out timestamp %lld / %d", outHeader->nTimeStamp, outHeader->nFilledLen);
914            notifyFillBufferDone(outHeader);
915            outHeader = NULL;
916        }
917
918        if (mEndOfInput) {
919            if (outputDelayRingBufferSamplesAvailable() > 0
920                    && outputDelayRingBufferSamplesAvailable()
921                            < mStreamInfo->frameSize * mStreamInfo->numChannels) {
922                ALOGE("not a complete frame of samples available");
923                mSignalledError = true;
924                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
925                return;
926            }
927
928            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
929                if (!mEndOfOutput) {
930                    // send empty block signaling EOS
931                    mEndOfOutput = true;
932                    BufferInfo *outInfo = *outQueue.begin();
933                    OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
934
935                    if (outHeader->nOffset != 0) {
936                        ALOGE("outHeader->nOffset != 0 is not handled");
937                        mSignalledError = true;
938                        notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
939                        return;
940                    }
941
942                    INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer
943                            + outHeader->nOffset);
944                    int32_t ns = 0;
945                    outHeader->nFilledLen = 0;
946                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
947
948                    outHeader->nTimeStamp = mBufferTimestamps.itemAt(0);
949                    mBufferTimestamps.clear();
950                    mBufferSizes.clear();
951                    mDecodedSizes.clear();
952
953                    mOutputBufferCount++;
954                    outInfo->mOwnedByUs = false;
955                    outQueue.erase(outQueue.begin());
956                    outInfo = NULL;
957                    notifyFillBufferDone(outHeader);
958                    outHeader = NULL;
959                }
960                break; // if outQueue not empty but no more output
961            }
962        }
963    }
964}
965
966void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
967    if (portIndex == 0) {
968        // Make sure that the next buffer output does not still
969        // depend on fragments from the last one decoded.
970        // drain all existing data
971        drainDecoder();
972        mBufferTimestamps.clear();
973        mBufferSizes.clear();
974        mDecodedSizes.clear();
975        mLastInHeader = NULL;
976    } else {
977        while (outputDelayRingBufferSamplesAvailable() > 0) {
978            int32_t ns = outputDelayRingBufferGetSamples(0,
979                    mStreamInfo->frameSize * mStreamInfo->numChannels);
980            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
981                ALOGE("not a complete frame of samples available");
982            }
983            mOutputBufferCount++;
984        }
985        mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
986    }
987}
988
989void SoftAAC2::drainDecoder() {
990    int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
991
992    // flush decoder until outputDelay is compensated
993    while (mOutputDelayCompensated > 0) {
994        // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
995        INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
996
997        // run DRC check
998        mDrcWrap.submitStreamData(mStreamInfo);
999        mDrcWrap.update();
1000
1001        AAC_DECODER_ERROR decoderErr =
1002            aacDecoder_DecodeFrame(mAACDecoder,
1003                                   tmpOutBuffer,
1004                                   2048 * MAX_CHANNEL_COUNT,
1005                                   AACDEC_FLUSH);
1006        if (decoderErr != AAC_DEC_OK) {
1007            ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
1008        }
1009
1010        int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
1011        if (tmpOutBufferSamples > mOutputDelayCompensated) {
1012            tmpOutBufferSamples = mOutputDelayCompensated;
1013        }
1014        outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
1015
1016        mOutputDelayCompensated -= tmpOutBufferSamples;
1017    }
1018}
1019
1020void SoftAAC2::onReset() {
1021    drainDecoder();
1022    // reset the "configured" state
1023    mInputBufferCount = 0;
1024    mOutputBufferCount = 0;
1025    mOutputDelayCompensated = 0;
1026    mOutputDelayRingBufferWritePos = 0;
1027    mOutputDelayRingBufferReadPos = 0;
1028    mOutputDelayRingBufferFilled = 0;
1029    mEndOfInput = false;
1030    mEndOfOutput = false;
1031    mBufferTimestamps.clear();
1032    mBufferSizes.clear();
1033    mDecodedSizes.clear();
1034    mLastInHeader = NULL;
1035
1036    // To make the codec behave the same before and after a reset, we need to invalidate the
1037    // streaminfo struct. This does that:
1038    mStreamInfo->sampleRate = 0; // TODO: mStreamInfo is read only
1039
1040    mSignalledError = false;
1041    mOutputPortSettingsChange = NONE;
1042}
1043
1044void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
1045    if (portIndex != 1) {
1046        return;
1047    }
1048
1049    switch (mOutputPortSettingsChange) {
1050        case NONE:
1051            break;
1052
1053        case AWAITING_DISABLED:
1054        {
1055            CHECK(!enabled);
1056            mOutputPortSettingsChange = AWAITING_ENABLED;
1057            break;
1058        }
1059
1060        default:
1061        {
1062            CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
1063            CHECK(enabled);
1064            mOutputPortSettingsChange = NONE;
1065            break;
1066        }
1067    }
1068}
1069
1070}  // namespace android
1071
1072android::SoftOMXComponent *createSoftOMXComponent(
1073        const char *name, const OMX_CALLBACKTYPE *callbacks,
1074        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
1075    return new android::SoftAAC2(name, callbacks, appData, component);
1076}
1077