void OmxAacDecoder::ResetDecoder()
{
if (ipMem && (iAacInitFlag != 0))
{
PVMP4AudioDecoderResetBuffer(ipMem);
}
}
// Initialize the aac reader.
int aac_init(audio_decoder_p decoder, void *dec_ext)
{
decoder->dec_ext = calloc(1, sizeof(tPVMP4AudioDecoderExternal));
RETURN_VAL_IF_FAIL((decoder->dec_ext != NULL), AUDIO_DECODER_ERROR);
decoder->dec_mem = calloc(1, PVMP4AudioDecoderGetMemRequirements());
RETURN_VAL_IF_FAIL((decoder->dec_mem != NULL), AUDIO_DECODER_ERROR);
*((tPVMP4AudioDecoderExternal *) decoder->dec_ext) = *((tPVMP4AudioDecoderExternal *) dec_ext);
PVMP4AudioDecoderResetBuffer(decoder->dec_mem);
Int err = PVMP4AudioDecoderInitLibrary((tPVMP4AudioDecoderExternal *) decoder->dec_ext, decoder->dec_mem);
RETURN_VAL_IF_FAIL((err == MP4AUDEC_SUCCESS), AUDIO_DECODER_ERROR);
return AUDIO_DECODER_OK;
}
int aac_decode(char *outbuf, const char *inbuf, int len)
{
int Status;
/* seek to first adts sync */
if( !dec_sync )
{
uint8_t *inp = (uint8_t *)inbuf;
if( (inp[0] == 0xFF) && ((inp[1] & 0xF6) == 0xF0) )
{
dec_sync = 1;
}
else
return 0;
}
append_inbuf(inbuf, len);
/* needed for some stream encoded on neon. */
if( pExt->inputBufferCurrentLength - pExt->inputBufferUsedLength < 256 )
{
return 0;
}
//pExt->pInputBuffer = (uint8_t *)inbuf;
pExt->pOutputBuffer = (short *)outbuf;
pExt->pOutputBuffer_plus = (short *)(outbuf+4096);
//pExt->inputBufferUsedLength=0;
//pExt->inputBufferCurrentLength = len;
if( !dec_configed )
{
Status = PVMP4AudioDecoderConfig(pExt, pMem);
/*fprintf(stderr, "config %d. used %d, current %d\n",
Status,
pExt->inputBufferUsedLength,
pExt->inputBufferCurrentLength);*/
}
Status = PVMP4AudioDecodeFrame(pExt, pMem);
if (MP4AUDEC_SUCCESS == Status || SUCCESS == Status) {
/*fprintf(stderr, "[SUCCESS] Status: SUCCESS "
"inputBufferUsedLength: %u, "
"inputBufferCurrentLength: %u, "
"remainderBits: %u, "
"frameLength: %u\n",
pExt->inputBufferUsedLength,
pExt->inputBufferCurrentLength,
pExt->remainderBits,
pExt->frameLength);*/
if( pExt->frameLength > 0 && !dec_configed )
{
dec_configed = 1;
}
}
else
{
fprintf(stderr, "aac decode error %d\n", Status);
fprintf(stderr, "used %d, current %d\n",
pExt->inputBufferUsedLength,
pExt->inputBufferCurrentLength);
#if 1
if( pExt->inputBufferUsedLength > pExt->inputBufferCurrentLength )
{
/* this is ridiculous, but does happen..
* try to re-init the decoder. */
dec_sync = 0;
dec_configed = 0;
pExt->inputBufferUsedLength=0;
pExt->inputBufferCurrentLength = 0;
PVMP4AudioDecoderResetBuffer(pMem);
PVMP4AudioDecoderInitLibrary(pExt, pMem);
}
#endif
return -1;
}
return pExt->frameLength*4;
}
/*
-----------------------------------------------------------------------------
CDecoder_AAC
ResetDecoderL
Stop decoder object. Reset decoder.
Parameters: none
Return Values: status
-----------------------------------------------------------------------------
*/
OSCL_EXPORT_REF void CDecoder_AAC::ResetDecoderL()
{
PVMP4AudioDecoderResetBuffer(pMem);
}
status_t AACDecoder::read(
MediaBuffer **out, const ReadOptions *options) {
/*
** end of aac audio stream in the case of initCheck is not OK,
** avoid abnormal playing later. @Jun 16, 2011. by hbb
*/
if(mInitCheck != OK) {
ALOGE("mInitCheck is not OK, so end aac audio stream");
return ERROR_END_OF_STREAM;
}
status_t err;
*out = NULL;
int64_t seekTimeUs;
ReadOptions::SeekMode mode;
if (options && options->getSeekTo(&seekTimeUs, &mode)) {
CHECK(seekTimeUs >= 0);
mNumSamplesOutput = 0;
if (mInputBuffer) {
mInputBuffer->release();
mInputBuffer = NULL;
}
// Make sure that the next buffer output does not still
// depend on fragments from the last one decoded.
PVMP4AudioDecoderResetBuffer(mDecoderBuf);
} else {
seekTimeUs = -1;
}
repeat:
if (mInputBuffer == NULL) {
err = mSource->read(&mInputBuffer, options);
if (err != OK) {
return err;
}
int64_t timeUs;
if (mInputBuffer->meta_data()->findInt64(kKeyTime, &timeUs)) {
if(mAnchorTimeUs != timeUs)
{
mAnchorTimeUs = timeUs;
mNumSamplesOutput = 0;
}
} else {
// We must have a new timestamp after seeking.
CHECK(seekTimeUs < 0);
}
}
MediaBuffer *buffer;
CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
mConfig->pInputBuffer =
(UChar *)mInputBuffer->data() + mInputBuffer->range_offset();
mConfig->inputBufferCurrentLength = mInputBuffer->range_length();
mConfig->inputBufferMaxLength = 0;
mConfig->inputBufferUsedLength = 0;
mConfig->remainderBits = 0;
mConfig->pOutputBuffer = static_cast<Int16 *>(buffer->data());
mConfig->pOutputBuffer_plus = &mConfig->pOutputBuffer[2048];
#if WRITE_FILE
if(aacFp)
fwrite(mConfig->pInputBuffer,1, mConfig->inputBufferCurrentLength,aacFp);
#endif
//ALOGE("inputlen %d input[0] = %x input[1]%x", mConfig->inputBufferCurrentLength,(mConfig->pInputBuffer)[0],(mConfig->pInputBuffer)[1]);
Int decoderErr = MP4AUDEC_SUCCESS;
if(mConfig->isMutilChannle)
{
decoderErr = PVMP4AudioDecodeFrameSixChannel(mConfig, mDecoderBuf);
}
else
{
decoderErr = PVMP4AudioDecodeFrame(mConfig, mDecoderBuf);
}
if (mInputBuffer != NULL) {
mInputBuffer->set_range(
mInputBuffer->range_offset() + mConfig->inputBufferUsedLength,
mInputBuffer->range_length() - mConfig->inputBufferUsedLength);
if (mInputBuffer->range_length() <= 3) {
mInputBuffer->release();
mInputBuffer = NULL;
}
}
//if the input data no enough,will drop this frame inputdata. get the next frame data.
if(decoderErr != MP4AUDEC_SUCCESS)
{
if(mInputBuffer)
{
mInputBuffer->release();
mInputBuffer = NULL;
}
if(buffer)
{
buffer->release();
buffer = NULL;
}
goto repeat;
}
/*
* AAC+/eAAC+ streams can be signalled in two ways: either explicitly
* or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
* rate system and the sampling rate in the final output is actually
* doubled compared with the core AAC decoder sampling rate.
*
* Explicit signalling is done by explicitly defining SBR audio object
* type in the bitstream. Implicit signalling is done by embedding
* SBR content in AAC extension payload specific to SBR, and hence
* requires an AAC decoder to perform pre-checks on actual audio frames.
*
* Thus, we could not say for sure whether a stream is
* AAC+/eAAC+ until the first data frame is decoded.
*/
if (++mNumDecodedBuffers <= 2) {
ALOGV("audio/extended audio object type: %d + %d",
mConfig->audioObjectType, mConfig->extendedAudioObjectType);
ALOGV("aac+ upsampling factor: %d desired channels: %d",
mConfig->aacPlusUpsamplingFactor, mConfig->desiredChannels);
CHECK(mNumDecodedBuffers > 0);
if (mNumDecodedBuffers == 1) {
mUpsamplingFactor = mConfig->aacPlusUpsamplingFactor;
// Check on the sampling rate to see whether it is changed.
int32_t sampleRate;
CHECK(mMeta->findInt32(kKeySampleRate, &sampleRate));
ALOGV("--->aac samplerae %d",sampleRate);
if (mConfig->samplingRate != sampleRate) {
mMeta->setInt32(kKeySampleRate, mConfig->samplingRate);
ALOGV("Sample rate was %d Hz, but now is %d Hz",
sampleRate, mConfig->samplingRate);
buffer->release();
// mInputBuffer->release();
// mInputBuffer = NULL;
return INFO_FORMAT_CHANGED;
}
} else { // mNumDecodedBuffers == 2
if (mConfig->extendedAudioObjectType == MP4AUDIO_AAC_LC ||
mConfig->extendedAudioObjectType == MP4AUDIO_LTP) {
if (mUpsamplingFactor == 2) {
// The stream turns out to be not aacPlus mode anyway
ALOGV("Disable AAC+/eAAC+ since extended audio object type is %d",
mConfig->extendedAudioObjectType);
mConfig->aacPlusEnabled = 0;
}
} else {
if (mUpsamplingFactor == 1) {
// aacPlus mode does not buy us anything, but to cause
// 1. CPU load to increase, and
// 2. a half speed of decoding
ALOGV("Disable AAC+/eAAC+ since upsampling factor is 1");
mConfig->aacPlusEnabled = 0;
}
}
}
}
size_t numOutBytes =
mConfig->frameLength * sizeof(int16_t) * mConfig->desiredChannels;
if (mUpsamplingFactor == 2) {
if (mConfig->desiredChannels == 1) {
memcpy(&mConfig->pOutputBuffer[1024], &mConfig->pOutputBuffer[2048], numOutBytes * 2);
}
numOutBytes *= 2;
}
if (decoderErr != MP4AUDEC_SUCCESS) {
ALOGW("AAC decoder returned error %d, substituting silence", decoderErr);
memset(buffer->data(), 0, numOutBytes);
// Discard input buffer.
if(mInputBuffer)
{
mInputBuffer->release();
mInputBuffer = NULL;
}
// fall through
}
buffer->set_range(0, numOutBytes);
buffer->meta_data()->setInt64(
kKeyTime,
mAnchorTimeUs
+ (mNumSamplesOutput * 1000000) / mConfig->samplingRate);
mNumSamplesOutput += mConfig->frameLength;
*out = buffer;
return OK;
}
