| /* |
| * Copyright (C) 2009 The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * limitations under the License. |
| */ |
| |
| #include "AACDecoder.h" |
| #define LOG_TAG "AACDecoder" |
| |
| #include "../../include/ESDS.h" |
| |
| #include "pvmp4audiodecoder_api.h" |
| |
| #include <media/stagefright/foundation/ADebug.h> |
| #include <media/stagefright/MediaBufferGroup.h> |
| #include <media/stagefright/MediaDefs.h> |
| #include <media/stagefright/MetaData.h> |
| |
| namespace android { |
| |
| AACDecoder::AACDecoder(const sp<MediaSource> &source) |
| : mSource(source), |
| mStarted(false), |
| mBufferGroup(NULL), |
| mConfig(new tPVMP4AudioDecoderExternal), |
| mDecoderBuf(NULL), |
| mAnchorTimeUs(0), |
| mNumSamplesOutput(0), |
| mInputBuffer(NULL) { |
| |
| sp<MetaData> srcFormat = mSource->getFormat(); |
| |
| int32_t sampleRate; |
| CHECK(srcFormat->findInt32(kKeySampleRate, &sampleRate)); |
| |
| mMeta = new MetaData; |
| mMeta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW); |
| |
| // We'll always output stereo, regardless of how many channels are |
| // present in the input due to decoder limitations. |
| mMeta->setInt32(kKeyChannelCount, 2); |
| mMeta->setInt32(kKeySampleRate, sampleRate); |
| |
| int64_t durationUs; |
| if (srcFormat->findInt64(kKeyDuration, &durationUs)) { |
| mMeta->setInt64(kKeyDuration, durationUs); |
| } |
| mMeta->setCString(kKeyDecoderComponent, "AACDecoder"); |
| |
| mInitCheck = initCheck(); |
| } |
| |
| status_t AACDecoder::initCheck() { |
| memset(mConfig, 0, sizeof(tPVMP4AudioDecoderExternal)); |
| mConfig->outputFormat = OUTPUTFORMAT_16PCM_INTERLEAVED; |
| mConfig->aacPlusEnabled = 1; |
| |
| // The software decoder doesn't properly support mono output on |
| // AACplus files. Always output stereo. |
| mConfig->desiredChannels = 2; |
| |
| UInt32 memRequirements = PVMP4AudioDecoderGetMemRequirements(); |
| mDecoderBuf = malloc(memRequirements); |
| |
| status_t err = PVMP4AudioDecoderInitLibrary(mConfig, mDecoderBuf); |
| if (err != MP4AUDEC_SUCCESS) { |
| LOGE("Failed to initialize MP4 audio decoder"); |
| return UNKNOWN_ERROR; |
| } |
| |
| uint32_t type; |
| const void *data; |
| size_t size; |
| sp<MetaData> meta = mSource->getFormat(); |
| if (meta->findData(kKeyESDS, &type, &data, &size)) { |
| ESDS esds((const char *)data, size); |
| CHECK_EQ(esds.InitCheck(), (status_t)OK); |
| |
| const void *codec_specific_data; |
| size_t codec_specific_data_size; |
| esds.getCodecSpecificInfo( |
| &codec_specific_data, &codec_specific_data_size); |
| |
| mConfig->pInputBuffer = (UChar *)codec_specific_data; |
| mConfig->inputBufferCurrentLength = codec_specific_data_size; |
| mConfig->inputBufferMaxLength = 0; |
| |
| if (PVMP4AudioDecoderConfig(mConfig, mDecoderBuf) |
| != MP4AUDEC_SUCCESS) { |
| return ERROR_UNSUPPORTED; |
| } |
| } |
| return OK; |
| } |
| |
| AACDecoder::~AACDecoder() { |
| if (mStarted) { |
| stop(); |
| } |
| |
| delete mConfig; |
| mConfig = NULL; |
| } |
| |
| status_t AACDecoder::start(MetaData *params) { |
| CHECK(!mStarted); |
| |
| mBufferGroup = new MediaBufferGroup; |
| mBufferGroup->add_buffer(new MediaBuffer(4096 * 2)); |
| |
| mSource->start(); |
| |
| mAnchorTimeUs = 0; |
| mNumSamplesOutput = 0; |
| mStarted = true; |
| mNumDecodedBuffers = 0; |
| mUpsamplingFactor = 2; |
| |
| return OK; |
| } |
| |
| status_t AACDecoder::stop() { |
| CHECK(mStarted); |
| |
| if (mInputBuffer) { |
| mInputBuffer->release(); |
| mInputBuffer = NULL; |
| } |
| |
| free(mDecoderBuf); |
| mDecoderBuf = NULL; |
| |
| delete mBufferGroup; |
| mBufferGroup = NULL; |
| |
| mSource->stop(); |
| |
| mStarted = false; |
| |
| return OK; |
| } |
| |
| sp<MetaData> AACDecoder::getFormat() { |
| return mMeta; |
| } |
| |
| status_t AACDecoder::read( |
| MediaBuffer **out, const ReadOptions *options) { |
| 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; |
| } |
| |
| if (mInputBuffer == NULL) { |
| err = mSource->read(&mInputBuffer, options); |
| |
| if (err != OK) { |
| return err; |
| } |
| |
| int64_t timeUs; |
| if (mInputBuffer->meta_data()->findInt64(kKeyTime, &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]; |
| mConfig->repositionFlag = false; |
| |
| Int decoderErr = PVMP4AudioDecodeFrame(mConfig, mDecoderBuf); |
| |
| /* |
| * 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) { |
| LOGV("audio/extended audio object type: %d + %d", |
| mConfig->audioObjectType, mConfig->extendedAudioObjectType); |
| LOGV("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)); |
| if (mConfig->samplingRate != sampleRate) { |
| mMeta->setInt32(kKeySampleRate, mConfig->samplingRate); |
| LOGW("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 |
| LOGW("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 |
| LOGW("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) { |
| LOGW("AAC decoder returned error %d, substituting silence", decoderErr); |
| |
| memset(buffer->data(), 0, numOutBytes); |
| |
| // Discard input buffer. |
| mInputBuffer->release(); |
| mInputBuffer = NULL; |
| |
| // fall through |
| } |
| |
| buffer->set_range(0, numOutBytes); |
| |
| if (mInputBuffer != NULL) { |
| mInputBuffer->set_range( |
| mInputBuffer->range_offset() + mConfig->inputBufferUsedLength, |
| mInputBuffer->range_length() - mConfig->inputBufferUsedLength); |
| |
| if (mInputBuffer->range_length() == 0) { |
| mInputBuffer->release(); |
| mInputBuffer = NULL; |
| } |
| } |
| |
| buffer->meta_data()->setInt64( |
| kKeyTime, |
| mAnchorTimeUs |
| + (mNumSamplesOutput * 1000000) / mConfig->samplingRate); |
| |
| mNumSamplesOutput += mConfig->frameLength * mUpsamplingFactor; |
| |
| *out = buffer; |
| |
| return OK; |
| } |
| |
| } // namespace android |