The software AAC encoder is now an OMX component.
Yay.
Change-Id: I74938a20b4e0a622836ea5184d3761180eb0f5de
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 613435d..7703058 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -38,7 +38,7 @@
static const int32_t kAudioBitRate = 12200;
static const int64_t kDurationUs = 10000000LL; // 10 seconds
-#if 1
+#if 0
class DummySource : public MediaSource {
public:
@@ -318,7 +318,7 @@
sp<MetaData> encMeta = new MetaData;
encMeta->setCString(kKeyMIMEType,
- 1 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
+ 0 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
encMeta->setInt32(kKeySampleRate, kSampleRate);
encMeta->setInt32(kKeyChannelCount, kNumChannels);
encMeta->setInt32(kKeyMaxInputSize, 8192);
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index 84f8282..4c30e04 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -172,6 +172,7 @@
uint32_t mFlags;
bool mIsEncoder;
+ bool mIsVideo;
char *mMIME;
char *mComponentName;
sp<MetaData> mOutputFormat;
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 0aeb515..4d61067 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -77,7 +77,6 @@
LOCAL_STATIC_LIBRARIES := \
libstagefright_color_conversion \
- libstagefright_aacenc \
libstagefright_amrnbenc \
libstagefright_amrwbenc \
libstagefright_avcenc \
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 60d9bb7..7597f64 100755
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -18,7 +18,6 @@
#define LOG_TAG "OMXCodec"
#include <utils/Log.h>
-#include "include/AACEncoder.h"
#include "include/AMRNBEncoder.h"
#include "include/AMRWBEncoder.h"
#include "include/AVCEncoder.h"
@@ -73,7 +72,6 @@
FACTORY_CREATE_ENCODER(AMRNBEncoder)
FACTORY_CREATE_ENCODER(AMRWBEncoder)
-FACTORY_CREATE_ENCODER(AACEncoder)
FACTORY_CREATE_ENCODER(AVCEncoder)
FACTORY_CREATE_ENCODER(M4vH263Encoder)
@@ -88,7 +86,6 @@
static const FactoryInfo kFactoryInfo[] = {
FACTORY_REF(AMRNBEncoder)
FACTORY_REF(AMRWBEncoder)
- FACTORY_REF(AACEncoder)
FACTORY_REF(AVCEncoder)
FACTORY_REF(M4vH263Encoder)
};
@@ -153,7 +150,7 @@
{ MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.TI.WBAMR.encode" },
{ MEDIA_MIMETYPE_AUDIO_AMR_WB, "AMRWBEncoder" },
{ MEDIA_MIMETYPE_AUDIO_AAC, "OMX.TI.AAC.encode" },
- { MEDIA_MIMETYPE_AUDIO_AAC, "AACEncoder" },
+ { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.google.aac.encoder" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.DUCATI1.VIDEO.MPEG4E" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.7x30.video.encoder.mpeg4" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.video.encoder.mpeg4" },
@@ -1487,6 +1484,7 @@
mQuirks(quirks),
mFlags(flags),
mIsEncoder(isEncoder),
+ mIsVideo(!strncasecmp("video/", mime, 6)),
mMIME(strdup(mime)),
mComponentName(strdup(componentName)),
mSource(source),
@@ -2192,7 +2190,7 @@
}
int64_t OMXCodec::retrieveDecodingTimeUs(bool isCodecSpecific) {
- CHECK(mIsEncoder);
+ CHECK(mIsEncoder && mIsVideo);
if (mDecodingTimeList.empty()) {
CHECK(mSignalledEOS || mNoMoreOutputData);
@@ -2387,7 +2385,7 @@
mNoMoreOutputData = true;
}
- if (mIsEncoder) {
+ if (mIsEncoder && mIsVideo) {
int64_t decodingTimeUs = retrieveDecodingTimeUs(isCodecSpecific);
buffer->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
}
@@ -3249,7 +3247,7 @@
int64_t lastBufferTimeUs;
CHECK(srcBuffer->meta_data()->findInt64(kKeyTime, &lastBufferTimeUs));
CHECK(lastBufferTimeUs >= 0);
- if (mIsEncoder) {
+ if (mIsEncoder && mIsVideo) {
mDecodingTimeList.push_back(lastBufferTimeUs);
}
diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk
index 8318ba4..34a2796 100644
--- a/media/libstagefright/codecs/aacenc/Android.mk
+++ b/media/libstagefright/codecs/aacenc/Android.mk
@@ -85,3 +85,29 @@
endif
include $(BUILD_STATIC_LIBRARY)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAACEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/include/media/stagefright/openmax \
+ frameworks/base/media/libstagefright/codecs/common/include \
+
+LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_aacenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_enc_common
+
+LOCAL_MODULE := libstagefright_soft_aacenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
new file mode 100644
index 0000000..c6724c2
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
@@ -0,0 +1,560 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAACEncoder"
+#include <utils/Log.h>
+
+#include "SoftAACEncoder.h"
+
+#include "voAAC.h"
+#include "cmnMemory.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAACEncoder::SoftAACEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncoderHandle(NULL),
+ mApiHandle(NULL),
+ mMemOperator(NULL),
+ mNumChannels(1),
+ mSampleRate(44100),
+ mBitRate(0),
+ mSentCodecSpecificData(false),
+ mInputSize(0),
+ mInputFrame(NULL),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+
+ setAudioParams();
+}
+
+SoftAACEncoder::~SoftAACEncoder() {
+ delete[] mInputFrame;
+ mInputFrame = NULL;
+
+ if (mEncoderHandle) {
+ CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ mEncoderHandle = NULL;
+ }
+
+ delete mApiHandle;
+ mApiHandle = NULL;
+
+ delete mMemOperator;
+ mMemOperator = NULL;
+}
+
+void SoftAACEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t) * 2;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+
+ addPort(def);
+}
+
+status_t SoftAACEncoder::initEncoder() {
+ mApiHandle = new VO_AUDIO_CODECAPI;
+
+ if (VO_ERR_NONE != voGetAACEncAPI(mApiHandle)) {
+ ALOGE("Failed to get api handle");
+ return UNKNOWN_ERROR;
+ }
+
+ mMemOperator = new VO_MEM_OPERATOR;
+ mMemOperator->Alloc = cmnMemAlloc;
+ mMemOperator->Copy = cmnMemCopy;
+ mMemOperator->Free = cmnMemFree;
+ mMemOperator->Set = cmnMemSet;
+ mMemOperator->Check = cmnMemCheck;
+
+ VO_CODEC_INIT_USERDATA userData;
+ memset(&userData, 0, sizeof(userData));
+ userData.memflag = VO_IMF_USERMEMOPERATOR;
+ userData.memData = (VO_PTR) mMemOperator;
+ if (VO_ERR_NONE !=
+ mApiHandle->Init(&mEncoderHandle, VO_AUDIO_CodingAAC, &userData)) {
+ ALOGE("Failed to init AAC encoder");
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAAC;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAac:
+ {
+ OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+ (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+ if (aacParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ aacParams->nBitRate = mBitRate;
+ aacParams->nAudioBandWidth = 0;
+ aacParams->nAACtools = 0;
+ aacParams->nAACERtools = 0;
+ aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
+ aacParams->eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4FF;
+ aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
+
+ aacParams->nChannels = mNumChannels;
+ aacParams->nSampleRate = mSampleRate;
+ aacParams->nFrameLength = 0;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+ pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+
+ pcmParams->nChannels = mNumChannels;
+ pcmParams->nSamplingRate = mSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.aac",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAAC)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAac:
+ {
+ OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+ (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+ if (aacParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = aacParams->nBitRate;
+ mNumChannels = aacParams->nChannels;
+ mSampleRate = aacParams->nSampleRate;
+
+ if (setAudioParams() != OK) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ mNumChannels = pcmParams->nChannels;
+ mSampleRate = pcmParams->nSamplingRate;
+
+ if (setAudioParams() != OK) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+status_t SoftAACEncoder::setAudioParams() {
+ // We call this whenever sample rate, number of channels or bitrate change
+ // in reponse to setParameter calls.
+
+ ALOGV("setAudioParams: %lu Hz, %lu channels, %lu bps",
+ mSampleRate, mNumChannels, mBitRate);
+
+ status_t err = setAudioSpecificConfigData();
+
+ if (err != OK) {
+ return err;
+ }
+
+ AACENC_PARAM params;
+ memset(¶ms, 0, sizeof(params));
+ params.sampleRate = mSampleRate;
+ params.bitRate = mBitRate;
+ params.nChannels = mNumChannels;
+ params.adtsUsed = 0; // We add adts header in the file writer if needed.
+ if (VO_ERR_NONE != mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AAC_ENCPARAM, ¶ms)) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) {
+ static const int32_t kSampleRateTable[] = {
+ 96000, 88200, 64000, 48000, 44100, 32000,
+ 24000, 22050, 16000, 12000, 11025, 8000
+ };
+ const int32_t tableSize =
+ sizeof(kSampleRateTable) / sizeof(kSampleRateTable[0]);
+
+ for (int32_t i = 0; i < tableSize; ++i) {
+ if (sampleRate == kSampleRateTable[i]) {
+ index = i;
+ return OK;
+ }
+ }
+
+ return UNKNOWN_ERROR;
+}
+
+status_t SoftAACEncoder::setAudioSpecificConfigData() {
+ // The AAC encoder's audio specific config really only encodes
+ // number of channels and the sample rate (mapped to an index into
+ // a fixed sample rate table).
+
+ int32_t index;
+ status_t err = getSampleRateTableIndex(mSampleRate, index);
+ if (err != OK) {
+ ALOGE("Unsupported sample rate (%lu Hz)", mSampleRate);
+ return err;
+ }
+
+ if (mNumChannels > 2 || mNumChannels <= 0) {
+ ALOGE("Unsupported number of channels(%lu)", mNumChannels);
+ return UNKNOWN_ERROR;
+ }
+
+ // OMX_AUDIO_AACObjectLC
+ mAudioSpecificConfigData[0] = ((0x02 << 3) | (index >> 1));
+ mAudioSpecificConfigData[1] = ((index & 0x01) << 7) | (mNumChannels << 3);
+
+ return OK;
+}
+
+void SoftAACEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ if (!mSentCodecSpecificData) {
+ // The very first thing we want to output is the codec specific
+ // data. It does not require any input data but we will need an
+ // output buffer to store it in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFilledLen = sizeof(mAudioSpecificConfigData);
+ outHeader->nFlags = OMX_BUFFERFLAG_CODECCONFIG;
+
+ uint8_t *out = outHeader->pBuffer + outHeader->nOffset;
+ memcpy(out, mAudioSpecificConfigData, sizeof(mAudioSpecificConfigData));
+
+#if 0
+ ALOGI("sending codec specific data.");
+ hexdump(out, sizeof(mAudioSpecificConfigData));
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ mSentCodecSpecificData = true;
+ }
+
+ size_t numBytesPerInputFrame =
+ mNumChannels * kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame * mNumChannels" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputFrame == NULL) {
+ mInputFrame = new int16_t[kNumSamplesPerFrame * mNumChannels];
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / mSampleRate)
+ / (mNumChannels * sizeof(int16_t));
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ VO_CODECBUFFER inputData;
+ memset(&inputData, 0, sizeof(inputData));
+ inputData.Buffer = (unsigned char *)mInputFrame;
+ inputData.Length = numBytesPerInputFrame;
+ CHECK(VO_ERR_NONE ==
+ mApiHandle->SetInputData(mEncoderHandle, &inputData));
+
+ VO_CODECBUFFER outputData;
+ memset(&outputData, 0, sizeof(outputData));
+ VO_AUDIO_OUTPUTINFO outputInfo;
+ memset(&outputInfo, 0, sizeof(outputInfo));
+
+ uint8_t *outPtr = (uint8_t *)outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ VO_U32 ret = VO_ERR_NONE;
+ size_t nOutputBytes = 0;
+ do {
+ outputData.Buffer = outPtr;
+ outputData.Length = outAvailable - nOutputBytes;
+ ret = mApiHandle->GetOutputData(
+ mEncoderHandle, &outputData, &outputInfo);
+ if (ret == VO_ERR_NONE) {
+ outPtr += outputData.Length;
+ nOutputBytes += outputData.Length;
+ }
+ } while (ret != VO_ERR_INPUT_BUFFER_SMALL);
+
+ outHeader->nFilledLen = nOutputBytes;
+
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
+ nOutputBytes, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAACEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.h b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
new file mode 100644
index 0000000..d148eb7
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+#ifndef SOFT_AAC_ENCODER_H_
+
+#define SOFT_AAC_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+struct VO_AUDIO_CODECAPI;
+struct VO_MEM_OPERATOR;
+
+namespace android {
+
+struct SoftAACEncoder : public SimpleSoftOMXComponent {
+ SoftAACEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAACEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 1024,
+ };
+
+ void *mEncoderHandle;
+ VO_AUDIO_CODECAPI *mApiHandle;
+ VO_MEM_OPERATOR *mMemOperator;
+
+ OMX_U32 mNumChannels;
+ OMX_U32 mSampleRate;
+ OMX_U32 mBitRate;
+
+ bool mSentCodecSpecificData;
+ size_t mInputSize;
+ int16_t *mInputFrame;
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+
+ uint8_t mAudioSpecificConfigData[2];
+
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ status_t setAudioSpecificConfigData();
+ status_t setAudioParams();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAACEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AAC_ENCODER_H_
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index da3ae42..cf9e8c9 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -35,6 +35,7 @@
} kComponents[] = {
{ "OMX.google.aac.decoder", "aacdec", "audio_decoder.aac" },
+ { "OMX.google.aac.encoder", "aacenc", "audio_encoder.aac" },
{ "OMX.google.amrnb.decoder", "amrdec", "audio_decoder.amrnb" },
{ "OMX.google.amrwb.decoder", "amrdec", "audio_decoder.amrwb" },
{ "OMX.google.h264.decoder", "h264dec", "video_decoder.avc" },