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-rw-r--r--media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp557
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diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
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+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
@@ -0,0 +1,557 @@
+/*
+ * 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 "SoftAACEncoder2"
+#include <utils/Log.h>
+
+#include "SoftAACEncoder2.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;
+}
+
+SoftAACEncoder2::SoftAACEncoder2(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mAACEncoder(NULL),
+ mNumChannels(1),
+ mSampleRate(44100),
+ mBitRate(0),
+ mAACProfile(OMX_AUDIO_AACObjectLC),
+ mSentCodecSpecificData(false),
+ mInputSize(0),
+ mInputFrame(NULL),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+ setAudioParams();
+}
+
+SoftAACEncoder2::~SoftAACEncoder2() {
+ aacEncClose(&mAACEncoder);
+
+ delete[] mInputFrame;
+ mInputFrame = NULL;
+}
+
+void SoftAACEncoder2::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 SoftAACEncoder2::initEncoder() {
+ if (AACENC_OK != aacEncOpen(&mAACEncoder, 0, 0)) {
+ ALOGE("Failed to init AAC encoder");
+ return UNKNOWN_ERROR;
+ }
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAACEncoder2::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_AACPROFILETYPE) mAACProfile;
+ 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 SoftAACEncoder2::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 (aacParams->eAACProfile != OMX_AUDIO_AACObjectNull) {
+ mAACProfile = aacParams->eAACProfile;
+ }
+
+ 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);
+ }
+}
+
+CHANNEL_MODE getChannelMode(OMX_U32 nChannels) {
+ CHANNEL_MODE chMode = MODE_INVALID;
+ switch (nChannels) {
+ case 1: chMode = MODE_1; break;
+ case 2: chMode = MODE_2; break;
+ case 3: chMode = MODE_1_2; break;
+ case 4: chMode = MODE_1_2_1; break;
+ case 5: chMode = MODE_1_2_2; break;
+ case 6: chMode = MODE_1_2_2_1; break;
+ default: chMode = MODE_INVALID;
+ }
+ return chMode;
+}
+
+status_t SoftAACEncoder2::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);
+
+ if (AACENC_OK != aacEncoder_SetParam(mAACEncoder, AACENC_AOT,
+ mAACProfile == OMX_AUDIO_AACObjectELD ? AOT_ER_AAC_ELD : AOT_AAC_LC)) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+
+ if (AACENC_OK != aacEncoder_SetParam(mAACEncoder, AACENC_SAMPLERATE, mSampleRate)) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+ if (AACENC_OK != aacEncoder_SetParam(mAACEncoder, AACENC_BITRATE, mBitRate)) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+ if (AACENC_OK != aacEncoder_SetParam(mAACEncoder, AACENC_CHANNELMODE,
+ getChannelMode(mNumChannels))) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+ if (AACENC_OK != aacEncoder_SetParam(mAACEncoder, AACENC_TRANSMUX, TT_MP4_RAW)) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+void SoftAACEncoder2::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;
+ }
+
+ if (AACENC_OK != aacEncEncode(mAACEncoder, NULL, NULL, NULL, NULL)) {
+ ALOGE("Failed to initialize AAC encoder");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ mSignalledError = true;
+ return;
+ }
+
+ AACENC_InfoStruct encInfo;
+ if (AACENC_OK != aacEncInfo(mAACEncoder, &encInfo)) {
+ ALOGE("Failed to get AAC encoder info");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ mSignalledError = true;
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFilledLen = encInfo.confSize;
+ outHeader->nFlags = OMX_BUFFERFLAG_CODECCONFIG;
+
+ uint8_t *out = outHeader->pBuffer + outHeader->nOffset;
+ memcpy(out, encInfo.confBuf, encInfo.confSize);
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ mSentCodecSpecificData = true;
+ }
+
+ size_t numBytesPerInputFrame =
+ mNumChannels * kNumSamplesPerFrame * sizeof(int16_t);
+
+ // BUGBUG: Fraunhofer's decoder chokes on large chunks of AAC-ELD
+ if (mAACProfile == OMX_AUDIO_AACObjectELD && numBytesPerInputFrame > 512) {
+ numBytesPerInputFrame = 512;
+ }
+
+ 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) {
+ 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;
+
+ uint8_t *outPtr = (uint8_t *)outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ AACENC_InArgs inargs;
+ AACENC_OutArgs outargs;
+ memset(&inargs, 0, sizeof(inargs));
+ memset(&outargs, 0, sizeof(outargs));
+ inargs.numInSamples = numBytesPerInputFrame / sizeof(int16_t);
+
+ void* inBuffer[] = { (unsigned char *)mInputFrame };
+ INT inBufferIds[] = { IN_AUDIO_DATA };
+ INT inBufferSize[] = { numBytesPerInputFrame };
+ INT inBufferElSize[] = { sizeof(int16_t) };
+
+ AACENC_BufDesc inBufDesc;
+ inBufDesc.numBufs = sizeof(inBuffer) / sizeof(void*);
+ inBufDesc.bufs = (void**)&inBuffer;
+ inBufDesc.bufferIdentifiers = inBufferIds;
+ inBufDesc.bufSizes = inBufferSize;
+ inBufDesc.bufElSizes = inBufferElSize;
+
+ void* outBuffer[] = { outPtr };
+ INT outBufferIds[] = { OUT_BITSTREAM_DATA };
+ INT outBufferSize[] = { 0 };
+ INT outBufferElSize[] = { sizeof(UCHAR) };
+
+ AACENC_BufDesc outBufDesc;
+ outBufDesc.numBufs = sizeof(outBuffer) / sizeof(void*);
+ outBufDesc.bufs = (void**)&outBuffer;
+ outBufDesc.bufferIdentifiers = outBufferIds;
+ outBufDesc.bufSizes = outBufferSize;
+ outBufDesc.bufElSizes = outBufferElSize;
+
+ // Encode the mInputFrame, which is treated as a modulo buffer
+ AACENC_ERROR encoderErr = AACENC_OK;
+ size_t nOutputBytes = 0;
+ do {
+ memset(&outargs, 0, sizeof(outargs));
+
+ outBuffer[0] = outPtr;
+ outBufferSize[0] = outAvailable - nOutputBytes;
+
+ encoderErr = aacEncEncode(mAACEncoder,
+ &inBufDesc,
+ &outBufDesc,
+ &inargs,
+ &outargs);
+
+ if (encoderErr == AACENC_OK) {
+ outPtr += outargs.numOutBytes;
+ nOutputBytes += outargs.numOutBytes;
+
+ if (outargs.numInSamples > 0) {
+ int numRemainingSamples = inargs.numInSamples - outargs.numInSamples;
+ if (numRemainingSamples > 0) {
+ memmove(mInputFrame,
+ &mInputFrame[outargs.numInSamples],
+ sizeof(int16_t) * numRemainingSamples);
+ }
+ inargs.numInSamples -= outargs.numInSamples;
+ }
+ }
+ } while (encoderErr == AACENC_OK && inargs.numInSamples > 0);
+
+ 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::SoftAACEncoder2(name, callbacks, appData, component);
+}
generated by cgit v1.1 at 2024-06-12 17:07:36 +0000