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authorDharmaray Kundargi <dharmaray@google.com>2011-01-16 16:02:42 -0800
committerDharmaray Kundargi <dharmaray@google.com>2011-01-16 19:09:33 -0800
commit643290dc4c83da23b1b8ff4ed71118203274bb15 (patch)
treefd23ca748eca6352c665c6dba3b625e1679d61c4 /libvideoeditor/lvpp/VideoEditorSRC.cpp
parent5358e878396e1c451e9f9ef07237c2e6ab662d49 (diff)
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videoeditor preview code on honeycomb
Change-Id: I9c3c9cb921ea697ab16732973d26ef9035cda2ee
Diffstat (limited to 'libvideoeditor/lvpp/VideoEditorSRC.cpp')
-rwxr-xr-xlibvideoeditor/lvpp/VideoEditorSRC.cpp391
1 files changed, 391 insertions, 0 deletions
diff --git a/libvideoeditor/lvpp/VideoEditorSRC.cpp b/libvideoeditor/lvpp/VideoEditorSRC.cpp
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+++ b/libvideoeditor/lvpp/VideoEditorSRC.cpp
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+/*
+ * Copyright (C) 2011 NXP Software
+ * Copyright (C) 2011 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 1
+#define LOG_TAG "VEAudioSource"
+#include <utils/Log.h>
+
+
+#include "VideoEditorSRC.h"
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaDebug.h>
+#include "AudioMixer.h"
+
+
+namespace android {
+
+VideoEditorSRC::VideoEditorSRC(
+ const sp<MediaSource> &source) {
+
+ LOGV("VideoEditorSRC::Create");
+ mSource = source;
+ mResampler = NULL;
+ mBitDepth = 16;
+ mChannelCnt = 0;
+ mSampleRate = 0;
+ mOutputSampleRate = DEFAULT_SAMPLING_FREQ;
+ mStarted = false;
+ mIsResamplingRequired = false;
+ mIsChannelConvertionRequired = false;
+ mInitialTimeStampUs = -1;
+ mAccuOutBufferSize = 0;
+ mSeekTimeUs = -1;
+ mLeftover = 0;
+ mLastReadSize = 0;
+#ifndef FROYO
+ mSeekMode = ReadOptions::SEEK_PREVIOUS_SYNC;
+#endif
+
+ mOutputFormat = new MetaData;
+
+ // Input Source validation
+ sp<MetaData> format = mSource->getFormat();
+ const char *mime;
+ bool success = format->findCString(kKeyMIMEType, &mime);
+ CHECK(success);
+ CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW));
+
+ //set the meta data of the output after convertion.
+ if(mOutputFormat != NULL) {
+ mOutputFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
+ mOutputFormat->setInt32(kKeySampleRate, DEFAULT_SAMPLING_FREQ);
+
+ //by default we convert all data to stereo
+ mOutputFormat->setInt32(kKeyChannelCount, 2);
+ } else {
+ LOGE("Meta data was not allocated.");
+ }
+
+ // Allocate a 1 sec buffer (test only, to be refined)
+ mInterframeBufferPosition = 0;
+ pInterframeBuffer = new uint8_t[DEFAULT_SAMPLING_FREQ * 2 * 2]; //stereo=2 * bytespersample=2
+
+
+}
+
+VideoEditorSRC::~VideoEditorSRC(){
+ if (mStarted == true)
+ stop();
+
+ if(mOutputFormat != NULL) {
+ mOutputFormat.clear();
+ mOutputFormat = NULL;
+ }
+
+ if (pInterframeBuffer != NULL){
+ delete pInterframeBuffer;
+ pInterframeBuffer = NULL;
+ }
+}
+
+void VideoEditorSRC::setResampling(int32_t sampleRate) {
+ Mutex::Autolock autoLock(mLock);
+ LOGV("VideoEditorSRC::setResampling called with samplreRate = %d", sampleRate);
+ if(sampleRate != DEFAULT_SAMPLING_FREQ) { //default case
+ LOGV("VideoEditor Audio resampler, freq set is other than default");
+ CHECK(mOutputFormat->setInt32(kKeySampleRate, DEFAULT_SAMPLING_FREQ));
+ }
+ mOutputSampleRate = sampleRate;
+ return;
+}
+
+// debug
+FILE *fp;
+
+
+status_t VideoEditorSRC::start (MetaData *params) {
+ Mutex::Autolock autoLock(mLock);
+
+ CHECK(!mStarted);
+ LOGV(" VideoEditorSRC:start() called");
+
+ sp<MetaData> format = mSource->getFormat();
+ const char *mime;
+ bool success = format->findCString(kKeyMIMEType, &mime);
+ CHECK(success);
+ CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW));
+
+ success = format->findInt32(kKeySampleRate, &mSampleRate);
+ CHECK(success);
+
+ int32_t numChannels;
+ success = format->findInt32(kKeyChannelCount, &mChannelCnt);
+ CHECK(success);
+
+ mInputFrameSize = mChannelCnt * 2; //2 is the byte depth
+ if(mSampleRate != mOutputSampleRate) {
+ LOGV("Resampling required (%d != %d)", mSampleRate, mOutputSampleRate);
+ mIsResamplingRequired = true;
+ LOGV("Create resampler %d %d %d", mBitDepth, mChannelCnt, mOutputSampleRate);
+
+ mResampler = AudioResampler::create(
+ mBitDepth, mChannelCnt, mOutputSampleRate, AudioResampler::DEFAULT);
+
+ if(mResampler == NULL) {
+ return NO_MEMORY;
+ }
+ LOGV("Set input rate %d", mSampleRate);
+ mResampler->setSampleRate(mSampleRate);
+ mResampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+
+ } else {
+ if(mChannelCnt != 2) { //we always make sure to provide stereo
+ LOGV("Only Channel convertion required");
+ mIsChannelConvertionRequired = true;
+ }
+ }
+ mSeekTimeUs = -1;
+#ifndef FROYO
+ mSeekMode = ReadOptions::SEEK_PREVIOUS_SYNC;
+#endif
+ mStarted = true;
+ mSource->start();
+
+ //+ debug
+ fp = fopen("/sdcard/output.pcm", "wb");
+
+ return OK;
+}
+
+status_t VideoEditorSRC::stop() {
+
+ Mutex::Autolock autoLock(mLock);
+ LOGV("VideoEditorSRC::stop()");
+ mSource->stop();
+ if(mResampler != NULL) {
+ delete mResampler;
+ mResampler = NULL;
+ }
+ mStarted = false;
+ mInitialTimeStampUs = -1;
+ mAccuOutBufferSize = 0;
+ mLeftover = 0;
+ mLastReadSize = 0;
+
+ //+ debug
+ fclose(fp);
+ return OK;
+}
+
+sp<MetaData> VideoEditorSRC::getFormat() {
+ LOGV("AudioSRC getFormat");
+ //Mutex::Autolock autoLock(mLock);
+ return mOutputFormat;
+}
+
+status_t VideoEditorSRC::read (
+ MediaBuffer **buffer_out, const ReadOptions *options) {
+ Mutex::Autolock autoLock(mLock);
+ *buffer_out = NULL;
+ int32_t leftover = 0;
+
+ LOGV("VideoEditorSRC::read");
+
+ if (!mStarted) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ if(mIsResamplingRequired == true) {
+
+ LOGV("mIsResamplingRequired = true");
+
+ // Store the seek parameters
+ int64_t seekTimeUs;
+#ifndef FROYO
+ ReadOptions::SeekMode mode = ReadOptions::SEEK_PREVIOUS_SYNC;
+ if (options && options->getSeekTo(&seekTimeUs, &mode)) {
+#else
+ if (options && options->getSeekTo(&seekTimeUs)) {
+#endif
+ LOGV("read Seek %lld", seekTimeUs);
+ mInitialTimeStampUs = -1;
+ mSeekTimeUs = seekTimeUs;
+#ifndef FROYO
+ mSeekMode = mode;
+#else
+ mReadOptions = *options;
+#endif
+ }
+
+ // We ask for 1024 frames in output
+ size_t outFrameCnt = 1024;
+ int32_t outBufferSize = (outFrameCnt) * 2 * sizeof(int16_t); //out is always 2 channels & 16 bits
+ int64_t outDurationUs = (outBufferSize * 1000000) /(mOutputSampleRate * 2 * sizeof(int16_t)); //2 channels out * 2 bytes per sample
+ LOGV("outBufferSize %d", outBufferSize);
+ LOGV("outFrameCnt %d", outFrameCnt);
+
+ pTmpBuffer = (int32_t*)malloc(outFrameCnt * 2 * sizeof(int32_t)); //out is always 2 channels and resampler out is 32 bits
+ memset(pTmpBuffer, 0x00, outFrameCnt * 2 * sizeof(int32_t));
+ // Resample to target quality
+ mResampler->resample(pTmpBuffer, outFrameCnt, this);
+ int16_t *reSampledBuffer = (int16_t*)malloc(outBufferSize);
+ memset(reSampledBuffer, 0x00, outBufferSize);
+
+ // Convert back to 16 bits
+ AudioMixer::ditherAndClamp((int32_t*)reSampledBuffer, pTmpBuffer, outFrameCnt);
+ LOGV("Resampled buffer size %d", outFrameCnt* 2 * sizeof(int16_t));
+
+ // Create new MediaBuffer
+ mCopyBuffer = new MediaBuffer((void*)reSampledBuffer, outBufferSize);
+
+ // Compute and set the new timestamp
+ sp<MetaData> to = mCopyBuffer->meta_data();
+ int64_t totalOutDurationUs = (mAccuOutBufferSize * 1000000) /(mOutputSampleRate * 2 * 2); //2 channels out * 2 bytes per sample
+ int64_t timeUs = mInitialTimeStampUs + totalOutDurationUs;
+ to->setInt64(kKeyTime, timeUs);
+ LOGV("buffer duration %lld timestamp %lld init %lld", outDurationUs, timeUs, mInitialTimeStampUs);
+
+ // update the accumulate size
+ mAccuOutBufferSize += outBufferSize;
+
+ mCopyBuffer->set_range(0, outBufferSize);
+ *buffer_out = mCopyBuffer;
+
+ free(pTmpBuffer);
+
+ } else if(mIsChannelConvertionRequired == true) {
+ //TODO convert to stereo here.
+ } else {
+ //LOGI("Resampling not required");
+ MediaBuffer *aBuffer;
+ status_t err = mSource->read(&aBuffer, options);
+ LOGV("mSource->read returned %d", err);
+ if(err != OK) {
+ *buffer_out = NULL;
+ mStarted = false;
+ return err;
+ }
+ *buffer_out = aBuffer;
+ }
+
+ return OK;
+}
+
+status_t VideoEditorSRC::getNextBuffer(AudioBufferProvider::Buffer *pBuffer) {
+ LOGV("Requesting %d", pBuffer->frameCount);
+ uint32_t availableFrames;
+ bool lastBuffer = false;
+ MediaBuffer *aBuffer;
+
+
+ //update the internal buffer
+ // Store the leftover at the beginning of the local buffer
+ if (mLeftover > 0) {
+ LOGV("Moving mLeftover =%d from %d", mLeftover, mLastReadSize);
+ if (mLastReadSize > 0) {
+ memcpy(pInterframeBuffer, (uint8_t*) (pInterframeBuffer + mLastReadSize), mLeftover);
+ }
+ mInterframeBufferPosition = mLeftover;
+ }
+ else {
+ mInterframeBufferPosition = 0;
+ }
+
+ availableFrames = mInterframeBufferPosition / (mChannelCnt*2);
+
+ while ((availableFrames < pBuffer->frameCount)&&(mStarted)) {
+ // if we seek, reset the initial time stamp and accumulated time
+#ifndef FROYO
+ ReadOptions options;
+ if (mSeekTimeUs >= 0) {
+ LOGV("%p cacheMore_l Seek requested = %lld", this, mSeekTimeUs);
+ ReadOptions::SeekMode mode = mSeekMode;
+ options.setSeekTo(mSeekTimeUs, mode);
+ mSeekTimeUs = -1;
+ }
+#else
+ ReadOptions options;
+ if (mSeekTimeUs >= 0) {
+ LOGV("%p cacheMore_l Seek requested = %lld", this, mSeekTimeUs);
+ options = mReadOptions;
+ mSeekTimeUs = -1;
+ }
+#endif
+ /* The first call to read() will require to buffer twice as much data */
+ /* This will be needed by the resampler */
+ status_t err = mSource->read(&aBuffer, &options);
+ LOGV("mSource->read returned %d", err);
+ if(err != OK) {
+ if (mInterframeBufferPosition == 0) {
+ mStarted = false;
+ }
+ //Empty the internal buffer if there is no more data left in the source
+ else {
+ lastBuffer = true;
+ mInputByteBuffer = pInterframeBuffer;
+ //clear the end of the buffer, just in case
+ memset(pInterframeBuffer+mInterframeBufferPosition, 0x00, DEFAULT_SAMPLING_FREQ * 2 * 2 - mInterframeBufferPosition);
+ mStarted = false;
+ }
+ }
+ else {
+ //copy the buffer
+ memcpy((uint8_t*) (pInterframeBuffer + mInterframeBufferPosition), (uint8_t*) (aBuffer->data() + aBuffer->range_offset()), aBuffer->range_length());
+ LOGV("Read from buffer %d", aBuffer->range_length());
+
+ mInterframeBufferPosition += aBuffer->range_length();
+ LOGV("Stored %d", mInterframeBufferPosition);
+
+ // Get the time stamp of the first buffer
+ if (mInitialTimeStampUs == -1) {
+ int64_t curTS;
+ sp<MetaData> from = aBuffer->meta_data();
+ from->findInt64(kKeyTime, &curTS);
+ LOGV("setting mInitialTimeStampUs to %lld", mInitialTimeStampUs);
+ mInitialTimeStampUs = curTS;
+ }
+
+ // release the buffer
+ aBuffer->release();
+ }
+ availableFrames = mInterframeBufferPosition / (mChannelCnt*2);
+ LOGV("availableFrames %d", availableFrames);
+ }
+
+ if (lastBuffer) {
+ pBuffer->frameCount = availableFrames;
+ }
+
+ //update the input buffer
+ pBuffer->raw = (void*)(pInterframeBuffer);
+
+ // Update how many bytes are left
+ // (actualReadSize is updated in getNextBuffer() called from resample())
+ int32_t actualReadSize = pBuffer->frameCount * mChannelCnt * 2;
+ mLeftover = mInterframeBufferPosition - actualReadSize;
+ LOGV("mLeftover %d", mLeftover);
+
+ mLastReadSize = actualReadSize;
+
+ //+ debug
+ //pBuffer->frameCount = 1024;
+ fwrite(pBuffer->raw, 1, pBuffer->frameCount * mChannelCnt * sizeof(int16_t), fp);
+
+ LOGV("inFrameCount %d", pBuffer->frameCount);
+
+ return OK;
+}
+
+
+void VideoEditorSRC::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
+ if(pBuffer->raw != NULL) {
+ pBuffer->raw = NULL;
+ }
+ pBuffer->frameCount = 0;
+}
+
+} //namespce android