Merge "Initial check-in for AACWriter"
This commit is contained in:
75
include/media/stagefright/AACWriter.h
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75
include/media/stagefright/AACWriter.h
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@@ -0,0 +1,75 @@
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/*
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* Copyright (C) 2011 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef AAC_WRITER_H_
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#define AAC_WRITER_H_
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#include <media/stagefright/MediaWriter.h>
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#include <utils/threads.h>
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namespace android {
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struct MediaSource;
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struct MetaData;
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struct AACWriter : public MediaWriter {
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AACWriter(const char *filename);
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AACWriter(int fd);
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status_t initCheck() const;
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virtual status_t addSource(const sp<MediaSource> &source);
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virtual bool reachedEOS();
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virtual status_t start(MetaData *params = NULL);
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virtual status_t stop();
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virtual status_t pause();
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protected:
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virtual ~AACWriter();
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private:
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enum {
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kAdtsHeaderLength = 7, // # of bytes for the adts header
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kSamplesPerFrame = 1024, // # of samples in a frame
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};
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int mFd;
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status_t mInitCheck;
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sp<MediaSource> mSource;
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bool mStarted;
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volatile bool mPaused;
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volatile bool mResumed;
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volatile bool mDone;
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volatile bool mReachedEOS;
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pthread_t mThread;
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int64_t mEstimatedSizeBytes;
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int64_t mEstimatedDurationUs;
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int32_t mChannelCount;
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int32_t mSampleRate;
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int32_t mFrameDurationUs;
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static void *ThreadWrapper(void *);
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status_t threadFunc();
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bool exceedsFileSizeLimit();
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bool exceedsFileDurationLimit();
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status_t writeAdtsHeader(uint32_t frameLength);
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DISALLOW_EVIL_CONSTRUCTORS(AACWriter);
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};
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} // namespace android
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#endif // AAC_WRITER_H_
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@@ -26,6 +26,7 @@
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#include <media/IMediaPlayerService.h>
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#include <media/stagefright/AudioSource.h>
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#include <media/stagefright/AMRWriter.h>
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#include <media/stagefright/AACWriter.h>
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#include <media/stagefright/CameraSource.h>
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#include <media/stagefright/VideoSourceDownSampler.h>
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#include <media/stagefright/CameraSourceTimeLapse.h>
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@@ -872,15 +873,21 @@ sp<MediaSource> StagefrightRecorder::createAudioSource() {
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}
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status_t StagefrightRecorder::startAACRecording() {
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CHECK(mOutputFormat == OUTPUT_FORMAT_AAC_ADIF ||
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mOutputFormat == OUTPUT_FORMAT_AAC_ADTS);
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// FIXME:
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// Add support for OUTPUT_FORMAT_AAC_ADIF
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CHECK(mOutputFormat == OUTPUT_FORMAT_AAC_ADTS);
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CHECK(mAudioEncoder == AUDIO_ENCODER_AAC);
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CHECK(mAudioSource != AUDIO_SOURCE_CNT);
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CHECK(0 == "AACWriter is not implemented yet");
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mWriter = new AACWriter(mOutputFd);
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status_t status = startRawAudioRecording();
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if (status != OK) {
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mWriter.clear();
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mWriter = NULL;
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}
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return OK;
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return status;
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}
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status_t StagefrightRecorder::startAMRRecording() {
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@@ -902,6 +909,16 @@ status_t StagefrightRecorder::startAMRRecording() {
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}
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}
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mWriter = new AMRWriter(mOutputFd);
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status_t status = startRawAudioRecording();
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if (status != OK) {
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mWriter.clear();
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mWriter = NULL;
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}
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return status;
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}
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status_t StagefrightRecorder::startRawAudioRecording() {
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if (mAudioSource >= AUDIO_SOURCE_CNT) {
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LOGE("Invalid audio source: %d", mAudioSource);
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return BAD_VALUE;
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@@ -917,7 +934,7 @@ status_t StagefrightRecorder::startAMRRecording() {
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return UNKNOWN_ERROR;
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}
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mWriter = new AMRWriter(mOutputFd);
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CHECK(mWriter != 0);
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mWriter->addSource(audioEncoder);
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if (mMaxFileDurationUs != 0) {
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@@ -69,6 +69,7 @@ private:
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sp<Surface> mPreviewSurface;
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sp<IMediaRecorderClient> mListener;
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sp<MediaWriter> mWriter, mWriterAux;
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int mOutputFd, mOutputFdAux;
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sp<AudioSource> mAudioSourceNode;
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audio_source_t mAudioSource;
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@@ -104,7 +105,6 @@ private:
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sp<CameraSourceTimeLapse> mCameraSourceTimeLapse;
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String8 mParams;
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int mOutputFd, mOutputFdAux;
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bool mIsMetaDataStoredInVideoBuffers;
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MediaProfiles *mEncoderProfiles;
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@@ -123,6 +123,7 @@ private:
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status_t startMPEG4Recording();
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status_t startAMRRecording();
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status_t startAACRecording();
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status_t startRawAudioRecording();
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status_t startRTPRecording();
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status_t startMPEG2TSRecording();
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sp<MediaSource> createAudioSource();
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382
media/libstagefright/AACWriter.cpp
Normal file
382
media/libstagefright/AACWriter.cpp
Normal file
@@ -0,0 +1,382 @@
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/*
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* Copyright (C) 2011 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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//#define LOG_NDEBUG 0
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#define LOG_TAG "AACWriter"
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#include <utils/Log.h>
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#include <media/stagefright/AACWriter.h>
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#include <media/stagefright/MediaBuffer.h>
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#include <media/stagefright/foundation/ADebug.h>
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#include <media/stagefright/MediaDefs.h>
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#include <media/stagefright/MediaErrors.h>
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#include <media/stagefright/MediaSource.h>
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#include <media/stagefright/MetaData.h>
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#include <media/mediarecorder.h>
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#include <sys/prctl.h>
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#include <sys/resource.h>
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#include <fcntl.h>
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namespace android {
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AACWriter::AACWriter(const char *filename)
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: mFd(-1),
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mInitCheck(NO_INIT),
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mStarted(false),
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mPaused(false),
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mResumed(false),
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mChannelCount(-1),
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mSampleRate(-1) {
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LOGV("AACWriter Constructor");
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mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR);
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if (mFd >= 0) {
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mInitCheck = OK;
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}
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}
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AACWriter::AACWriter(int fd)
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: mFd(dup(fd)),
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mInitCheck(mFd < 0? NO_INIT: OK),
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mStarted(false),
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mPaused(false),
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mResumed(false),
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mChannelCount(-1),
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mSampleRate(-1) {
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}
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AACWriter::~AACWriter() {
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if (mStarted) {
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stop();
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}
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if (mFd != -1) {
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close(mFd);
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mFd = -1;
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}
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}
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status_t AACWriter::initCheck() const {
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return mInitCheck;
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}
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static int writeInt8(int fd, uint8_t x) {
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return ::write(fd, &x, 1);
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}
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status_t AACWriter::addSource(const sp<MediaSource> &source) {
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if (mInitCheck != OK) {
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return mInitCheck;
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}
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if (mSource != NULL) {
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LOGE("AAC files only support a single track of audio.");
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return UNKNOWN_ERROR;
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}
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sp<MetaData> meta = source->getFormat();
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const char *mime;
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CHECK(meta->findCString(kKeyMIMEType, &mime));
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CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC));
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CHECK(meta->findInt32(kKeyChannelCount, &mChannelCount));
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CHECK(meta->findInt32(kKeySampleRate, &mSampleRate));
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CHECK(mChannelCount >= 1 && mChannelCount <= 2);
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mSource = source;
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return OK;
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}
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status_t AACWriter::start(MetaData *params) {
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if (mInitCheck != OK) {
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return mInitCheck;
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}
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if (mSource == NULL) {
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return UNKNOWN_ERROR;
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}
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if (mStarted && mPaused) {
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mPaused = false;
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mResumed = true;
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return OK;
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} else if (mStarted) {
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// Already started, does nothing
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return OK;
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}
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mFrameDurationUs = (kSamplesPerFrame * 1000000LL + (mSampleRate >> 1))
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/ mSampleRate;
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status_t err = mSource->start();
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if (err != OK) {
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return err;
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}
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pthread_attr_t attr;
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pthread_attr_init(&attr);
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pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
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mReachedEOS = false;
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mDone = false;
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pthread_create(&mThread, &attr, ThreadWrapper, this);
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pthread_attr_destroy(&attr);
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mStarted = true;
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return OK;
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}
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status_t AACWriter::pause() {
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if (!mStarted) {
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return OK;
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}
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mPaused = true;
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return OK;
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}
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status_t AACWriter::stop() {
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if (!mStarted) {
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return OK;
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}
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mDone = true;
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void *dummy;
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pthread_join(mThread, &dummy);
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status_t err = (status_t) dummy;
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{
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status_t status = mSource->stop();
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if (err == OK &&
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(status != OK && status != ERROR_END_OF_STREAM)) {
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err = status;
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}
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}
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mStarted = false;
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return err;
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}
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bool AACWriter::exceedsFileSizeLimit() {
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if (mMaxFileSizeLimitBytes == 0) {
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return false;
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}
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return mEstimatedSizeBytes >= mMaxFileSizeLimitBytes;
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}
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bool AACWriter::exceedsFileDurationLimit() {
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if (mMaxFileDurationLimitUs == 0) {
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return false;
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}
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return mEstimatedDurationUs >= mMaxFileDurationLimitUs;
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}
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// static
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void *AACWriter::ThreadWrapper(void *me) {
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return (void *) static_cast<AACWriter *>(me)->threadFunc();
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}
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/*
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* Returns an index into the sample rate table if the
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* given sample rate is found; otherwise, returns -1.
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*/
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static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) {
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static const int kSampleRateTable[] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000
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};
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const int tableSize =
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sizeof(kSampleRateTable) / sizeof(kSampleRateTable[0]);
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*tableIndex = 0;
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for (int index = 0; index < tableSize; ++index) {
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if (sampleRate == kSampleRateTable[index]) {
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LOGV("Sample rate: %d and index: %d",
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sampleRate, index);
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*tableIndex = index;
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return true;
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}
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}
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LOGE("Sampling rate %d bps is not supported", sampleRate);
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return false;
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}
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/*
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* ADTS (Audio data transport stream) header structure.
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* It consists of 7 or 9 bytes (with or without CRC):
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* 12 bits of syncword 0xFFF, all bits must be 1
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* 1 bit of field ID. 0 for MPEG-4, and 1 for MPEG-2
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* 2 bits of MPEG layer. If in MPEG-TS, set to 0
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* 1 bit of protection absense. Set to 1 if no CRC.
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* 2 bits of profile code. Set to 1 (The MPEG-4 Audio
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* object type minus 1. We are using AAC-LC = 2)
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* 4 bits of sampling frequency index code (15 is not allowed)
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* 1 bit of private stream. Set to 0.
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* 3 bits of channel configuration code. 0 resevered for inband PCM
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* 1 bit of originality. Set to 0.
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* 1 bit of home. Set to 0.
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* 1 bit of copyrighted steam. Set to 0.
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* 1 bit of copyright start. Set to 0.
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* 13 bits of frame length. It included 7 ot 9 bytes header length.
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* it is set to (protection absense? 7: 9) + size(AAC frame)
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* 11 bits of buffer fullness. 0x7FF for VBR.
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* 2 bits of frames count in one packet. Set to 0.
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*/
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status_t AACWriter::writeAdtsHeader(uint32_t frameLength) {
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uint8_t data = 0xFF;
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write(mFd, &data, 1);
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const uint8_t kFieldId = 0;
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const uint8_t kMpegLayer = 0;
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const uint8_t kProtectionAbsense = 1; // 1: kAdtsHeaderLength = 7
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data = 0xF0;
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data |= (kFieldId << 3);
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data |= (kMpegLayer << 1);
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data |= kProtectionAbsense;
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write(mFd, &data, 1);
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|
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const uint8_t kProfileCode = 1; // AAC-LC
|
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uint8_t kSampleFreqIndex;
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CHECK(getSampleRateTableIndex(mSampleRate, &kSampleFreqIndex));
|
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const uint8_t kPrivateStream = 0;
|
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const uint8_t kChannelConfigCode = mChannelCount;
|
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data = (kProfileCode << 6);
|
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data |= (kSampleFreqIndex << 2);
|
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data |= (kPrivateStream << 1);
|
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data |= (kChannelConfigCode >> 2);
|
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write(mFd, &data, 1);
|
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|
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// 4 bits from originality to copyright start
|
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const uint8_t kCopyright = 0;
|
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const uint32_t kFrameLength = frameLength;
|
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data = ((kChannelConfigCode & 3) << 6);
|
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data |= (kCopyright << 2);
|
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data |= ((kFrameLength & 0x1800) >> 11);
|
||||
write(mFd, &data, 1);
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|
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data = ((kFrameLength & 0x07F8) >> 3);
|
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write(mFd, &data, 1);
|
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|
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const uint32_t kBufferFullness = 0x7FF; // VBR
|
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data = ((kFrameLength & 0x07) << 5);
|
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data |= ((kBufferFullness & 0x07C0) >> 6);
|
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write(mFd, &data, 1);
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||||
|
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const uint8_t kFrameCount = 0;
|
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data = ((kBufferFullness & 0x03F) << 2);
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data |= kFrameCount;
|
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write(mFd, &data, 1);
|
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|
||||
return OK;
|
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}
|
||||
|
||||
status_t AACWriter::threadFunc() {
|
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mEstimatedDurationUs = 0;
|
||||
mEstimatedSizeBytes = 0;
|
||||
int64_t previousPausedDurationUs = 0;
|
||||
int64_t maxTimestampUs = 0;
|
||||
status_t err = OK;
|
||||
|
||||
prctl(PR_SET_NAME, (unsigned long)"AACWriterThread", 0, 0, 0);
|
||||
|
||||
while (!mDone && err == OK) {
|
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MediaBuffer *buffer;
|
||||
err = mSource->read(&buffer);
|
||||
|
||||
if (err != OK) {
|
||||
break;
|
||||
}
|
||||
|
||||
if (mPaused) {
|
||||
buffer->release();
|
||||
buffer = NULL;
|
||||
continue;
|
||||
}
|
||||
|
||||
mEstimatedSizeBytes += kAdtsHeaderLength + buffer->range_length();
|
||||
if (exceedsFileSizeLimit()) {
|
||||
buffer->release();
|
||||
buffer = NULL;
|
||||
notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_FILESIZE_REACHED, 0);
|
||||
break;
|
||||
}
|
||||
|
||||
int32_t isCodecSpecific = 0;
|
||||
if (buffer->meta_data()->findInt32(kKeyIsCodecConfig, &isCodecSpecific) && isCodecSpecific) {
|
||||
LOGV("Drop codec specific info buffer");
|
||||
buffer->release();
|
||||
buffer = NULL;
|
||||
continue;
|
||||
}
|
||||
|
||||
int64_t timestampUs;
|
||||
CHECK(buffer->meta_data()->findInt64(kKeyTime, ×tampUs));
|
||||
if (timestampUs > mEstimatedDurationUs) {
|
||||
mEstimatedDurationUs = timestampUs;
|
||||
}
|
||||
if (mResumed) {
|
||||
previousPausedDurationUs += (timestampUs - maxTimestampUs - mFrameDurationUs);
|
||||
mResumed = false;
|
||||
}
|
||||
timestampUs -= previousPausedDurationUs;
|
||||
LOGV("time stamp: %lld, previous paused duration: %lld",
|
||||
timestampUs, previousPausedDurationUs);
|
||||
if (timestampUs > maxTimestampUs) {
|
||||
maxTimestampUs = timestampUs;
|
||||
}
|
||||
|
||||
if (exceedsFileDurationLimit()) {
|
||||
buffer->release();
|
||||
buffer = NULL;
|
||||
notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_DURATION_REACHED, 0);
|
||||
break;
|
||||
}
|
||||
|
||||
// Each output AAC audio frame to the file contains
|
||||
// 1. an ADTS header, followed by
|
||||
// 2. the compressed audio data.
|
||||
ssize_t dataLength = buffer->range_length();
|
||||
uint8_t *data = (uint8_t *)buffer->data() + buffer->range_offset();
|
||||
if (writeAdtsHeader(kAdtsHeaderLength + dataLength) != OK ||
|
||||
dataLength != write(mFd, data, dataLength)) {
|
||||
err = ERROR_IO;
|
||||
}
|
||||
|
||||
buffer->release();
|
||||
buffer = NULL;
|
||||
}
|
||||
|
||||
close(mFd);
|
||||
mFd = -1;
|
||||
mReachedEOS = true;
|
||||
if (err == ERROR_END_OF_STREAM) {
|
||||
return OK;
|
||||
}
|
||||
return err;
|
||||
}
|
||||
|
||||
bool AACWriter::reachedEOS() {
|
||||
return mReachedEOS;
|
||||
}
|
||||
|
||||
} // namespace android
|
||||
@@ -6,6 +6,7 @@ include frameworks/base/media/libstagefright/codecs/common/Config.mk
|
||||
LOCAL_SRC_FILES:= \
|
||||
ACodec.cpp \
|
||||
AACExtractor.cpp \
|
||||
AACWriter.cpp \
|
||||
AMRExtractor.cpp \
|
||||
AMRWriter.cpp \
|
||||
AVIExtractor.cpp \
|
||||
|
||||
@@ -84,7 +84,7 @@ status_t AACEncoder::initCheck() {
|
||||
params.sampleRate = mSampleRate;
|
||||
params.bitRate = mBitRate;
|
||||
params.nChannels = mChannels;
|
||||
params.adtsUsed = 0; // For MP4 file, don't use adts format$
|
||||
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)) {
|
||||
LOGE("Failed to set AAC encoder parameters");
|
||||
return UNKNOWN_ERROR;
|
||||
|
||||
Reference in New Issue
Block a user