Decoding raw AAC with MediaCodec without using MediaExtractor

Question

I successfully decoded and play the mp4 (AAC) file using MediaExtractor and MediaCodec with the code below.I want to decode raw AAC (in another file, with same encoding format) to PCM. The problem is that I don't know how to set SampleSize and presentationTimeUs without mediaExtractor. How can I set above parameters without using MediaExtractor?

//songwav.mp4 file is created from PCM with this format

MediaFormat outputFormat = MediaFormat.createAudioFormat(
            "audio/mp4a-latm", 44100, 2);
    outputFormat.setInteger(MediaFormat.KEY_AAC_PROFILE,
            MediaCodecInfo.CodecProfileLevel.AACObjectLC);
    outputFormat.setInteger(MediaFormat.KEY_BIT_RATE,
            128000);

//decoding

String inputfilePath = Environment.getExternalStorageDirectory()
            .getPath() + "/" + "songwav.mp4";
String outputFilePath = Environment.getExternalStorageDirectory()
            .getPath() + "/" + "songwavmp4.pcm";
OutputStream outputStream = new FileOutputStream(outputFilePath);
    MediaCodec codec;
    AudioTrack audioTrack;

// extractor gets information about the stream
MediaExtractor extractor = new MediaExtractor();
extractor.setDataSource(inputfilePath);

        MediaFormat format = extractor.getTrackFormat(0);
    String mime = format.getString(MediaFormat.KEY_MIME);

    // the actual decoder
    codec = MediaCodec.createDecoderByType(mime);
    codec.configure(format, null /* surface */, null /* crypto */, 0 /* flags */);
    codec.start();
    ByteBuffer[] codecInputBuffers = codec.getInputBuffers();
    ByteBuffer[] codecOutputBuffers = codec.getOutputBuffers();

    // get the sample rate to configure AudioTrack
    int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);

    // create our AudioTrack instance
    audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
            AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT,
            AudioTrack.getMinBufferSize(sampleRate,
                    AudioFormat.CHANNEL_OUT_STEREO,
                    AudioFormat.ENCODING_PCM_16BIT), AudioTrack.MODE_STREAM);

    // start playing, we will feed you later
    audioTrack.play();
    extractor.selectTrack(0);

    // start decoding
    final long kTimeOutUs = 10000;
    MediaCodec.BufferInfo BufInfo = new MediaCodec.BufferInfo();
    boolean sawInputEOS = false;
    boolean sawOutputEOS = false;

    int inputBufIndex;

    int counter=0;
while (!sawOutputEOS) {


        counter++;
        if (!sawInputEOS) {
            inputBufIndex = codec.dequeueInputBuffer(kTimeOutUs);
            // Log.d(LOG_TAG, " bufIndexCheck " + bufIndexCheck);
            if (inputBufIndex >= 0) {
                ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];

                int sampleSize = extractor
                        .readSampleData(dstBuf, 0 /* offset */);

                long presentationTimeUs = 0;

                if (sampleSize < 0) {

                    sawInputEOS = true;
                    sampleSize = 0;
                } else {

                    presentationTimeUs = extractor.getSampleTime();
                }
                // can throw illegal state exception (???)

                codec.queueInputBuffer(inputBufIndex, 0 /* offset */,
                        sampleSize, presentationTimeUs,
                        sawInputEOS ? MediaCodec.BUFFER_FLAG_END_OF_STREAM
                                : 0);

                if (!sawInputEOS) {
                    extractor.advance();
                }
            } else {
                Log.e("sohail", "inputBufIndex " + inputBufIndex);
            }
        }

        int res = codec.dequeueOutputBuffer(BufInfo, kTimeOutUs);

        if (res >= 0) {
            Log.i("sohail","decoding: deqOutputBuffer >=0, counter="+counter);
            // Log.d(LOG_TAG, "got frame, size " + info.size + "/" +
            // info.presentationTimeUs);
            if (BufInfo.size > 0) {
                // noOutputCounter = 0;
            }

            int outputBufIndex = res;
            ByteBuffer buf = codecOutputBuffers[outputBufIndex];

            final byte[] chunk = new byte[BufInfo.size];
            buf.get(chunk);
            buf.clear();

            if (chunk.length > 0) {
                // play
                audioTrack.write(chunk, 0, chunk.length);
                // write to file
                outputStream.write(chunk);

            }
            codec.releaseOutputBuffer(outputBufIndex, false /* render */);
            if ((BufInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
                Log.i("sohail", "saw output EOS.");
                sawOutputEOS = true;
            }
        } else if (res == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
            codecOutputBuffers = codec.getOutputBuffers();

            Log.i("sohail", "output buffers have changed.");
        } else if (res == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
            MediaFormat oformat = codec.getOutputFormat();

            Log.i("sohail", "output format has changed to " + oformat);
        } else {
            Log.i("sohail", "dequeueOutputBuffer returned " + res);
        }
    }

    Log.d(LOG_TAG, "stopping...");

    // ////////closing
    if (audioTrack != null) {
        audioTrack.flush();
        audioTrack.release();
        audioTrack = null;
    }

    outputStream.flush();
    outputStream.close();

    codec.stop();