Encode wav to AAC on Android

Question

You can use MediaRecorder to record a stream directly to AAC but there doesn't seem to be a way to encode an existing PCM/WAV file to AAC. The ability to encode to AAC exists natively in Android and I'd like to use that. Is there no way to do it with a pre-existing audio file?

Mythopoeic

Answer 1

Look at this beautiful (and perfectly working) example: Mp4ParserSample

Look at the final part of the class (rows 335-442), the convert Runnable object just does the job! You have to shape that code to your needs, adjust the input and the output file paths and the conversion parameters (sampling rate, bit rate, etc).

public static final String AUDIO_RECORDING_FILE_NAME = "audio_Capturing-190814-034638.422.wav"; // Input PCM file
public static final String COMPRESSED_AUDIO_FILE_NAME = "convertedmp4.m4a"; // Output MP4/M4A file
public static final String COMPRESSED_AUDIO_FILE_MIME_TYPE = "audio/mp4a-latm";
public static final int COMPRESSED_AUDIO_FILE_BIT_RATE = 64000; // 64kbps
public static final int SAMPLING_RATE = 48000;
public static final int BUFFER_SIZE = 48000;
public static final int CODEC_TIMEOUT_IN_MS = 5000;
String LOGTAG = "CONVERT AUDIO";
Runnable convert = new Runnable() {
    @TargetApi(Build.VERSION_CODES.JELLY_BEAN_MR2)
    @Override
    public void run() {
        android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_BACKGROUND);
        try {
            String filePath = Environment.getExternalStorageDirectory().getPath() + "/" + AUDIO_RECORDING_FILE_NAME;
            File inputFile = new File(filePath);
            FileInputStream fis = new FileInputStream(inputFile);

            File outputFile = new File(Environment.getExternalStorageDirectory().getAbsolutePath() + "/" + COMPRESSED_AUDIO_FILE_NAME);
            if (outputFile.exists()) outputFile.delete();

            MediaMuxer mux = new MediaMuxer(outputFile.getAbsolutePath(), MediaMuxer.OutputFormat.MUXER_OUTPUT_MPEG_4);

            MediaFormat outputFormat = MediaFormat.createAudioFormat(COMPRESSED_AUDIO_FILE_MIME_TYPE,SAMPLING_RATE, 1);
            outputFormat.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
            outputFormat.setInteger(MediaFormat.KEY_BIT_RATE, COMPRESSED_AUDIO_FILE_BIT_RATE);
            outputFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, 16384);

            MediaCodec codec = MediaCodec.createEncoderByType(COMPRESSED_AUDIO_FILE_MIME_TYPE);
            codec.configure(outputFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
            codec.start();

            ByteBuffer[] codecInputBuffers = codec.getInputBuffers(); // Note: Array of buffers
            ByteBuffer[] codecOutputBuffers = codec.getOutputBuffers();

            MediaCodec.BufferInfo outBuffInfo = new MediaCodec.BufferInfo();
            byte[] tempBuffer = new byte[BUFFER_SIZE];
            boolean hasMoreData = true;
            double presentationTimeUs = 0;
            int audioTrackIdx = 0;
            int totalBytesRead = 0;
            int percentComplete = 0;
            do {
                int inputBufIndex = 0;
                while (inputBufIndex != -1 && hasMoreData) {
                    inputBufIndex = codec.dequeueInputBuffer(CODEC_TIMEOUT_IN_MS);

                    if (inputBufIndex >= 0) {
                        ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];
                        dstBuf.clear();

                        int bytesRead = fis.read(tempBuffer, 0, dstBuf.limit());
                        Log.e("bytesRead","Readed "+bytesRead);
                        if (bytesRead == -1) { // -1 implies EOS
                            hasMoreData = false;
                            codec.queueInputBuffer(inputBufIndex, 0, 0, (long) presentationTimeUs, MediaCodec.BUFFER_FLAG_END_OF_STREAM);
                        } else {
                            totalBytesRead += bytesRead;
                            dstBuf.put(tempBuffer, 0, bytesRead);
                            codec.queueInputBuffer(inputBufIndex, 0, bytesRead, (long) presentationTimeUs, 0);
                            presentationTimeUs = 1000000l * (totalBytesRead / 2) / SAMPLING_RATE;
                        }
                    }
                }
                // Drain audio
                int outputBufIndex = 0;
                while (outputBufIndex != MediaCodec.INFO_TRY_AGAIN_LATER) {
                    outputBufIndex = codec.dequeueOutputBuffer(outBuffInfo, CODEC_TIMEOUT_IN_MS);
                    if (outputBufIndex >= 0) {
                        ByteBuffer encodedData = codecOutputBuffers[outputBufIndex];
                        encodedData.position(outBuffInfo.offset);
                        encodedData.limit(outBuffInfo.offset + outBuffInfo.size);
                        if ((outBuffInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0 && outBuffInfo.size != 0) {
                            codec.releaseOutputBuffer(outputBufIndex, false);
                        }else{
                            mux.writeSampleData(audioTrackIdx, codecOutputBuffers[outputBufIndex], outBuffInfo);
                            codec.releaseOutputBuffer(outputBufIndex, false);
                        }
                    } else if (outputBufIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
                        outputFormat = codec.getOutputFormat();
                        Log.v(LOGTAG, "Output format changed - " + outputFormat);
                        audioTrackIdx = mux.addTrack(outputFormat);
                        mux.start();
                    } else if (outputBufIndex == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
                        Log.e(LOGTAG, "Output buffers changed during encode!");
                    } else if (outputBufIndex == MediaCodec.INFO_TRY_AGAIN_LATER) {
                        // NO OP
                    } else {
                        Log.e(LOGTAG, "Unknown return code from dequeueOutputBuffer - " + outputBufIndex);
                    }
                }
                percentComplete = (int) Math.round(((float) totalBytesRead / (float) inputFile.length()) * 100.0);
                Log.v(LOGTAG, "Conversion % - " + percentComplete);
            } while (outBuffInfo.flags != MediaCodec.BUFFER_FLAG_END_OF_STREAM);
            fis.close();
            mux.stop();
            mux.release();
            Log.v(LOGTAG, "Compression done ...");
        } catch (FileNotFoundException e) {
            Log.e(LOGTAG, "File not found!", e);
        } catch (IOException e) {
            Log.e(LOGTAG, "IO exception!", e);
        }

        //mStop = false;
        // Notify UI thread...
    }
};

Whelan

This worked well and creates an AAC file I can play on my phone, but when I try to merge the AAC file with an MP4 video using MP4Parser, I get an error java.io.IOException: Expected Start Word 0xfff - Are the headers correct in the output? – Lamia

Why is the MediaFormat.KEY_MAX_INPUT_SIZE set to 16384, while the BUFFER_SIZE is 48000? – Gigantism

Answer 2

you can get your hands dirty with the native code and use the IOMX C++ interface to decoders in the framework. But this is build sensitive and will not work on other phones and android flavours.

Another option is port an opensource aac encoder like ffmpeg and write an app over it over jni. Will atleast work with phones with same architecture (arm-9, cortex a8..).

JB has a MediaCodec just to fulfill your wishes. But the problem would be the install base for devices with JB will be lean for some more time.

http://developer.android.com/about/versions/android-4.1.html#Multimedia

Schoolroom

Answer 3

I think you can use this library.

https://github.com/timsu/android-aac-enc

http://betaful.com/post/82668810035/encoding-aac-audio-in-android

Stravinsky