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PODecode AAC to PCM with ffmpeg on android
 

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  1. PODecode AAC to PCM with ffmpeg on android
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    <p>I have built ffmpeg 0.8.12 (love) with the android NDK (r8c) on ubuntu. I then use the generated library in another android application through JNI.</p> <p>Essentially what I want to do is pass a byte stream from java to my c jni function and use ffmpeg to decode it into a PCM audio buffer which will then be passed back to java to be played using Android's AudioTrack. I can successfully pass the buffer through to jni (have checked the values) and ffmpeg seems to initialise correctly, but when it tries to decode the first frame, it throws an error in the aac_decode_frame_int method in aacdec.c "channel element 0.0 is not allocated". The aac file plays fine and is valid.</p> <p>Here is my jni code to do the decoding</p> <pre><code>jint Java_com_example_testffmpeg_MainActivity_decodeAacBytes(JNIEnv * env, jobject this, jbyteArray input, jint numBytes) { //copy bytes from java jbyte* bufferPtr = (*env)-&gt;GetByteArrayElements(env, input, NULL); uint8_t inputBytes[numBytes + FF_INPUT_BUFFER_PADDING_SIZE]; memset(inputBytes, 0, numBytes + FF_INPUT_BUFFER_PADDING_SIZE); memcpy(inputBytes, bufferPtr, numBytes); (*env)-&gt;ReleaseByteArrayElements(env, input, bufferPtr, 0); av_register_all(); AVCodec *codec = avcodec_find_decoder(CODEC_ID_AAC); if (codec == NULL) { LOGE("Cant find AAC codec\n"); return 0; } LOGI("AAC codec found\n"); AVCodecContext *avCtx = avcodec_alloc_context(); if (avCtx == NULL) { LOGE("Could not allocate codec context\n"); return 0; } LOGI("codec context allocated\n"); if (avcodec_open2(avCtx, codec, NULL) &lt; 0) { LOGE("Could not open codec\n"); return 0; } LOGI("AAC codec opened"); //the input buffer AVPacket avPacket; av_init_packet(&amp;avPacket); LOGI("AVPacket initialised\n"); avPacket.size = numBytes; //input buffer size avPacket.data = inputBytes; // the input buffer int outSize; int len; uint8_t *outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE); while (avPacket.size &gt; 0) { outSize = AVCODEC_MAX_AUDIO_FRAME_SIZE; len = avcodec_decode_audio3(avCtx, (short *) outbuf, &amp;outSize, &amp;avPacket); if (len &lt; 0) { LOGE("Error while decoding\n"); return 0; } if (outSize &gt; 0) { LOGI("Decoded some stuff\n"); } avPacket.size -= len; avPacket.data += len; } LOGI("Freeing memory\n"); av_free_packet(&amp;avPacket); avcodec_close(avCtx); av_free(avCtx); return 0; } </code></pre> <p>The problem occurs in the call to avcodec_decode_audio3, when the decoding first occurs. I have stepped through the ffmpeg code, but can't find the problem. Any help would be greatly appreciated!</p>
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    1. COHave you tried to decode the file on the host machine? Is it decoded by ffmpeg? What about SoX, it is capable on doing that, also normalize, fade-in/out, change volume, etc?
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      1. USrraallvv
    2. COI have not tried decoding the file by using ffmpeg in the console - I am trying to decode from within my application code, so this wouldn't help me anyway. The file does play correctly in any player on my phone, so it is not corrupt.
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    3. COI mean to compile SoX with NDK, please see [this link](http://stackoverflow.com/questions/8582765/android-ndk-linking-problems).
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