use ffmpeg api to convert audio files. crash on avcodec_encode_audio2

use ffmpeg api to convert audio files. crash on avcodec_encode_audio2

From the examples I got the basic idea of this code. However I am not
sure, what I am missing, as muxing.c demuxing.c and decoding_encoding.c
all use different approaches.
The process of converting an audio file to another file should go roughly
like this: inputfile -demux-> audiostream -read-> inPackets
-decode2frames-> frames -encode2packets-> outPackets -write-> audiostream
-mux-> outputfile
However I found the following comment in demuxing.c: /* Write the raw
audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
My questions about this are:
Can I expect a frame that was retrieved by calling one of the decoder
functions, f.e. avcodec_decode_audio4 to hold suitable values to directly
put it into an encoder or is the resampling step mentioned in the comment
mandatory?
Am I taking the right approach? ffmpeg is very asymmetric, i.e. if there
is a function open_file_for_input there might not be a function
open_file_for_output. Also there are different versions of many functions
(avcodec_decode_audio[1-4]) and different naming schemes, so it's very
hard to tell, if the general approach is right, or actually an ugly
mixture of techniques that where used at different version bumps of
ffmpeg.
ffmpeg uses a lot of specific terms, like 'planar sampling' or 'packed
format' and I am having a hard time, finding definitions for these terms.
Is it possible to write working code, without deep knowledge of audio?
Here is my code so far that right now crashes at avcodec_encode_audio2 and
I don't know why.
int Java_com_fscz_ffmpeg_Audio_convert(JNIEnv * env, jobject this, jstring
jformat, jstring jcodec, jstring jsource, jstring jdest) {
jboolean isCopy;
jclass configClass = (*env)->FindClass(env, "com.fscz.ffmpeg.Config");
jfieldID fid = (*env)->GetStaticFieldID(env, configClass,
"ffmpeg_logging", "I");
logging = (*env)->GetStaticIntField(env, configClass, fid);
/// open input
const char* sourceFile = (*env)->GetStringUTFChars(env, jsource, &isCopy);
AVFormatContext* pInputCtx;
AVStream* pInputStream;
open_input(sourceFile, &pInputCtx, &pInputStream);
// open output
const char* destFile = (*env)->GetStringUTFChars(env, jdest, &isCopy);
const char* cformat = (*env)->GetStringUTFChars(env, jformat, &isCopy);
const char* ccodec = (*env)->GetStringUTFChars(env, jcodec, &isCopy);
AVFormatContext* pOutputCtx;
AVOutputFormat* pOutputFmt;
AVStream* pOutputStream;
open_output(cformat, ccodec, destFile, &pOutputCtx, &pOutputFmt,
&pOutputStream);
/// decode/encode
error = avformat_write_header(pOutputCtx, NULL);
DIE_IF_LESS_ZERO(error, "error writing output stream header to file: %s,
error: %s"
, destFile, e2s(error));
AVFrame* frame = avcodec_alloc_frame();
DIE_IF_UNDEFINED(frame, "Could not allocate audio frame");
frame->pts = 0;
LOGI("allocate packet");
AVPacket pktIn;
AVPacket pktOut;
LOGI("done");
int got_frame, got_packet, len, frame_count = 0;
int64_t processed_time = 0, duration = pInputStream->duration;
while (av_read_frame(pInputCtx, &pktIn) >= 0) {
do {
len = avcodec_decode_audio4(pInputStream->codec, frame,
&got_frame, &pktIn);
DIE_IF_LESS_ZERO(len, "Error decoding frame: %s", e2s(len));
if (len < 0) break;
len = FFMIN(len, pktIn.size);
size_t unpadded_linesize = frame->nb_samples *
av_get_bytes_per_sample(frame->format);
LOGI("audio_frame n:%d nb_samples:%d pts:%s\n",
frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts,
&(pInputStream->codec->time_base)));
if (got_frame) {
do {
av_init_packet(&pktOut);
pktOut.data = NULL;
pktOut.size = 0;
LOGI("encode frame");
DIE_IF_UNDEFINED(pOutputStream->codec, "no output
codec");
DIE_IF_UNDEFINED(frame->nb_samples, "no nb samples");
DIE_IF_UNDEFINED(pOutputStream->codec->internal, "no
internal");
LOGI("tests done");
len = avcodec_encode_audio2(pOutputStream->codec,
&pktOut, frame, &got_packet);
LOGI("encode done");
DIE_IF_LESS_ZERO(len, "Error (re)encoding frame: %s",
e2s(len));
} while (!got_packet);
// write packet;
LOGI("write packet");
/* Write the compressed frame to the media file. */
error = av_interleaved_write_frame(pOutputCtx, &pktOut);
DIE_IF_LESS_ZERO(error, "Error while writing audio frame:
%s", e2s(error));
av_free_packet(&pktOut);
}
pktIn.data += len;
pktIn.size -= len;
} while (pktIn.size > 0);
av_free_packet(&pktIn);
}
LOGI("write trailer");
av_write_trailer(pOutputCtx);
LOGI("end");
/// close resources
avcodec_free_frame(&frame);
avcodec_close(pInputStream->codec);
av_free(pInputStream->codec);
avcodec_close(pOutputStream->codec);
av_free(pOutputStream->codec);
avformat_close_input(&pInputCtx);
avformat_free_context(pOutputCtx);
return 0;
}