/**
* Initialize our main context FFAudioIO, so that SwrContext, decode buffers and the encoder are set
* to reasonable values.
*
* @param JNIEnv env
* @param aio our context, FFAudioIO
* @return a negative value, if something went wrong
*/
int ff_init_audioio(JNIEnv *env, FFAudioIO *aio) {
int res = 0;
int nb_planes;
AVCodec *codec = NULL;
aio->timestamp = 0;
// allocate pointer to the audio buffers, i.e. the multiple planes/channels.
nb_planes = av_sample_fmt_is_planar(aio->stream->codecpar->format)
? aio->stream->codecpar->channels
: 1;
// always init SWR to keep code simpler
res = init_swr(env, aio);
if (res < 0) {
// exception is already thrown
goto bail;
}
// if for some reason the codec delivers 24bit, we need to encode its output to little endian
if (aio->stream->codecpar->bits_per_coded_sample == 24) {
codec = ff_find_encoder(aio->stream->codecpar->format, aio->stream->codecpar->bits_per_coded_sample, ff_big_endian(aio->stream->codecpar->codec_id), 1);
if (!codec) {
res = AVERROR(EINVAL);
throwIOExceptionIfError(env, res, "Could not find suitable encoder codec.");
goto bail;
}
res = ff_init_encoder(env, aio, codec);
if (res<0) {
throwIOExceptionIfError(env, res, "Could not initialize encoder codec.");
goto bail;
}
}
// allocate the buffer the codec decodes to
aio->audio_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
if (!aio->audio_data) {
res = AVERROR(ENOMEM);
throwIOExceptionIfError(env, res, "Could not allocate audio data buffers.");
goto bail;
}
aio->decode_frame = av_frame_alloc();
if (!aio->decode_frame) {
res = AVERROR(ENOMEM);
throwIOExceptionIfError(env, res, "Could not allocate frame.");
goto bail;
}
// initialize packet
av_init_packet(&(aio->decode_packet));
aio->decode_packet.data = NULL;
aio->decode_packet.size = 0;
bail:
return res;
}
static int create_ffaudiofileformats(JNIEnv *env, AVFormatContext *format_context, jobjectArray *array, jstring url) {
int res = 0;
jlong duration_in_microseconds = -1;
jfloat frame_rate = -1;
jobject vbr = NULL;
jboolean big_endian = 1;
jobject audio_format = NULL;
jint frame_size = -1;
jint sample_size = -1;
int audio_stream_count = 0;
int audio_stream_number = 0;
// count possible audio streams
int i;
for (i=0; i<format_context->nb_streams; i++) {
AVStream* stream = format_context->streams[i];
if (stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
audio_stream_count++;
}
}
#ifdef DEBUG
fprintf(stderr, "Found %i audio streams.\n", audio_stream_count);
#endif
// are there any audio streams at all?
if (audio_stream_count == 0) {
throwUnsupportedAudioFileExceptionIfError(env, -1, "Failed to find audio stream");
goto bail;
}
// create output array
*array = (*env)->NewObjectArray(env, audio_stream_count, (*env)->FindClass(env, "javax/sound/sampled/AudioFileFormat"), NULL);
if (array == NULL) {
goto bail;
}
#ifdef DEBUG
fprintf(stderr, "Created audio file format array.\n");
#endif
// iterate over audio streams
for (i=0; i<format_context->nb_streams; i++) {
AVStream* stream = format_context->streams[i];
if (stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
/*
res = ff_open_stream(env, stream);
if (res) {
goto bail;
}
*/
// create object
duration_in_microseconds = duration(format_context, stream);
frame_rate = get_frame_rate(stream, duration_in_microseconds);
big_endian = ff_big_endian(stream->codecpar->codec_id);
if (is_pcm(stream->codecpar->codec_id)) {
frame_size = (stream->codecpar->bits_per_coded_sample / 8) * stream->codecpar->channels;
}
// TODO: Support VBR.
sample_size = stream->codecpar->bits_per_raw_sample
? stream->codecpar->bits_per_raw_sample
: stream->codecpar->bits_per_coded_sample;
#ifdef DEBUG
fprintf(stderr, "stream->codecpar->bits_per_coded_sample: %i\n", stream->codecpar->bits_per_coded_sample);
fprintf(stderr, "stream->codecpar->bits_per_raw_sample : %i\n", stream->codecpar->bits_per_raw_sample);
fprintf(stderr, "stream->codecpar->bit_rate : %i\n", stream->codecpar->bit_rate);
fprintf(stderr, "format_context->packet_size : %i\n", format_context->packet_size);
fprintf(stderr, "frames : %" PRId64 "\n", stream->nb_frames);
fprintf(stderr, "sample_rate: %i\n", stream->codecpar->sample_rate);
fprintf(stderr, "sampleSize : %i\n", stream->codecpar->bits_per_coded_sample);
fprintf(stderr, "channels : %i\n", stream->codecpar->channels);
fprintf(stderr, "frame_size : %i\n", (int)frame_size);
fprintf(stderr, "codec_id : %i\n", stream->codecpar->codec_id);
fprintf(stderr, "duration : %" PRId64 "\n", (int64_t)duration_in_microseconds);
fprintf(stderr, "frame_rate : %f\n", frame_rate);
if (big_endian) {
fprintf(stderr, "big_endian : true\n");
} else {
fprintf(stderr, "big_endian : false\n");
}
#endif
audio_format = create_ffaudiofileformat(env, url,
stream->codecpar->codec_id,
(jfloat)stream->codecpar->sample_rate,
sample_size,
stream->codecpar->channels,
frame_size,
frame_rate,
big_endian,
duration_in_microseconds,
stream->codecpar->bit_rate,
vbr);
(*env)->SetObjectArrayElement(env, *array, audio_stream_number, audio_format);
audio_stream_number++;
// clean up
/*
if (stream && stream->codec) {
avcodec_close(stream->codec);
}
*/
}
}
bail:
return res;
}