// Input is a single Atrac3+ packet.
bool SimpleAT3::Decode(void* inbuf, int inbytes, uint8_t *outbuf, int *outbytes) {
#ifdef USE_FFMPEG
AVPacket packet = {0};
av_init_packet(&packet);
packet.data = static_cast<uint8_t *>(inbuf);
packet.size = inbytes;
*outbytes = 0;
int got_frame = 0;
avcodec_get_frame_defaults(frame_);
int len = avcodec_decode_audio4(codecCtx_, frame_, &got_frame, &packet);
if (len < 0) {
ERROR_LOG(ME, "Error decoding Atrac3+ frame");
// TODO: cleanup
return false;
}
if (got_frame) {
int data_size = av_samples_get_buffer_size(
NULL,
codecCtx_->channels,
frame_->nb_samples,
codecCtx_->sample_fmt, 1);
int numSamples = frame_->nb_samples;
int swrRet = swr_convert(swrCtx_, &outbuf, numSamples, (const u8 **)frame_->extended_data, numSamples);
if (swrRet < 0) {
ERROR_LOG(ME, "swr_convert: Error while converting %d", swrRet);
return false;
}
// We always convert to stereo.
__AdjustBGMVolume((s16 *)outbuf, numSamples * 2);
}
return true;
#else
// Zero bytes output. No need to memset.
*outbytes = 0;
return true;
#endif // USE_FFMPEG
}
bool SimpleAudio::Decode(void* inbuf, int inbytes, uint8_t *outbuf, int *outbytes) {
#ifdef USE_FFMPEG
AVPacket packet;
av_init_packet(&packet);
packet.data = static_cast<uint8_t *>(inbuf);
packet.size = inbytes;
int got_frame = 0;
av_frame_unref(frame_);
*outbytes = 0;
srcPos = 0;
int len = avcodec_decode_audio4(codecCtx_, frame_, &got_frame, &packet);
if (len < 0) {
ERROR_LOG(ME, "Error decoding Audio frame");
// TODO: cleanup
return false;
}
av_free_packet(&packet);
// get bytes consumed in source
srcPos = len;
if (got_frame) {
// Initializing the sample rate convert. We will use it to convert float output into int.
int64_t wanted_channel_layout = AV_CH_LAYOUT_STEREO; // we want stereo output layout
int64_t dec_channel_layout = frame_->channel_layout; // decoded channel layout
swrCtx_ = swr_alloc_set_opts(
swrCtx_,
wanted_channel_layout,
AV_SAMPLE_FMT_S16,
wanted_resample_freq,
dec_channel_layout,
codecCtx_->sample_fmt,
codecCtx_->sample_rate,
0,
NULL);
if (!swrCtx_ || swr_init(swrCtx_) < 0) {
ERROR_LOG(ME, "swr_init: Failed to initialize the resampling context");
avcodec_close(codecCtx_);
codec_ = 0;
return false;
}
// convert audio to AV_SAMPLE_FMT_S16
int swrRet = swr_convert(swrCtx_, &outbuf, frame_->nb_samples, (const u8 **)frame_->extended_data, frame_->nb_samples);
if (swrRet < 0) {
ERROR_LOG(ME, "swr_convert: Error while converting %d", swrRet);
return false;
}
swr_free(&swrCtx_);
// output samples per frame, we should *2 since we have two channels
outSamples = swrRet * 2;
// each sample occupies 2 bytes
*outbytes = outSamples * 2;
// We always convert to stereo.
__AdjustBGMVolume((s16 *)outbuf, frame_->nb_samples * 2);
// Save outbuf into pcm audio, you can uncomment this line to save and check the decoded audio into pcm file.
// SaveAudio("dump.pcm", outbuf, *outbytes);
}
return true;
#else
// Zero bytes output. No need to memset.
*outbytes = 0;
return true;
#endif // USE_FFMPEG
}
