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authorMatthias P. Braendli <matthias.braendli@mpb.li>2017-07-24 14:43:24 +0200
committerMatthias P. Braendli <matthias.braendli@mpb.li>2017-07-24 14:43:24 +0200
commit62409ce26f1fd41d915c4022088fcb8b133ae6c6 (patch)
tree9ce0442470430445daf09ad6deb5f52026078c4b /src/odr-audioenc.cpp
parent864cebd1d753a603de9a24d26a830bad87e2a42f (diff)
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Add a --decoder option for loopback testing
Diffstat (limited to 'src/odr-audioenc.cpp')
-rw-r--r--src/odr-audioenc.cpp28
1 files changed, 28 insertions, 0 deletions
diff --git a/src/odr-audioenc.cpp b/src/odr-audioenc.cpp
index 84218c8..edb98f7 100644
--- a/src/odr-audioenc.cpp
+++ b/src/odr-audioenc.cpp
@@ -53,6 +53,7 @@
#include "JackInput.h"
#include "VLCInput.h"
#include "SampleQueue.h"
+#include "AACDecoder.h"
#include "zmq.hpp"
#include "common.h"
#include "wavfile.h"
@@ -177,6 +178,7 @@ void usage(const char* name)
" --sbr Force the usage of SBR (HE-AAC)\n"
" --ps Force the usage of SBR and PS (HE-AACv2)\n"
" -B, --bandwidth=VALUE Set the AAC encoder bandwidth to VALUE [Hz].\n"
+ " --decode=FILE Decode the AAC back to a wav file (loopback test).\n"
" Output and pad parameters:\n"
" -o, --output=URI Output ZMQ uri. (e.g. 'tcp://localhost:9000')\n"
" -or- Output file uri. (e.g. 'file.dabp')\n"
@@ -419,6 +421,8 @@ int main(int argc, char *argv[])
AACENC_InfoStruct info = { 0 };
int aot = AOT_NONE;
+ std::string decode_wavfilename;
+
std::string dab_channel_mode;
int dab_psy_model = 1;
std::deque<uint8_t> toolame_output_buffer;
@@ -454,6 +458,7 @@ int main(int argc, char *argv[])
{"dabmode", required_argument, 0, 4 },
{"dabpsy", required_argument, 0, 5 },
{"device", required_argument, 0, 'd'},
+ {"decode", required_argument, 0, 6 },
{"format", required_argument, 0, 'f'},
{"input", required_argument, 0, 'i'},
{"jack", required_argument, 0, 'j'},
@@ -528,6 +533,9 @@ int main(int argc, char *argv[])
case 5: // DAB psy model
dab_psy_model = std::stoi(optarg);
break;
+ case 6: // Enable loopback decoder for AAC
+ decode_wavfilename = optarg;
+ break;
case 'a':
selected_encoder = encoder_selection_t::toolame_dab;
break;
@@ -756,6 +764,7 @@ int main(int argc, char *argv[])
vec_u8 input_buf;
HANDLE_AACENCODER encoder;
+ std::unique_ptr<AACDecoder> decoder;
if (selected_encoder == encoder_selection_t::fdk_dabplus) {
int subchannel_index = bitrate / 8;
@@ -777,6 +786,10 @@ int main(int argc, char *argv[])
input_size);
input_buf.resize(input_size);
+
+ if (not decode_wavfilename.empty()) {
+ decoder.reset(new AACDecoder(decode_wavfilename.c_str()));
+ }
}
else if (selected_encoder == encoder_selection_t::toolame_dab) {
int err = toolame_init();
@@ -821,6 +834,11 @@ int main(int argc, char *argv[])
}
input_buf.resize(channels * 1152 * BYTES_PER_SAMPLE);
+
+ if (not decode_wavfilename.empty()) {
+ fprintf(stderr, "--decode not supported for DAB\n");
+ return 1;
+ }
}
/* We assume that we need to call the encoder
@@ -1225,6 +1243,16 @@ int main(int argc, char *argv[])
}
}
+ if (numOutBytes != 0 and decoder) {
+ try {
+ decoder->decode_frame(outbuf.data(), numOutBytes);
+ }
+ catch (std::runtime_error &e) {
+ fprintf(stderr, "Decoding failed with: %s\n", e.what());
+ return 1;
+ }
+ }
+
/* Check if the encoder has generated output data.
* DAB+ requires RS encoding, which is not done in ODR-DabMux and not necessary
* for DAB.