From bb697e902b19a25a8f84de2ff0ade2f601303352 Mon Sep 17 00:00:00 2001 From: "Matthias P. Braendli" Date: Fri, 16 May 2014 10:20:04 +0200 Subject: Fix INTERNAL ERROR and add AOT options --- src/dabplus-enc.cpp | 62 ++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 42 insertions(+), 20 deletions(-) (limited to 'src') diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp index da837fb..a02444f 100644 --- a/src/dabplus-enc.cpp +++ b/src/dabplus-enc.cpp @@ -86,15 +86,20 @@ void usage(const char* name) { " -f, --format={ wav, raw } Set input file format (default: wav).\n" " Encoder parameters:\n" " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n" - " -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n" - " -or- Output file uri. (e.g. 'file.dab')\n" - " -or- a single dash '-' to denote stdout\n" " -a, --afterburner Turn on AAC encoder quality increaser.\n" - " -p, --pad=BYTES Set PAD size in bytes.\n" - " -P, --pad-fifo=FILENAME Set PAD data input fifo name (default: /tmp/pad.fifo).\n" " -c, --channels={ 1, 2 } Nb of input channels (default: 2).\n" " -r, --rate={ 32000, 48000 } Input sample rate (default: 48000).\n" + " --aaclc Force the usage of AAC-LC (no SBR, no PS)\n" + " --sbr Force the usage of SBR\n" + " --ps Force the usage of PS\n" + " Output and pad parameters:\n" + " -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n" + " -or- Output file uri. (e.g. 'file.dab')\n" + " -or- a single dash '-' to denote stdout\n" " -k, --secret-key=FILE Enable ZMQ encryption with the given secret key.\n" + " -p, --pad=BYTES Set PAD size in bytes.\n" + " -P, --pad-fifo=FILENAME Set PAD data input fifo name" + " (default: /tmp/pad.fifo).\n" " -l, --level Show peak audio level indication.\n" "\n" "Only the tcp:// zeromq transport has been tested until now,\n" @@ -108,12 +113,11 @@ int prepare_aac_encoder( int subchannel_index, int channels, int sample_rate, - int afterburner) + int afterburner, + int *aot) { HANDLE_AACENCODER handle = *encoder; - int aot = AOT_DABPLUS_AAC_LC; - CHANNEL_MODE mode; switch (channels) { case 1: mode = MODE_1; break; @@ -131,19 +135,24 @@ int prepare_aac_encoder( *encoder = handle; - if(channels == 2 && subchannel_index <= 6) - aot = AOT_DABPLUS_PS; - else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) - aot = AOT_DABPLUS_SBR; + if (*aot == AOT_NONE) { + + if(channels == 2 && subchannel_index <= 6) { + *aot = AOT_DABPLUS_PS; + } + else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) { + *aot = AOT_DABPLUS_SBR; + } + } fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n", subchannel_index, - aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "", - aot == AOT_DABPLUS_SBR ? "HE-AAC" : "", - aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "", + *aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "", + *aot == AOT_DABPLUS_SBR ? "HE-AAC" : "", + *aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "", channels, sample_rate); - if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) { + if (aacEncoder_SetParam(handle, AACENC_AOT, *aot) != AACENC_OK) { fprintf(stderr, "Unable to set the AOT\n"); return 1; } @@ -221,6 +230,7 @@ int main(int argc, char *argv[]) bool afterburner = false; bool drift_compensation = false; AACENC_InfoStruct info = { 0 }; + int aot = AOT_NONE; /* Keep track of peaks */ int peak_left = 0; @@ -257,6 +267,9 @@ int main(int argc, char *argv[]) {"drift-comp", no_argument, 0, 'D'}, {"help", no_argument, 0, 'h'}, {"level", no_argument, 0, 'l'}, + {"aaclc", no_argument, 0, 0 }, + {"sbr", no_argument, 0, 1 }, + {"ps", no_argument, 0, 2 }, {0,0,0,0}, }; @@ -269,6 +282,15 @@ int main(int argc, char *argv[]) while(ch != -1) { ch = getopt_long(argc, argv, "ahDlb:c:f:i:k:o:r:d:p:P:", longopts, &index); switch (ch) { + case 0: // AAC-LC + aot = AOT_DABPLUS_AAC_LC; + break; + case 1: // SBR + aot = AOT_DABPLUS_SBR; + break; + case 2: // PS + aot = AOT_DABPLUS_PS; + break; case 'a': afterburner = true; break; @@ -339,7 +361,8 @@ int main(int argc, char *argv[]) * frame. This information is used when the alsa drift compensation * is active */ - const int enc_calls_per_output = sample_rate / 16000; + const int enc_calls_per_output = + (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000; zmq::context_t zmq_ctx; zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB); @@ -407,7 +430,7 @@ int main(int argc, char *argv[]) HANDLE_AACENCODER encoder; if (prepare_aac_encoder(&encoder, subchannel_index, channels, - sample_rate, afterburner) != 0) { + sample_rate, afterburner, &aot) != 0) { fprintf(stderr, "Encoder preparation failed\n"); return 2; } @@ -646,8 +669,7 @@ int main(int argc, char *argv[]) if (out_args.numOutBytes != 0) { // Our timing code depends on this - if (! ((sample_rate == 32000 && calls == 2) || - (sample_rate == 48000 && calls == 3)) ) { + if (calls != enc_calls_per_output) { fprintf(stderr, "INTERNAL ERROR! sample rate %d, calls %d\n", sample_rate, calls); } -- cgit v1.2.3