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authorMatthias P. Braendli <matthias.braendli@mpb.li>2014-03-07 09:58:58 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2014-03-07 09:58:58 +0100
commitc4c87f4c1e0e053422b8c0f0c5d9378cdd29fe58 (patch)
tree3de0458110ae65307a9d898efe7055a72264d82f /src/AlsaDabplus.cpp
parent9ae8f2259ca1626d0879b2c73e0065307081f505 (diff)
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add first code bits of new alsa encoder
Diffstat (limited to 'src/AlsaDabplus.cpp')
-rw-r--r--src/AlsaDabplus.cpp252
1 files changed, 252 insertions, 0 deletions
diff --git a/src/AlsaDabplus.cpp b/src/AlsaDabplus.cpp
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+/* ------------------------------------------------------------------
+ * Copyright (C) 2011 Martin Storsjo
+ * Copyright (C) 2013,2014 Matthias P. Braendli
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
+ * express or implied.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ * -------------------------------------------------------------------
+ */
+
+#include "AlsaInput.h"
+#include "SampleQueue.h"
+#include "zmq.hpp"
+
+#include <string>
+#include <getopt.h>
+#include <cstdio>
+#include <stdint.h>
+
+#include "libAACenc/include/aacenc_lib.h"
+
+using namespace std;
+
+void usage(const char* name) {
+ fprintf(stderr, "%s [OPTION...]\n", name);
+ fprintf(stderr,
+" -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
+//" -d, --data=FILENAME Set data filename.\n"
+//" -g, --fs-bug Turn on FS bug mitigation.\n"
+//" -i, --input=FILENAME Input filename (default: stdin).\n"
+" -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n"
+" -a, --afterburner Turn on AAC encoder quality increaser.\n"
+//" -m, --message Turn on AAC frame messages.\n"
+//" -p, --pad=BYTES Set PAD size in bytes.\n"
+//" -f, --format={ wav, raw } Set input file format (default: wav).\n"
+" -d, --device=alsa_device Set ALSA input device (default: \"default\").\n"
+" -c, --channels={ 1, 2 } Nb of input channels for raw input (default: 2).\n"
+" -r, --rate={ 32000, 48000 } Sample rate for raw input (default: 48000).\n"
+//" -t, --type=TYPE Set data type (dls|pad|packet|dg).\n"
+//" -v, --verbose=LEVEL Set verbosity level.\n"
+//" -V, --version Print version and exit.\n"
+//" --mi=[ 0, ... ] Set AAC frame messages interval in milliseconds.\n"
+//" --ma=[ 0, ... ] Set AAC frame messages attack time in milliseconds.\n"
+//" -l, --lp Set frame size to 1024 instead of 960.\n"
+"\n"
+"Only the tcp:// zeromq transport has been tested until now.\n"
+
+);
+
+}
+
+int prepare_aac_encoder(
+ HANDLE_AACENCODER *encoder,
+ int subchannel_index,
+ int channels,
+ int sample_rate,
+ int afterburner)
+{
+ HANDLE_AACENCODER handle = *encoder;
+
+ int aot = AOT_DABPLUS_AAC_LC;
+
+ CHANNEL_MODE mode;
+ switch (channels) {
+ case 1: mode = MODE_1; break;
+ case 2: mode = MODE_2; break;
+ default:
+ fprintf(stderr, "Unsupported channels number %d\n", channels);
+ return 1;
+ }
+
+
+ if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
+ fprintf(stderr, "Unable to open encoder\n");
+ return 1;
+ }
+
+
+ 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" : "",
+ channels, sample_rate);
+
+ if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the AOT\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the sample rate\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the channel mode\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the wav channel order\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the granule length\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the RAW transmux\n");
+ return 1;
+ }
+
+ /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*)
+ * != AACENC_OK) {
+ fprintf(stderr, "Unable to set the bitrate mode\n");
+ return 1;
+ }*/
+
+
+ fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
+ if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the bitrate\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the afterburner mode\n");
+ return 1;
+ }
+ if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
+ fprintf(stderr, "Unable to initialize the encoder\n");
+ return 1;
+ }
+ return 0;
+}
+
+
+#define no_argument 0
+#define required_argument 1
+#define optional_argument 2
+
+int main(int argc, char *argv[]) {
+ int subchannel_index = 8; //64kbps subchannel
+ int ch=0;
+ int err;
+ const char *alsa_device = "default";
+ const char *outuri = NULL;
+ int sample_rate=48000, channels=2;
+ const int bytes_per_sample = 2;
+ void *rs_handler = NULL;
+ int afterburner = 0;
+ HANDLE_AACENCODER handle;
+ AACENC_InfoStruct info = { 0 };
+
+ const struct option longopts[] = {
+ {"bitrate", required_argument, 0, 'b'},
+ {"output", required_argument, 0, 'o'},
+ {"device", required_argument, 0, 'd'},
+ {"rate", required_argument, 0, 'r'},
+ {"channels", required_argument, 0, 'c'},
+ {"afterburner", no_argument, 0, 'a'},
+ {"help", no_argument, 0, 'h'},
+ {0,0,0,0},
+ };
+
+ int index;
+ while(ch != -1) {
+ ch = getopt_long(argc, argv, "hab:c:o:r:d:", longopts, &index);
+ switch (ch) {
+ case 'd':
+ alsa_device = optarg;
+ break;
+ case 'a':
+ afterburner = 1;
+ break;
+ case 'b':
+ subchannel_index = atoi(optarg) / 8;
+ break;
+ case 'c':
+ channels = atoi(optarg);
+ break;
+ case 'r':
+ sample_rate = atoi(optarg);
+ break;
+ case 'o':
+ outuri = optarg;
+ break;
+ case '?':
+ case 'h':
+ usage(argv[0]);
+ return 1;
+ }
+ }
+
+ if(subchannel_index < 1 || subchannel_index > 24) {
+ fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n",
+ subchannel_index);
+ return 1;
+ }
+
+ fprintf(stderr, "Setting up ZeroMQ socket\n");
+ if (!outuri) {
+ fprintf(stderr, "ZeroMQ output URI not defined\n");
+ return 1;
+ }
+
+ zmq::context_t zmq_ctx;
+ zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
+ zmq_sock.connect(outuri);
+
+ HANDLE_AACENCODER encoder;
+
+ if (prepare_aac_encoder(&encoder, subchannel_index, channels,
+ sample_rate, afterburner) != 0) {
+ fprintf(stderr, "Encoder preparation failed\n");
+ return 2;
+ }
+
+ if (aacEncInfo(handle, &info) != AACENC_OK) {
+ fprintf(stderr, "Unable to get the encoder info\n");
+ return 1;
+ }
+
+ fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength);
+
+ // Each DAB+ frame will need input_size audio bytes
+ int input_size = channels * bytes_per_sample * info.frameLength;
+ uint8_t input_buf[input_size];
+
+ int max_size = input_size + NUM_SAMPLES_PER_CALL;
+
+ SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);
+
+ AlsaInput alsa_in(alsa_device, channels, sample_rate, queue);
+
+ if (alsa_in.prepare() != 0) {
+ fprintf(stderr, "Alsa preparation failed\n");
+ return 1;
+ }
+
+