add first code bits of new alsa encoder

1 files changed, 252 insertions, 0 deletions

diff --git a/src/AlsaDabplus.cpp b/src/AlsaDabplus.cpp
new file mode 100644
index 0000000..e94f00a
--- /dev/null
+++ b/src/AlsaDabplus.cpp
@@ -0,0 +1,252 @@
+/* ------------------------------------------------------------------
+ * 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;
+    }
+
+