1 files changed, 272 insertions, 0 deletions
diff --git a/test/test_opus.cc b/test/test_opus.cc
new file mode 100644
index 0000000..68ea0f4
--- /dev/null
+++ b/test/test_opus.cc
@@ -0,0 +1,272 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set et sw=2 ts=2: */
+/***************************************************************************
+ *            test_init.cc
+ *
+ *  Mon Sep  2 14:02:16 CEST 2013
+ *  Copyright 2013 Bent Bisballe Nyeng
+ *  deva@aasimon.org
+ ****************************************************************************/
+
+/*
+ *  This file is part of lrtp.
+ *
+ * lrtp is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * lrtp is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with lrtp; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#include <lrtp.h>
+
+#include <stdio.h>
+
+#include <string>
+#include <vector>
+
+#include <math.h>
+#include <opus/opus.h>
+#include <limits.h>
+#include <ao/ao.h>
+
+#define KEY "123456789012345678901234567890123456789012345678901234567890"
+#define SSRC 1234567890
+
+#define FS 8000
+
+#define F1 440
+#define AF1 0.3
+
+#define F2 500
+#define AF2 0.7
+
+void dump(const char *title, const char *buf, size_t size)
+{
+  printf("%12s: ", title);
+  for(int i = 0; i < size; i++) {
+    if(i % 8 == 0) printf(" ");
+    printf("%02x ", (unsigned char)*buf++);
+  }
+  printf("\n");
+}
+
+class Audio {
+public:
+  Audio() {
+    ao_initialize();
+
+    device = NULL;
+    ao_sample_format format;
+    
+    int default_driver = ao_default_driver_id();
+    if(default_driver == -1) {
+      printf("Error could not default driver.\n");
+      return;
+    }   
+    printf("Default driver: %d\n", default_driver);
+
+    format.bits = 16;
+    format.channels = 2;
+    format.rate = FS;
+    format.byte_format = AO_FMT_LITTLE;
+				
+    device = ao_open_live(default_driver, &format, NULL);
+    if(device == NULL) {
+      printf("Error opening device.\n");
+      return;
+    }
+  }
+
+  ~Audio() {
+    if(device) ao_close(device);
+    ao_shutdown();
+  }
+
+  void play(char *pcm, size_t size) {
+    ao_play(device, pcm, size);
+  }
+
+private:
+  ao_device *device;
+};
+
+int main()
+{
+  size_t channels = 2;
+  size_t ms[] = { 120, 240, 480, 960, 1920, 2880 };
+
+  std::vector<std::string> packets;
+  unsigned int csrc = 42;
+  
+  double sin_x = 0;
+  size_t ts = 0;
+
+  printf("========== Encode ==========\n");
+
+  { // Encode
+    struct lrtp_t *lrtp = lrtp_init(KEY, SSRC);
+
+    struct lrtp_profile_t *profile =
+      lrtp_create_profile(lrtp, PROFILE_OPUS, csrc,
+                          //OPTION_RAW_PKG_SIZE, pkg_size,
+                          OPTION_END);
+
+    char packet[16*1024];
+    size_t packetsize = sizeof(packet);
+    
+    int err;
+    OpusEncoder *opus = opus_encoder_create(FS, channels,
+                                            OPUS_APPLICATION_AUDIO, &err);
+    printf("Opus create err: %d\n", err);
+
+    opus_encoder_ctl(opus, OPUS_SET_BITRATE(64000));// [500;512000]
+    opus_encoder_ctl(opus, OPUS_SET_COMPLEXITY(10)); // [0;10]
+    opus_encoder_ctl(opus, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC));	
+
+    int cnt = 0;
+    size_t timestamp = 0;
+    for(unsigned int ts = 0; ts < 120; ts++) {
+      printf("packet #%d\n", ts);
+
+      size_t idx = rand() % (sizeof(ms)/sizeof(size_t));
+      printf("idx: %d\n", idx);
+      size_t pcmsize = ms[idx] / (48000.0 / FS); // Number of samples pr channel
+      short *pcm = new short[100000/*pcmsize * channels*/];
+      for(int i = 0 ; i < pcmsize; i++) {
+        sin_x++;
+        
+        double amp1 = sin((2*M_PI/(double)FS)*(double)sin_x * AF1) * SHRT_MAX;
+        double amp2 = sin((2*M_PI/(double)FS)*(double)sin_x * AF2) * SHRT_MAX;
+        
+        pcm[i*2]   = (short)(sin(2*M_PI/FS*(double)sin_x * F1) * amp1);
+        pcm[i*2+1] = (short)(sin(2*M_PI/FS*(double)sin_x * F2) * amp2);
+      }
+
+      // Master timestamp is sample number in 48kHz (Opus RFC states this)
+      timestamp += pcmsize * 48000 / FS;
+
+      //      size_t pcmsize = pcmsize * channels * sizeof(short);
+
+      char frame[pcmsize];
+      int framesize = sizeof(frame);
+      framesize = opus_encode(opus, pcm, pcmsize,
+                              (unsigned char*)frame, framesize);
+
+      if(framesize < 0) {
+        printf("Opus error: %s\n", opus_strerror(framesize));	
+      }
+
+      printf("Opus Packet: %d bytes from %d bytes\n", pcmsize, framesize);
+
+      int ret = lrtp_enqueue_frame(profile, frame, framesize);
+      while( (ret = lrtp_pack(lrtp, packet, sizeof(packet))) != 0) {
+        std::string p;
+        p.append(packet, ret);
+        packets.push_back(p);
+      }
+
+      delete[] pcm;
+    }
+
+    opus_encoder_destroy(opus);
+    lrtp_destroy_profile(lrtp, csrc);
+    lrtp_close(lrtp);
+  }
+
+  printf("========== Decode ==========\n");
+
+  { // Decode
+    struct lrtp_t *lrtp = lrtp_init(KEY, SSRC);
+
+    struct lrtp_profile_t *profile =
+      lrtp_create_profile(lrtp, PROFILE_OPUS, csrc,
+                          //OPTION_RAW_PKG_SIZE, pkg_size,
+                          OPTION_END);
+
+    int err;
+    OpusDecoder *opus = opus_decoder_create(FS, channels, &err);
+    printf("Opus create err: %d\n", err);
+
+    int idx = (sizeof(ms)/sizeof(size_t)) - 1;
+    printf("idx: %d\n", idx);
+
+    Audio audio;
+
+    char frame[16*1024];
+    
+    int cnt = 0;
+    std::vector<std::string>::iterator i = packets.begin();
+    while(i != packets.end()) {
+      size_t packetsize = i->size();
+      const char *packet = i->data();
+      unsigned int ts;
+
+      printf("unpack sz: %d - %p\n", packetsize, packet);
+
+      lrtp_unpack(lrtp, packet, packetsize);
+      int ret;
+      while((ret = lrtp_dequeue_frame(lrtp, frame, sizeof(frame), &csrc, &ts))
+            != 0) {
+        size_t pcmsize = 16*1024;//ms[idx] / (48000 / FS);
+        short *pcm = new short[pcmsize * channels];
+        printf("pcmsize %d\n", pcmsize); fflush(stdout);
+        int res = opus_decode(opus, (const unsigned char*)frame, ret,
+                              pcm, pcmsize, 0);
+
+        printf("Decompressed %d bytes\n", res);
+        //      pcmsize = res * channels * sizeof(short);
+
+        audio.play((char *)pcm, res * channels * sizeof(short));
+
+        delete[] pcm;
+      }
+
+      i++;
+    }
+
+
+    /*
+
+    std::vector<std::string>::iterator i = packets.begin();
+    while(i != packets.end()) {
+      size_t packetsize = i->size();
+      printf("unpack sz: %d\n", packetsize);
+      const char *packet = i->data();
+      unsigned int ts;
+
+      framesize = sizeof(frame);
+
+      lrtp_unpack(lrtp, packet, packetsize, frame, &framesize, &csrc, &ts);
+      printf("Got %d bytes, csrc %d, ts: %d\n", framesize, csrc, ts);
+
+      size_t pcmsize = 16*1024;//ms[idx] / (48000 / FS);
+      short *pcm = new short[pcmsize * channels];
+      printf("pcmsize %d\n", pcmsize); fflush(stdout);
+      int res = opus_decode(opus, (const unsigned char*)frame, framesize,
+                            pcm, pcmsize, 0);
+      framesize = sizeof(frame);
+
+      printf("Decompressed %d bytes\n", res);
+      //      pcmsize = res * channels * sizeof(short);
+
+			audio.play((char *)pcm, res * channels * sizeof(short));
+
+      delete[] pcm;
+      i++;
+    }
+    */
+    opus_decoder_destroy(opus);
+    lrtp_destroy_profile(lrtp, csrc);
+    lrtp_close(lrtp);
+  }
+
+  return 0;
+}