Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal.

It is advisable to set the packet loss rate of FEC conservatively. Say, if the estimated loss rate is 5%, we can set it to 1%. The risk of degradation in quality is small and the overall performance is good.

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6844 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/acm2/acm_opus.cc

@@ -15,6 +15,8 @@
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
+
+#define SIGN(x) (x > 0 ? 1 : -1)
 #endif
 
 namespace webrtc {
@@ -27,7 +29,9 @@ ACMOpus::ACMOpus(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       sample_freq_(0),
       bitrate_(0),
-      channels_(1) {
+      channels_(1),
+      fec_enabled_(false),
+      packet_loss_rate_(0) {
   return;
 }
 
@@ -70,7 +74,9 @@ ACMOpus::ACMOpus(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       sample_freq_(32000),  // Default sampling frequency.
       bitrate_(20000),  // Default bit-rate.
-      channels_(1) {  // Default mono
+      channels_(1),  // Default mono.
+      fec_enabled_(false),  // Default FEC is off.
+      packet_loss_rate_(0) {
   codec_id_ = codec_id;
   // Opus has internal DTX, but we dont use it for now.
   has_internal_dtx_ = false;
@@ -188,6 +194,7 @@ int16_t ACMOpus::SetBitRateSafe(const int32_t rate) {
                  "SetBitRateSafe: Invalid rate Opus");
     return -1;
   }
+  // Initial packet loss rate.
 
   bitrate_ = rate;
 
@@ -217,11 +224,40 @@ int ACMOpus::SetFEC(bool enable_fec) {
 }
 
 int ACMOpus::SetPacketLossRate(int loss_rate) {
-  // Ask the encoder to change the target packet loss rate.
-  if (WebRtcOpus_SetPacketLossRate(encoder_inst_ptr_, loss_rate) == 0) {
-    packet_loss_rate_ = loss_rate;
+  // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
+  // the input loss rate rounded down to various levels, because a robustly good
+  // audio quality is achieved by lowering the packet loss lower down.
+  // Additionally, to prevent toggling, margins are used, i.e., when jumping to
+  // a loss rate from below, a higher threshold is used than jumping to the same
+  // level from above.
+  const int kPacketLossRateHigh = 20;
+  const int kPacketLossRateMedium = 10;
+  const int kPacketLossRateLow = 1;
+  const int kLossRateHighMargin = 2;
+  const int kLossRateMediumMargin = 1;
+  int opt_loss_rate;
+  if (loss_rate >= kPacketLossRateHigh + kLossRateHighMargin *
+      SIGN(kPacketLossRateHigh - packet_loss_rate_)) {
+    opt_loss_rate = kPacketLossRateHigh;
+  } else if (loss_rate >= kPacketLossRateMedium + kLossRateMediumMargin *
+      SIGN(kPacketLossRateMedium - packet_loss_rate_)) {
+    opt_loss_rate = kPacketLossRateMedium;
+  } else if (loss_rate >= kPacketLossRateLow) {
+    opt_loss_rate = kPacketLossRateLow;
+  } else {
+    opt_loss_rate = 0;
+  }
+
+  if (packet_loss_rate_ == opt_loss_rate) {
     return 0;
   }
+
+  // Ask the encoder to change the target packet loss rate.
+  if (WebRtcOpus_SetPacketLossRate(encoder_inst_ptr_, opt_loss_rate) == 0) {
+    packet_loss_rate_ = opt_loss_rate;
+    return 0;
+  }
+
   return -1;
 }
 

webrtc/modules/audio_coding/main/acm2/acm_opus_unittest.cc

@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/main/acm2/acm_opus.h"
+
+#include "gtest/gtest.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+namespace {
+  const CodecInst kOpusCodecInst = {105, "opus", 48000, 960, 1, 32000};
+  // These constants correspond to those used in ACMOpus::SetPacketLossRate().
+  const int kPacketLossRateHigh = 20;
+  const int kPacketLossRateMedium = 10;
+  const int kPacketLossRateLow = 1;
+  const int kLossRateHighMargin = 2;
+  const int kLossRateMediumMargin = 1;
+}  // namespace
+
+class AcmOpusTest : public ACMOpus {
+ public:
+  explicit AcmOpusTest(int16_t codec_id)
+      : ACMOpus(codec_id) {}
+  ~AcmOpusTest() {}
+  int packet_loss_rate() { return packet_loss_rate_; }
+
+  void TestSetPacketLossRate(int from, int to, int expected_return);
+};
+
+#ifdef WEBRTC_CODEC_OPUS
+void AcmOpusTest::TestSetPacketLossRate(int from, int to, int expected_return) {
+  for (int loss = from; loss <= to; (to >= from) ? ++loss : --loss) {
+    EXPECT_EQ(0, SetPacketLossRate(loss));
+    EXPECT_EQ(expected_return, packet_loss_rate());
+  }
+}
+
+TEST(AcmOpusTest, PacketLossRateOptimized) {
+  AcmOpusTest opus(ACMCodecDB::kOpus);
+  WebRtcACMCodecParams params;
+  memcpy(&(params.codec_inst), &kOpusCodecInst, sizeof(CodecInst));
+  EXPECT_EQ(0, opus.InitEncoder(&params, true));
+  EXPECT_EQ(0, opus.SetFEC(true));
+
+  // Note that the order of the following calls is critical.
+  opus.TestSetPacketLossRate(0, 0, 0);
+  opus.TestSetPacketLossRate(kPacketLossRateLow,
+                             kPacketLossRateMedium + kLossRateMediumMargin - 1,
+                             kPacketLossRateLow);
+  opus.TestSetPacketLossRate(kPacketLossRateMedium + kLossRateMediumMargin,
+                             kPacketLossRateHigh + kLossRateHighMargin - 1,
+                             kPacketLossRateMedium);
+  opus.TestSetPacketLossRate(kPacketLossRateHigh + kLossRateHighMargin,
+                             100,
+                             kPacketLossRateHigh);
+  opus.TestSetPacketLossRate(kPacketLossRateHigh + kLossRateHighMargin,
+                             kPacketLossRateHigh - kLossRateHighMargin,
+                             kPacketLossRateHigh);
+  opus.TestSetPacketLossRate(kPacketLossRateHigh - kLossRateHighMargin - 1,
+                             kPacketLossRateMedium - kLossRateMediumMargin,
+                             kPacketLossRateMedium);
+  opus.TestSetPacketLossRate(kPacketLossRateMedium - kLossRateMediumMargin - 1,
+                             kPacketLossRateLow,
+                             kPacketLossRateLow);
+  opus.TestSetPacketLossRate(0, 0, 0);
+}
+#else
+void AcmOpusTest:TestSetPacketLossRate(int /* from */, int /* to */,
+                                       int /* expected_return */) {
+  return;
+}
+#endif  // WEBRTC_CODEC_OPUS
+
+}  // namespace acm2
+
+}  // namespace webrtc

webrtc/modules/modules.gyp

@@ -101,6 +101,7 @@
             '<(webrtc_root)/test/test.gyp:rtcp_packet_parser',
           ],
           'sources': [
+            'audio_coding/main/acm2/acm_opus_unittest.cc',
             'audio_coding/main/acm2/acm_receiver_unittest.cc',
             'audio_coding/main/acm2/audio_coding_module_unittest.cc',
             'audio_coding/main/acm2/call_statistics_unittest.cc',