Reapply "Advertise G722 as 8 kHz rather than 16 kHz""

This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.

BUG=3951
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/media/webrtc/webrtcvoiceengine.cc

@@ -74,7 +74,7 @@ static const CodecPref kCodecPrefs[] = {
   { "ISAC",   32000,  1, 104, true },
   { "CELT",   32000,  1, 109, true },
   { "CELT",   32000,  2, 110, true },
-  { "G722",   16000,  1, 9,   false },
+  { "G722",   8000,   1, 9,   false },
   { "ILBC",   8000,   1, 102, false },
   { "PCMU",   8000,   1, 0,   false },
   { "PCMA",   8000,   1, 8,   false },
@@ -110,6 +110,7 @@ static const int kDefaultAudioDeviceId = 0;
 
 static const char kIsacCodecName[] = "ISAC";
 static const char kL16CodecName[] = "L16";
+static const char kG722CodecName[] = "G722";
 
 // Parameter used for NACK.
 // This value is equivalent to 5 seconds of audio data at 20 ms per packet.
@@ -485,12 +486,24 @@ static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
   voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
 }
 
+// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
+// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
+// codec.
+static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
+  if (_stricmp(voe_codec->plname, kG722CodecName) == 0) {
+    // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
+    // has changed, and this special case is no longer needed.
+    ASSERT(voe_codec->plfreq != new_plfreq);
+    voe_codec->plfreq = new_plfreq;
+  }
+}
+
 void WebRtcVoiceEngine::ConstructCodecs() {
   LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
   for (int i = 0; i < ncodecs; ++i) {
     webrtc::CodecInst voe_codec;
-    if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
+    if (GetVoeCodec(i, voe_codec)) {
       // Skip uncompressed formats.
       if (_stricmp(voe_codec.plname, kL16CodecName) == 0) {
         continue;
@@ -540,6 +553,15 @@ void WebRtcVoiceEngine::ConstructCodecs() {
   std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable);
 }
 
+bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst& codec) {
+  if (voe_wrapper_->codec()->GetCodec(index, codec) != -1) {
+    // Change the sample rate of G722 to 8000 to match SDP.
+    MaybeFixupG722(&codec, 8000);
+    return true;
+  }
+  return false;
+}
+
 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
   LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
   if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) {
@@ -1224,7 +1246,7 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
   for (int i = 0; i < ncodecs; ++i) {
     webrtc::CodecInst voe_codec;
-    if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
+    if (GetVoeCodec(i, voe_codec)) {
       AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
                        voe_codec.rate, voe_codec.channels, 0);
       bool multi_rate = IsCodecMultiRate(voe_codec);
@@ -1243,6 +1265,9 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
             voe_codec.rate = in.bitrate;
           }
 
+          // Reset G722 sample rate to 16000 to match WebRTC.
+          MaybeFixupG722(&voe_codec, 16000);
+
           // Apply codec-specific settings.
           if (IsIsac(codec)) {
             // If ISAC and an explicit bitrate is not specified,

talk/media/webrtc/webrtcvoiceengine.h

@@ -199,6 +199,7 @@ class WebRtcVoiceEngine
 
   void Construct();
   void ConstructCodecs();
+  bool GetVoeCodec(int index, webrtc::CodecInst& codec);
   bool InitInternal();
   bool EnsureSoundclipEngineInit();
   void SetTraceFilter(int filter);

talk/media/webrtc/webrtcvoiceengine_unittest.cc

@@ -52,14 +52,16 @@ static const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0);
 static const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
 static const cricket::AudioCodec kCeltCodec(110, "CELT", 32000, 64000, 2, 0);
 static const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2, 0);
+static const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 0);
+static const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1, 0);
 static const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1, 0);
 static const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1, 0);
 static const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0);
 static const cricket::AudioCodec
     kTelephoneEventCodec(106, "telephone-event", 8000, 0, 1, 0);
 static const cricket::AudioCodec* const kAudioCodecs[] = {
-    &kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kRedCodec,
+    &kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kG722CodecVoE,
-    &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec,
+    &kRedCodec, &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec,
 };
 const char kRingbackTone[] = "RIFF____WAVE____ABCD1234";
 static uint32 kSsrc1 = 0x99;
@@ -770,6 +772,20 @@ TEST_F(WebRtcVoiceEngineTestFake, DontResetSetSendCodec) {
   EXPECT_EQ(1, voe_.GetNumSetSendCodecs());
 }
 
+// Verify that G722 is set with 16000 samples per second to WebRTC.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecG722) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kG722CodecSdp);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("G722", gcodec.plname);
+  EXPECT_EQ(1, gcodec.channels);
+  EXPECT_EQ(16000, gcodec.plfreq);
+}
+
 // Test that if clockrate is not 48000 for opus, we fail.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
   EXPECT_TRUE(SetupEngine());
@@ -3208,7 +3224,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
   EXPECT_TRUE(engine.FindCodec(
       cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0)));
   EXPECT_TRUE(engine.FindCodec(
-      cricket::AudioCodec(96, "G722", 16000, 0, 1, 0)));
+      cricket::AudioCodec(96, "G722", 8000, 0, 1, 0)));
   EXPECT_TRUE(engine.FindCodec(
       cricket::AudioCodec(96, "red", 8000, 0, 1, 0)));
   EXPECT_TRUE(engine.FindCodec(
@@ -3225,7 +3241,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
   EXPECT_TRUE(engine.FindCodec(
       cricket::AudioCodec(8, "", 8000, 0, 1, 0)));   // PCMA
   EXPECT_TRUE(engine.FindCodec(
-      cricket::AudioCodec(9, "", 16000, 0, 1, 0)));  // G722
+      cricket::AudioCodec(9, "", 8000, 0, 1, 0)));  // G722
   EXPECT_TRUE(engine.FindCodec(
       cricket::AudioCodec(13, "", 8000, 0, 1, 0)));  // CN
   // Check sample/bitrate matching.
@@ -3248,7 +3264,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
       EXPECT_EQ(103, it->id);
     } else if (it->name == "ISAC" && it->clockrate == 32000) {
       EXPECT_EQ(104, it->id);
-    } else if (it->name == "G722" && it->clockrate == 16000) {
+    } else if (it->name == "G722" && it->clockrate == 8000) {
       EXPECT_EQ(9, it->id);
     } else if (it->name == "telephone-event") {
       EXPECT_EQ(126, it->id);

talk/session/media/mediasessionclient_unittest.cc

@@ -61,7 +61,7 @@ static const cricket::AudioCodec kAudioCodecs[] = {
   cricket::AudioCodec(119, "ISACLC", 16000, 40000, 1, 16),
   cricket::AudioCodec(99,  "speex",  16000, 22000, 1, 15),
   cricket::AudioCodec(97,  "IPCMWB", 16000, 80000, 1, 14),
-  cricket::AudioCodec(9,   "G722",   16000, 64000, 1, 13),
+  cricket::AudioCodec(9,   "G722",   8000,  64000, 1, 13),
   cricket::AudioCodec(102, "iLBC",   8000,  13300, 1, 12),
   cricket::AudioCodec(98,  "speex",  8000,  11000, 1, 11),
   cricket::AudioCodec(3,   "GSM",    8000,  13000, 1, 10),
@@ -81,7 +81,7 @@ static const cricket::AudioCodec kAudioCodecs[] = {
 static const cricket::AudioCodec kAudioCodecsDifferentPreference[] = {
   cricket::AudioCodec(104, "ISAC",   32000, -1,    1, 17),
   cricket::AudioCodec(97,  "IPCMWB", 16000, 80000, 1, 14),
-  cricket::AudioCodec(9,   "G722",   16000, 64000, 1, 13),
+  cricket::AudioCodec(9,   "G722",   8000,  64000, 1, 13),
   cricket::AudioCodec(119, "ISACLC", 16000, 40000, 1, 16),
   cricket::AudioCodec(103, "ISAC",   16000, -1,    1, 18),
   cricket::AudioCodec(99,  "speex",  16000, 22000, 1, 15),
@@ -197,7 +197,7 @@ const std::string kGingleInitiate(
      "      <payload-type xmlns='http://www.google.com/session/phone' " \
      "        id='97' name='IPCMWB' clockrate='16000' bitrate='80000' />   " \
      "      <payload-type xmlns='http://www.google.com/session/phone' " \
-     "        id='9' name='G722' clockrate='16000' bitrate='64000' /> " \
+     "        id='9' name='G722' clockrate='8000' bitrate='64000' /> " \
      "      <payload-type xmlns='http://www.google.com/session/phone' " \
      "        id='102' name='iLBC' clockrate='8000' bitrate='13300' />" \
      "      <payload-type xmlns='http://www.google.com/session/phone' " \
@@ -248,7 +248,7 @@ const std::string kJingleInitiate(
      "          <parameter name='bitrate' value='80000'/>               " \
      "        </payload-type>                                           " \
      "        <payload-type                                             " \
-     "          id='9' name='G722' clockrate='16000'>                   " \
+     "          id='9' name='G722' clockrate='8000'>                   " \
      "          <parameter name='bitrate' value='64000'/>               " \
      "        </payload-type>                                           " \
      "        <payload-type                                             " \
@@ -1918,7 +1918,7 @@ class MediaSessionClientTest : public sigslot::has_slots<> {
      e = NextFromPayloadType(e);
     ASSERT_TRUE(e != NULL);
     codec = AudioCodecFromPayloadType(e);
-    VerifyAudioCodec(codec, 9, "G722", 16000, 64000, 1);
+    VerifyAudioCodec(codec, 9, "G722", 8000, 64000, 1);
 
     e = NextFromPayloadType(e);
     ASSERT_TRUE(e != NULL);
@@ -2112,7 +2112,7 @@ class MediaSessionClientTest : public sigslot::has_slots<> {
     codec = AudioCodecFromPayloadType(e);
     ASSERT_EQ(9, codec.id);
     ASSERT_EQ("G722", codec.name);
-    ASSERT_EQ(16000, codec.clockrate);
+    ASSERT_EQ(8000, codec.clockrate);
     ASSERT_EQ(64000, codec.bitrate);
     ASSERT_EQ(1, codec.channels);