diff options
author | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-11-06 08:55:01 +0000 |
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committer | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-11-06 08:55:01 +0000 |
commit | 05e3f539cc794fe04d2e9b2d85c437f7e069ba6c (patch) | |
tree | 2ec185229c4268532ee7ee7e3262b2c4aa64c911 | |
parent | 7d974c11e23898cd59838c79751b96c45b09ec4b (diff) | |
download | talk-05e3f539cc794fe04d2e9b2d85c437f7e069ba6c.tar.gz |
Advertise G722 as 8 kHz rather than 16 kHz
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.
R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.
Review URL: https://webrtc-codereview.appspot.com/27879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
-rw-r--r-- | media/webrtc/webrtcvoiceengine.cc | 29 | ||||
-rw-r--r-- | media/webrtc/webrtcvoiceengine.h | 1 | ||||
-rw-r--r-- | media/webrtc/webrtcvoiceengine_unittest.cc | 26 |
3 files changed, 49 insertions, 7 deletions
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc index 95e16e4..6394f09 100644 --- a/media/webrtc/webrtcvoiceengine.cc +++ b/media/webrtc/webrtcvoiceengine.cc @@ -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, diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h index f19059b..34b9f3c 100644 --- a/media/webrtc/webrtcvoiceengine.h +++ b/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); diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc index 5deabd2..5eb6e24 100644 --- a/media/webrtc/webrtcvoiceengine_unittest.cc +++ b/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, - &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec, + &kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kG722CodecVoE, + &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); |