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Removing CELT.

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CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.

BUG=
R=pbos@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36099004

Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org 2015-02-17 12:36:41 +00:00
parent 2c1bcf2cb4
commit f9b5c1b3d0
6 changed files with 24 additions and 76 deletions
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18
talk/app/webrtc/webrtcsdp_unittest.cc
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@ -157,7 +157,7 @@ static const char kSdpFullString[] =
"dummy_session_params\r\n" "dummy_session_params\r\n"
"a=rtpmap:111 opus/48000/2\r\n" "a=rtpmap:111 opus/48000/2\r\n"
"a=rtpmap:103 ISAC/16000\r\n" "a=rtpmap:103 ISAC/16000\r\n"
"a=rtpmap:104 CELT/32000/2\r\n" "a=rtpmap:104 ISAC/32000\r\n"
"a=ssrc:1 cname:stream_1_cname\r\n" "a=ssrc:1 cname:stream_1_cname\r\n"
"a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n" "a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n"
"a=ssrc:1 mslabel:local_stream_1\r\n" "a=ssrc:1 mslabel:local_stream_1\r\n"
@ -224,7 +224,7 @@ static const char kSdpString[] =
"dummy_session_params\r\n" "dummy_session_params\r\n"
"a=rtpmap:111 opus/48000/2\r\n" "a=rtpmap:111 opus/48000/2\r\n"
"a=rtpmap:103 ISAC/16000\r\n" "a=rtpmap:103 ISAC/16000\r\n"
"a=rtpmap:104 CELT/32000/2\r\n" "a=rtpmap:104 ISAC/32000\r\n"
"a=ssrc:1 cname:stream_1_cname\r\n" "a=ssrc:1 cname:stream_1_cname\r\n"
"a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n" "a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n"
"a=ssrc:1 mslabel:local_stream_1\r\n" "a=ssrc:1 mslabel:local_stream_1\r\n"
@ -531,7 +531,7 @@ class WebRtcSdpTest : public testing::Test {
AudioCodec opus(111, "opus", 48000, 0, 2, 3); AudioCodec opus(111, "opus", 48000, 0, 2, 3);
audio_desc_->AddCodec(opus); audio_desc_->AddCodec(opus);
audio_desc_->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1, 2)); audio_desc_->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1, 2));
audio_desc_->AddCodec(AudioCodec(104, "CELT", 32000, 0, 2, 1)); audio_desc_->AddCodec(AudioCodec(104, "ISAC", 32000, 56000, 1, 1));
desc_.AddContent(kAudioContentName, NS_JINGLE_RTP, audio_desc_); desc_.AddContent(kAudioContentName, NS_JINGLE_RTP, audio_desc_);
// VideoContentDescription // VideoContentDescription
@ -1181,13 +1181,12 @@ class WebRtcSdpTest : public testing::Test {
// description. // description.
"a=msid-semantic: WMS\r\n" "a=msid-semantic: WMS\r\n"
// Pl type 111 preferred. // Pl type 111 preferred.
"m=audio 9 RTP/SAVPF 111 104 103 102\r\n" "m=audio 9 RTP/SAVPF 111 104 103\r\n"
// Pltype 111 listed before 103 and 104 in the map. // Pltype 111 listed before 103 and 104 in the map.
"a=rtpmap:111 opus/48000/2\r\n" "a=rtpmap:111 opus/48000/2\r\n"
// Pltype 103 listed before 104. // Pltype 103 listed before 104.
"a=rtpmap:103 ISAC/16000\r\n" "a=rtpmap:103 ISAC/16000\r\n"
"a=rtpmap:104 CELT/32000/2\r\n" "a=rtpmap:104 ISAC/32000\r\n"
"a=rtpmap:102 ISAC/32000/1\r\n"
"a=fmtp:111 0-15,66,70\r\n" "a=fmtp:111 0-15,66,70\r\n"
"a=fmtp:111 "; "a=fmtp:111 ";
std::ostringstream os; std::ostringstream os;
@ -1794,7 +1793,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmap) {
"t=0 0\r\n" "t=0 0\r\n"
"m=audio 49232 RTP/AVP 0 18 103\r\n" "m=audio 49232 RTP/AVP 0 18 103\r\n"
// Codec that doesn't appear in the m= line will be ignored. // Codec that doesn't appear in the m= line will be ignored.
"a=rtpmap:104 CELT/32000/2\r\n" "a=rtpmap:104 ISAC/32000\r\n"
// The rtpmap line for static payload codec is optional. // The rtpmap line for static payload codec is optional.
"a=rtpmap:18 G729/16000\r\n" "a=rtpmap:18 G729/16000\r\n"
"a=rtpmap:103 ISAC/16000\r\n"; "a=rtpmap:103 ISAC/16000\r\n";
@ -2407,7 +2406,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithReorderedPltypes) {
"a=rtpmap:111 opus/48000/2\r\n" // Pltype 111 listed before 103 and 104 "a=rtpmap:111 opus/48000/2\r\n" // Pltype 111 listed before 103 and 104
// in the map. // in the map.
"a=rtpmap:103 ISAC/16000\r\n" // Pltype 103 listed before 104 in the map. "a=rtpmap:103 ISAC/16000\r\n" // Pltype 103 listed before 104 in the map.
"a=rtpmap:104 CELT/32000/2\r\n"; "a=rtpmap:104 ISAC/32000\r\n";
// Deserialize // Deserialize
EXPECT_TRUE(SdpDeserialize(kSdpWithReorderedPlTypesString, &jdesc_output)); EXPECT_TRUE(SdpDeserialize(kSdpWithReorderedPlTypesString, &jdesc_output));
@ -2417,7 +2416,8 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithReorderedPltypes) {
const AudioContentDescription* acd = const AudioContentDescription* acd =
static_cast<const AudioContentDescription*>(ac->description); static_cast<const AudioContentDescription*>(ac->description);
ASSERT_FALSE(acd->codecs().empty()); ASSERT_FALSE(acd->codecs().empty());
EXPECT_EQ("CELT", acd->codecs()[0].name); EXPECT_EQ("ISAC", acd->codecs()[0].name);
EXPECT_EQ(32000, acd->codecs()[0].clockrate);
EXPECT_EQ(104, acd->codecs()[0].id); EXPECT_EQ(104, acd->codecs()[0].id);
} }

2
talk/media/webrtc/webrtcvoiceengine.cc
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@ -72,8 +72,6 @@ static const CodecPref kCodecPrefs[] = {
{ "OPUS", 48000, 2, 111, true }, { "OPUS", 48000, 2, 111, true },
{ "ISAC", 16000, 1, 103, true }, { "ISAC", 16000, 1, 103, true },
{ "ISAC", 32000, 1, 104, true }, { "ISAC", 32000, 1, 104, true },
{ "CELT", 32000, 1, 109, true },
{ "CELT", 32000, 2, 110, true },
// G722 should be advertised as 8000 Hz because of the RFC "bug". // G722 should be advertised as 8000 Hz because of the RFC "bug".
{ "G722", 8000, 1, 9, false }, { "G722", 8000, 1, 9, false },
{ "ILBC", 8000, 1, 102, false }, { "ILBC", 8000, 1, 102, false },

66
talk/media/webrtc/webrtcvoiceengine_unittest.cc
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@ -50,7 +50,6 @@ using cricket::kRtpAbsoluteSenderTimeHeaderExtension;
static const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0); 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 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 kOpusCodec(111, "opus", 48000, 64000, 2, 0);
static const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 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 kG722CodecSdp(9, "G722", 8000, 64000, 1, 0);
@ -60,8 +59,8 @@ static const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0);
static const cricket::AudioCodec static const cricket::AudioCodec
kTelephoneEventCodec(106, "telephone-event", 8000, 0, 1, 0); kTelephoneEventCodec(106, "telephone-event", 8000, 0, 1, 0);
static const cricket::AudioCodec* const kAudioCodecs[] = { static const cricket::AudioCodec* const kAudioCodecs[] = {
&kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kG722CodecVoE, &kPcmuCodec, &kIsacCodec, &kOpusCodec, &kG722CodecVoE, &kRedCodec,
&kRedCodec, &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec, &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec,
}; };
const char kRingbackTone[] = "RIFF____WAVE____ABCD1234"; const char kRingbackTone[] = "RIFF____WAVE____ABCD1234";
static uint32 kSsrc1 = 0x99; static uint32 kSsrc1 = 0x99;
@ -645,9 +644,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
// PCMU, default bitrate == 64000. // PCMU, default bitrate == 64000.
TestSendBandwidth(kPcmuCodec, -1, true, 64000); TestSendBandwidth(kPcmuCodec, -1, true, 64000);
// CELT, default bitrate == 64000.
TestSendBandwidth(kCeltCodec, 0, true, 64000);
// opus, default bitrate == 64000. // opus, default bitrate == 64000.
TestSendBandwidth(kOpusCodec, -1, true, 64000); TestSendBandwidth(kOpusCodec, -1, true, 64000);
} }
@ -662,10 +658,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
TestSendBandwidth(kIsacCodec, 128000, true, 128000); TestSendBandwidth(kIsacCodec, 128000, true, 128000);
TestSendBandwidth(kIsacCodec, 16000, true, 16000); TestSendBandwidth(kIsacCodec, 16000, true, 16000);
// CELT, default bitrate == 64000.
TestSendBandwidth(kCeltCodec, 96000, true, 96000);
TestSendBandwidth(kCeltCodec, 32000, true, 32000);
// opus, default bitrate == 64000. // opus, default bitrate == 64000.
TestSendBandwidth(kOpusCodec, 96000, true, 96000); TestSendBandwidth(kOpusCodec, 96000, true, 96000);
TestSendBandwidth(kOpusCodec, 48000, true, 48000); TestSendBandwidth(kOpusCodec, 48000, true, 48000);
@ -1410,59 +1402,31 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) {
voe_.GetMaxEncodingBandwidth(channel_num)); voe_.GetMaxEncodingBandwidth(channel_num));
} }
// Test that we can apply CELT with stereo mode but fail with mono mode. // Test that we can switch back and forth between Opus and ISAC with CN.
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCelt) { TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) {
EXPECT_TRUE(SetupEngine()); EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel(); int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> codecs; std::vector<cricket::AudioCodec> opus_codecs;
codecs.push_back(kCeltCodec); opus_codecs.push_back(kOpusCodec);
codecs.push_back(kIsacCodec); EXPECT_TRUE(channel_->SetSendCodecs(opus_codecs));
codecs[0].id = 96;
codecs[0].channels = 2;
codecs[0].bitrate = 96000;
codecs[1].bitrate = 64000;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
webrtc::CodecInst gcodec; webrtc::CodecInst gcodec;
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_EQ(96, gcodec.pltype); EXPECT_EQ(111, gcodec.pltype);
EXPECT_EQ(96000, gcodec.rate); EXPECT_STREQ("opus", gcodec.plname);
EXPECT_EQ(2, gcodec.channels);
EXPECT_STREQ("CELT", gcodec.plname);
// Doesn't support mono, expect it to fall back to the next codec in the list.
codecs[0].channels = 1;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_EQ(103, gcodec.pltype);
EXPECT_EQ(1, gcodec.channels);
EXPECT_EQ(64000, gcodec.rate);
EXPECT_STREQ("ISAC", gcodec.plname);
}
// Test that we can switch back and forth between CELT and ISAC with CN.
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacCeltSwitching) {
EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> celt_codecs;
celt_codecs.push_back(kCeltCodec);
EXPECT_TRUE(channel_->SetSendCodecs(celt_codecs));
webrtc::CodecInst gcodec;
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_EQ(110, gcodec.pltype);
EXPECT_STREQ("CELT", gcodec.plname);
std::vector<cricket::AudioCodec> isac_codecs; std::vector<cricket::AudioCodec> isac_codecs;
isac_codecs.push_back(kIsacCodec); isac_codecs.push_back(kIsacCodec);
isac_codecs.push_back(kCn16000Codec); isac_codecs.push_back(kCn16000Codec);
isac_codecs.push_back(kCeltCodec); isac_codecs.push_back(kOpusCodec);
EXPECT_TRUE(channel_->SetSendCodecs(isac_codecs)); EXPECT_TRUE(channel_->SetSendCodecs(isac_codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_EQ(103, gcodec.pltype); EXPECT_EQ(103, gcodec.pltype);
EXPECT_STREQ("ISAC", gcodec.plname); EXPECT_STREQ("ISAC", gcodec.plname);
EXPECT_TRUE(channel_->SetSendCodecs(celt_codecs)); EXPECT_TRUE(channel_->SetSendCodecs(opus_codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_EQ(110, gcodec.pltype); EXPECT_EQ(111, gcodec.pltype);
EXPECT_STREQ("CELT", gcodec.plname); EXPECT_STREQ("opus", gcodec.plname);
} }
// Test that we handle various ways of specifying bitrate. // Test that we handle various ways of specifying bitrate.
@ -2482,8 +2446,8 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) {
cricket::StreamParams::CreateLegacy(kSsrc1))); cricket::StreamParams::CreateLegacy(kSsrc1)));
int channel_num2 = voe_.GetLastChannel(); int channel_num2 = voe_.GetLastChannel();
webrtc::CodecInst gcodec; webrtc::CodecInst gcodec;
rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname), "CELT"); rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname), "opus");
gcodec.plfreq = 32000; gcodec.plfreq = 48000;
gcodec.channels = 2; gcodec.channels = 2;
EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec)); EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec));
} }

1
talk/session/media/channelmanager_unittest.cc
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@ -41,7 +41,6 @@ namespace cricket {
static const AudioCodec kAudioCodecs[] = { static const AudioCodec kAudioCodecs[] = {
AudioCodec(97, "voice", 1, 2, 3, 0), AudioCodec(97, "voice", 1, 2, 3, 0),
AudioCodec(110, "CELT", 32000, 48000, 2, 0),
AudioCodec(111, "OPUS", 48000, 32000, 2, 0), AudioCodec(111, "OPUS", 48000, 32000, 2, 0),
}; };

12
webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
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@ -466,9 +466,6 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst, int* mirror_id) {
} else if (STR_CASE_CMP("speex", codec_inst.plname) == 0) { } else if (STR_CASE_CMP("speex", codec_inst.plname) == 0) {
return IsSpeexRateValid(codec_inst.rate) return IsSpeexRateValid(codec_inst.rate)
? codec_id : kInvalidRate; ? codec_id : kInvalidRate;
} else if (STR_CASE_CMP("celt", codec_inst.plname) == 0) {
return IsCeltRateValid(codec_inst.rate)
? codec_id : kInvalidRate;
} }
return IsRateValid(codec_id, codec_inst.rate) ? return IsRateValid(codec_id, codec_inst.rate) ?
@ -874,15 +871,6 @@ bool ACMCodecDB::IsOpusRateValid(int rate) {
return true; return true;
} }
// Checks if the bitrate is valid for Celt.
bool ACMCodecDB::IsCeltRateValid(int rate) {
if ((rate >= 48000) && (rate <= 128000)) {
return true;
} else {
return false;
}
}
// Checks if the payload type is in the valid range. // Checks if the payload type is in the valid range.
bool ACMCodecDB::ValidPayloadType(int payload_type) { bool ACMCodecDB::ValidPayloadType(int payload_type) {
if ((payload_type < 0) || (payload_type > 127)) { if ((payload_type < 0) || (payload_type > 127)) {

1
webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
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@ -329,7 +329,6 @@ class ACMCodecDB {
static bool IsG7291RateValid(int rate); static bool IsG7291RateValid(int rate);
static bool IsSpeexRateValid(int rate); static bool IsSpeexRateValid(int rate);
static bool IsOpusRateValid(int rate); static bool IsOpusRateValid(int rate);
static bool IsCeltRateValid(int rate);
// Check if the payload type is valid, meaning that it is in the valid range // Check if the payload type is valid, meaning that it is in the valid range
// of 0 to 127. // of 0 to 127.