Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(189)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/media/engine/webrtcvoiceengine_unittest.cc

Issue 2337473002: Multi frequency DTMF support - receiver side (Closed)
Patch Set: rebase Created 4 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/media/engine/webrtcvoiceengine.cc ('k') | webrtc/modules/audio_coding/acm2/acm_codec_database.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 6f0ff6334bd229989c7e7ba181abb3c2dc952b84..1b7b5697aa574081764bd5516bc5748728f5d0be 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -42,11 +42,11 @@ const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1);
-const cricket::AudioCodec kTelephoneEventCodec(106,
- "telephone-event",
- 8000,
- 0,
- 1);
+const cricket::AudioCodec
+ kTelephoneEventCodec1(106, "telephone-event", 8000, 0, 1);
+const cricket::AudioCodec
+ kTelephoneEventCodec2(107, "telephone-event", 32000, 0, 1);
+
const uint32_t kSsrc1 = 0x99;
const uint32_t kSsrc2 = 2;
const uint32_t kSsrc3 = 3;
@@ -235,7 +235,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
SetSend(true);
EXPECT_FALSE(channel_->CanInsertDtmf());
EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
- send_parameters_.codecs.push_back(kTelephoneEventCodec);
+ send_parameters_.codecs.push_back(kTelephoneEventCodec1);
SetSendParameters(send_parameters_);
EXPECT_TRUE(channel_->CanInsertDtmf());
@@ -255,7 +255,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
EXPECT_EQ(-1, telephone_event.payload_type);
EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123));
telephone_event = GetSendStream(kSsrc1).GetLatestTelephoneEvent();
- EXPECT_EQ(kTelephoneEventCodec.id, telephone_event.payload_type);
+ EXPECT_EQ(kTelephoneEventCodec1.id, telephone_event.payload_type);
EXPECT_EQ(2, telephone_event.event_code);
EXPECT_EQ(123, telephone_event.duration_ms);
}
@@ -634,7 +634,10 @@ TEST_F(WebRtcVoiceEngineTestFake, FindCodec) {
// Find ISAC with explicit clockrate and 0 bitrate.
EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kIsacCodec, &codec_inst));
// Find telephone-event with explicit clockrate and 0 bitrate.
- EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec,
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec1,
+ &codec_inst));
+ // Find telephone-event with explicit clockrate and 0 bitrate.
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec2,
&codec_inst));
// Find ISAC with a different payload id.
codec = kIsacCodec;
@@ -667,12 +670,14 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
cricket::AudioRecvParameters parameters;
parameters.codecs.push_back(kIsacCodec);
parameters.codecs.push_back(kPcmuCodec);
- parameters.codecs.push_back(kTelephoneEventCodec);
- parameters.codecs[0].id = 106; // collide with existing telephone-event
+ parameters.codecs.push_back(kTelephoneEventCodec1);
+ parameters.codecs.push_back(kTelephoneEventCodec2);
+ parameters.codecs[0].id = 106; // collide with existing CN 32k
parameters.codecs[2].id = 126;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrc1));
int channel_num = voe_.GetLastChannel();
+
webrtc::CodecInst gcodec;
rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
gcodec.plfreq = 16000;
@@ -680,11 +685,17 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
EXPECT_EQ(106, gcodec.pltype);
EXPECT_STREQ("ISAC", gcodec.plname);
+
rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
gcodec.plfreq = 8000;
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
EXPECT_EQ(126, gcodec.pltype);
EXPECT_STREQ("telephone-event", gcodec.plname);
+
+ gcodec.plfreq = 32000;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+ EXPECT_EQ(107, gcodec.pltype);
+ EXPECT_STREQ("telephone-event", gcodec.plname);
}
// Test that we fail to set an unknown inbound codec.
@@ -776,12 +787,14 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
cricket::AudioRecvParameters parameters;
parameters.codecs.push_back(kIsacCodec);
parameters.codecs.push_back(kPcmuCodec);
- parameters.codecs.push_back(kTelephoneEventCodec);
- parameters.codecs[0].id = 106; // collide with existing telephone-event
+ parameters.codecs.push_back(kTelephoneEventCodec1);
+ parameters.codecs.push_back(kTelephoneEventCodec2);
+ parameters.codecs[0].id = 106; // collide with existing CN 32k
parameters.codecs[2].id = 126;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrc1));
int channel_num2 = voe_.GetLastChannel();
+
webrtc::CodecInst gcodec;
rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
gcodec.plfreq = 16000;
@@ -789,19 +802,25 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
EXPECT_EQ(106, gcodec.pltype);
EXPECT_STREQ("ISAC", gcodec.plname);
+
rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
gcodec.plfreq = 8000;
gcodec.channels = 1;
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
EXPECT_EQ(126, gcodec.pltype);
EXPECT_STREQ("telephone-event", gcodec.plname);
+
+ gcodec.plfreq = 32000;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+ EXPECT_EQ(107, gcodec.pltype);
+ EXPECT_STREQ("telephone-event", gcodec.plname);
}
TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
parameters.codecs.push_back(kIsacCodec);
- parameters.codecs[0].id = 106; // collide with existing telephone-event
+ parameters.codecs[0].id = 106; // collide with existing CN 32k
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
int channel_num2 = voe_.GetLastChannel();
@@ -1849,7 +1868,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) {
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
- parameters.codecs.push_back(kTelephoneEventCodec);
+ parameters.codecs.push_back(kTelephoneEventCodec1);
parameters.codecs.push_back(kIsacCodec);
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs[0].id = 98; // DTMF
@@ -1865,7 +1884,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
- parameters.codecs.push_back(kTelephoneEventCodec);
+ parameters.codecs.push_back(kTelephoneEventCodec1);
parameters.codecs.push_back(kIsacCodec);
parameters.codecs[0].id = 0; // DTMF
parameters.codecs[1].id = 96;
@@ -1909,7 +1928,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
// TODO(juberti): cn 32000
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs.push_back(kCn8000Codec);
- parameters.codecs.push_back(kTelephoneEventCodec);
+ parameters.codecs.push_back(kTelephoneEventCodec1);
parameters.codecs[0].id = 96;
parameters.codecs[2].id = 97; // wideband CN
parameters.codecs[4].id = 98; // DTMF
@@ -1933,7 +1952,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
// TODO(juberti): cn 32000
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs.push_back(kCn8000Codec);
- parameters.codecs.push_back(kTelephoneEventCodec);
+ parameters.codecs.push_back(kTelephoneEventCodec1);
parameters.codecs[0].id = 96;
parameters.codecs[2].id = 97; // wideband CN
parameters.codecs[4].id = 98; // DTMF
@@ -2006,7 +2025,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs.push_back(kCn8000Codec);
- parameters.codecs.push_back(kTelephoneEventCodec);
+ parameters.codecs.push_back(kTelephoneEventCodec1);
parameters.codecs[0].name = "iSaC";
parameters.codecs[0].id = 96;
parameters.codecs[2].id = 97; // wideband CN
@@ -3404,6 +3423,12 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
cricket::AudioCodec(96, "CN", 16000, 0, 1), nullptr));
EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
cricket::AudioCodec(96, "telephone-event", 8000, 0, 1), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "telephone-event", 16000, 0, 1), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "telephone-event", 32000, 0, 1), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "telephone-event", 48000, 0, 1), nullptr));
// Check codecs with an id by id.
EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
cricket::AudioCodec(0, "", 8000, 0, 1), nullptr)); // PCMU
@@ -3433,27 +3458,34 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
// type assignments checked here? It shouldn't really matter.
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory());
- for (std::vector<cricket::AudioCodec>::const_iterator it =
- engine.send_codecs().begin();
- it != engine.send_codecs().end(); ++it) {
- if (it->name == "CN" && it->clockrate == 16000) {
- EXPECT_EQ(105, it->id);
- } else if (it->name == "CN" && it->clockrate == 32000) {
- EXPECT_EQ(106, it->id);
- } else if (it->name == "ISAC" && it->clockrate == 16000) {
- 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 == 8000) {
- EXPECT_EQ(9, it->id);
- } else if (it->name == "telephone-event") {
- EXPECT_EQ(126, it->id);
- } else if (it->name == "opus") {
- EXPECT_EQ(111, it->id);
- ASSERT_TRUE(it->params.find("minptime") != it->params.end());
- EXPECT_EQ("10", it->params.find("minptime")->second);
- ASSERT_TRUE(it->params.find("useinbandfec") != it->params.end());
- EXPECT_EQ("1", it->params.find("useinbandfec")->second);
+ for (const cricket::AudioCodec& codec : engine.send_codecs()) {
+ if (codec.name == "CN" && codec.clockrate == 16000) {
+ EXPECT_EQ(105, codec.id);
+ } else if (codec.name == "CN" && codec.clockrate == 32000) {
+ EXPECT_EQ(106, codec.id);
+ } else if (codec.name == "ISAC" && codec.clockrate == 16000) {
+ EXPECT_EQ(103, codec.id);
+ } else if (codec.name == "ISAC" && codec.clockrate == 32000) {
+ EXPECT_EQ(104, codec.id);
+ } else if (codec.name == "G722" && codec.clockrate == 8000) {
+ EXPECT_EQ(9, codec.id);
+ } else if (codec.name == "telephone-event" && codec.clockrate == 8000) {
+ EXPECT_EQ(126, codec.id);
+ // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned.
+ // Remove these checks once both send and receive side assigns payload types
+ // dynamically.
+ } else if (codec.name == "telephone-event" && codec.clockrate == 16000) {
+ EXPECT_EQ(113, codec.id);
+ } else if (codec.name == "telephone-event" && codec.clockrate == 32000) {
+ EXPECT_EQ(112, codec.id);
+ } else if (codec.name == "telephone-event" && codec.clockrate == 48000) {
+ EXPECT_EQ(110, codec.id);
+ } else if (codec.name == "opus") {
+ EXPECT_EQ(111, codec.id);
+ ASSERT_TRUE(codec.params.find("minptime") != codec.params.end());
+ EXPECT_EQ("10", codec.params.find("minptime")->second);
+ ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end());
+ EXPECT_EQ("1", codec.params.find("useinbandfec")->second);
}
}
}
« no previous file with comments | « webrtc/media/engine/webrtcvoiceengine.cc ('k') | webrtc/modules/audio_coding/acm2/acm_codec_database.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698