Support for two audio codec lists down into WebRtcVoiceEngine.
Added the plumbing necessary to get two different lists of codecs from
WebRtcVoiceEngine up to MediaSessionDescriptionFactory.
This should be the last step in this set of CLs. Once
https://codereview.webrtc.org/1991233004/ has landed, it's possible to
implement the ReceiveCodecs getter with the info from the
AudioDecoderFactory. The factory needs to be updated to actually
produce the correct list, as well.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2013053002
Cr-Commit-Position: refs/heads/master@{#13131}
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index db5495b..674d54f 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -1336,7 +1336,8 @@
const cricket::AudioCodec kCNCodec2(103, "CN", 16000, 0, 1);
// Add kCNCodec for dtmf test.
- std::vector<cricket::AudioCodec> codecs = media_engine_->audio_codecs();;
+ std::vector<cricket::AudioCodec> codecs =
+ media_engine_->audio_send_codecs();
codecs.push_back(kCNCodec1);
codecs.push_back(kCNCodec2);
media_engine_->SetAudioCodecs(codecs);
diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
index 784967a..b2ec124 100644
--- a/webrtc/media/base/fakemediaengine.h
+++ b/webrtc/media/base/fakemediaengine.h
@@ -767,8 +767,11 @@
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
- const std::vector<AudioCodec>& codecs() { return codecs_; }
- void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
+ // TODO(ossu): For proper testing, These should either individually settable
+ // or the voice engine should reference mockable factories.
+ const std::vector<AudioCodec>& send_codecs() { return codecs_; }
+ const std::vector<AudioCodec>& recv_codecs() { return codecs_; }
+ void SetCodecs(const std::vector<AudioCodec>& codecs) { codecs_ = codecs; }
bool GetOutputVolume(int* level) {
*level = output_volume_;
diff --git a/webrtc/media/base/mediaengine.h b/webrtc/media/base/mediaengine.h
index bd1ce43..9204fc1 100644
--- a/webrtc/media/base/mediaengine.h
+++ b/webrtc/media/base/mediaengine.h
@@ -81,7 +81,8 @@
// Gets the current microphone level, as a value between 0 and 10.
virtual int GetInputLevel() = 0;
- virtual const std::vector<AudioCodec>& audio_codecs() = 0;
+ virtual const std::vector<AudioCodec>& audio_send_codecs() = 0;
+ virtual const std::vector<AudioCodec>& audio_recv_codecs() = 0;
virtual RtpCapabilities GetAudioCapabilities() = 0;
virtual const std::vector<VideoCodec>& video_codecs() = 0;
virtual RtpCapabilities GetVideoCapabilities() = 0;
@@ -163,8 +164,11 @@
virtual int GetInputLevel() {
return voice_.GetInputLevel();
}
- virtual const std::vector<AudioCodec>& audio_codecs() {
- return voice_.codecs();
+ virtual const std::vector<AudioCodec>& audio_send_codecs() {
+ return voice_.send_codecs();
+ }
+ virtual const std::vector<AudioCodec>& audio_recv_codecs() {
+ return voice_.recv_codecs();
}
virtual RtpCapabilities GetAudioCapabilities() {
return voice_.GetCapabilities();
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index b78d73a..fa6e257 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -948,7 +948,12 @@
static_cast<int>(ulevel) : -1;
}
-const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
+const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+ return codecs_;
+}
+
+const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
return codecs_;
}
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 63ef3f3..b7da44f 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -65,7 +65,8 @@
bool SetOutputVolume(int level);
int GetInputLevel();
- const std::vector<AudioCodec>& codecs();
+ const std::vector<AudioCodec>& send_codecs() const;
+ const std::vector<AudioCodec>& recv_codecs() const;
RtpCapabilities GetCapabilities() const;
// For tracking WebRtc channels. Needed because we have to pause them
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 331e625..c6cac1b 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -525,8 +525,10 @@
// Tests that the list of supported codecs is created properly and ordered
// correctly (such that opus appears first).
+// TODO(ossu): This test should move into a separate builtin audio codecs
+// module.
TEST_F(WebRtcVoiceEngineTestFake, CodecOrder) {
- const std::vector<cricket::AudioCodec>& codecs = engine_->codecs();
+ const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs();
ASSERT_FALSE(codecs.empty());
EXPECT_STRCASEEQ("opus", codecs[0].name.c_str());
EXPECT_EQ(48000, codecs[0].clockrate);
@@ -535,7 +537,7 @@
}
TEST_F(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) {
- const std::vector<cricket::AudioCodec>& codecs = engine_->codecs();
+ const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs();
bool opus_found = false;
for (cricket::AudioCodec codec : codecs) {
if (codec.name == "opus") {
@@ -831,7 +833,7 @@
EXPECT_TRUE(SetupChannel());
const int kDesiredBitrate = 128000;
cricket::AudioSendParameters parameters;
- parameters.codecs = engine_->codecs();
+ parameters.codecs = engine_->send_codecs();
parameters.max_bandwidth_bps = kDesiredBitrate;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
@@ -1579,7 +1581,7 @@
EXPECT_FALSE(
call_.GetAudioReceiveStream(kSsrc1)->GetConfig().rtp.transport_cc);
- send_parameters.codecs = engine_->codecs();
+ send_parameters.codecs = engine_->send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(send_parameters));
ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrc1) != nullptr);
EXPECT_TRUE(
@@ -3576,6 +3578,8 @@
}
// Tests that the library is configured with the codecs we want.
+// TODO(ossu): This test should move into the builtin audio codecs module
+// eventually.
TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
// TODO(ossu): These tests should move into a future "builtin audio codecs"
// module.
@@ -3636,7 +3640,7 @@
cricket::WebRtcVoiceEngine engine(nullptr,
webrtc::CreateBuiltinAudioDecoderFactory());
for (std::vector<cricket::AudioCodec>::const_iterator it =
- engine.codecs().begin(); it != engine.codecs().end(); ++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) {
@@ -3701,6 +3705,6 @@
cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
cricket::AudioOptions(), call.get());
cricket::AudioRecvParameters parameters;
- parameters.codecs = engine.codecs();
+ parameters.codecs = engine.recv_codecs();
EXPECT_TRUE(channel.SetRecvParameters(parameters));
}
diff --git a/webrtc/pc/channelmanager.cc b/webrtc/pc/channelmanager.cc
index 334a712..afac930 100644
--- a/webrtc/pc/channelmanager.cc
+++ b/webrtc/pc/channelmanager.cc
@@ -100,15 +100,14 @@
}
}
-void ChannelManager::GetSupportedAudioCodecs(
+void ChannelManager::GetSupportedAudioSendCodecs(
std::vector<AudioCodec>* codecs) const {
- codecs->clear();
+ *codecs = media_engine_->audio_send_codecs();
+}
- for (std::vector<AudioCodec>::const_iterator it =
- media_engine_->audio_codecs().begin();
- it != media_engine_->audio_codecs().end(); ++it) {
- codecs->push_back(*it);
- }
+void ChannelManager::GetSupportedAudioReceiveCodecs(
+ std::vector<AudioCodec>* codecs) const {
+ *codecs = media_engine_->audio_recv_codecs();
}
void ChannelManager::GetSupportedAudioRtpHeaderExtensions(
diff --git a/webrtc/pc/channelmanager.h b/webrtc/pc/channelmanager.h
index 3b0c0e9..16adf87 100644
--- a/webrtc/pc/channelmanager.h
+++ b/webrtc/pc/channelmanager.h
@@ -72,7 +72,8 @@
// Retrieves the list of supported audio & video codec types.
// Can be called before starting the media engine.
- void GetSupportedAudioCodecs(std::vector<AudioCodec>* codecs) const;
+ void GetSupportedAudioSendCodecs(std::vector<AudioCodec>* codecs) const;
+ void GetSupportedAudioReceiveCodecs(std::vector<AudioCodec>* codecs) const;
void GetSupportedAudioRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
void GetSupportedVideoCodecs(std::vector<VideoCodec>* codecs) const;
void GetSupportedVideoRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
diff --git a/webrtc/pc/mediasession.cc b/webrtc/pc/mediasession.cc
index 5d9af98..937a2c1 100644
--- a/webrtc/pc/mediasession.cc
+++ b/webrtc/pc/mediasession.cc
@@ -1273,16 +1273,19 @@
: secure_(SEC_DISABLED),
add_legacy_(true),
transport_desc_factory_(transport_desc_factory) {
- channel_manager->GetSupportedAudioCodecs(&audio_sendrecv_codecs_);
+ channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
+ channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
+ channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_);
channel_manager->GetSupportedVideoCodecs(&video_codecs_);
channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_);
channel_manager->GetSupportedDataCodecs(&data_codecs_);
- audio_send_codecs_ = audio_sendrecv_codecs_;
- audio_recv_codecs_ = audio_sendrecv_codecs_;
+ NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
+ &audio_sendrecv_codecs_);
}
-const AudioCodecs& MediaSessionDescriptionFactory::audio_codecs() const {
+const AudioCodecs& MediaSessionDescriptionFactory::audio_sendrecv_codecs()
+ const {
return audio_sendrecv_codecs_;
}
diff --git a/webrtc/pc/mediasession.h b/webrtc/pc/mediasession.h
index 7a43b00..34354dc 100644
--- a/webrtc/pc/mediasession.h
+++ b/webrtc/pc/mediasession.h
@@ -428,7 +428,7 @@
MediaSessionDescriptionFactory(ChannelManager* cmanager,
const TransportDescriptionFactory* factory);
- const AudioCodecs& audio_codecs() const;
+ const AudioCodecs& audio_sendrecv_codecs() const;
const AudioCodecs& audio_send_codecs() const;
const AudioCodecs& audio_recv_codecs() const;
void set_audio_codecs(const AudioCodecs& send_codecs,
diff --git a/webrtc/pc/mediasession_unittest.cc b/webrtc/pc/mediasession_unittest.cc
index c898315..8ad6526 100644
--- a/webrtc/pc/mediasession_unittest.cc
+++ b/webrtc/pc/mediasession_unittest.cc
@@ -474,7 +474,7 @@
const AudioContentDescription* acd =
static_cast<const AudioContentDescription*>(ac->description);
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
- EXPECT_EQ(f1_.audio_codecs(), acd->codecs());
+ EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
@@ -500,7 +500,7 @@
const VideoContentDescription* vcd =
static_cast<const VideoContentDescription*>(vc->description);
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
- EXPECT_EQ(f1_.audio_codecs(), acd->codecs());
+ EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
@@ -520,7 +520,7 @@
// duplicate RTP payload types.
TEST_F(MediaSessionDescriptionFactoryTest, TestBundleOfferWithSameCodecPlType) {
const VideoCodec& offered_video_codec = f2_.video_codecs()[0];
- const AudioCodec& offered_audio_codec = f2_.audio_codecs()[0];
+ const AudioCodec& offered_audio_codec = f2_.audio_sendrecv_codecs()[0];
const DataCodec& offered_data_codec = f2_.data_codecs()[0];
ASSERT_EQ(offered_video_codec.id, offered_audio_codec.id);
ASSERT_EQ(offered_video_codec.id, offered_data_codec.id);
@@ -607,7 +607,7 @@
const DataContentDescription* dcd =
static_cast<const DataContentDescription*>(dc->description);
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
- EXPECT_EQ(f1_.audio_codecs(), acd->codecs());
+ EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
@@ -1222,7 +1222,7 @@
const DataContentDescription* dcd =
static_cast<const DataContentDescription*>(dc->description);
EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
- EXPECT_EQ(f1_.audio_codecs(), acd->codecs());
+ EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
const StreamParamsVec& audio_streams = acd->streams();
ASSERT_EQ(2U, audio_streams.size());
@@ -2486,25 +2486,25 @@
sf.set_audio_codecs(send_codecs, recv_codecs);
EXPECT_TRUE(sf.audio_send_codecs() == send_codecs);
EXPECT_TRUE(sf.audio_recv_codecs() == recv_codecs);
- EXPECT_TRUE(sf.audio_codecs() == sendrecv_codecs);
+ EXPECT_TRUE(sf.audio_sendrecv_codecs() == sendrecv_codecs);
// Test empty send codecs list
sf.set_audio_codecs(no_codecs, recv_codecs);
EXPECT_TRUE(sf.audio_send_codecs() == no_codecs);
EXPECT_TRUE(sf.audio_recv_codecs() == recv_codecs);
- EXPECT_TRUE(sf.audio_codecs() == no_codecs);
+ EXPECT_TRUE(sf.audio_sendrecv_codecs() == no_codecs);
// Test empty recv codecs list
sf.set_audio_codecs(send_codecs, no_codecs);
EXPECT_TRUE(sf.audio_send_codecs() == send_codecs);
EXPECT_TRUE(sf.audio_recv_codecs() == no_codecs);
- EXPECT_TRUE(sf.audio_codecs() == no_codecs);
+ EXPECT_TRUE(sf.audio_sendrecv_codecs() == no_codecs);
// Test all empty codec lists
sf.set_audio_codecs(no_codecs, no_codecs);
EXPECT_TRUE(sf.audio_send_codecs() == no_codecs);
EXPECT_TRUE(sf.audio_recv_codecs() == no_codecs);
- EXPECT_TRUE(sf.audio_codecs() == no_codecs);
+ EXPECT_TRUE(sf.audio_sendrecv_codecs() == no_codecs);
}
namespace {
