Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: https://chromium.googlesource.com/external/webrtc/+/1724cfbdbafef4736fedda37312b0286f1eb03d0
TBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 3b562f9..d2b30ca 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -987,10 +987,9 @@
return -1;
}
- return 0;
-}
+ // --- Register all supported codecs to the receiving side of the
+ // RTP/RTCP module
-void Channel::RegisterLegacyCodecs() {
CodecInst codec;
const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
@@ -1042,6 +1041,8 @@
}
}
}
+
+ return 0;
}
void Channel::Terminate() {
@@ -1359,11 +1360,6 @@
return 0;
}
-void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
- rtp_payload_registry_->SetAudioReceivePayloads(codecs);
- audio_coding_->SetReceiveCodecs(codecs);
-}
-
int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
return SetRecPayloadType(codec.pltype, CodecInstToSdp(codec));
}
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 1365ae8..0a12f21 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -151,7 +151,6 @@
uint32_t instanceId,
const VoEBase::ChannelConfig& config);
int32_t Init();
- void RegisterLegacyCodecs();
void Terminate();
int32_t SetEngineInformation(Statistics& engineStatistics,
OutputMixer& outputMixer,
@@ -169,8 +168,6 @@
// go.
const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
- void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
-
// API methods
// VoEBase
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index d67011e..45cbf10 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -173,12 +173,6 @@
RTC_DCHECK_EQ(0, result);
}
-void ChannelProxy::SetReceiveCodecs(
- const std::map<int, SdpAudioFormat>& codecs) {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- channel()->SetReceiveCodecs(codecs);
-}
-
void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetSink(std::move(sink));
@@ -385,11 +379,6 @@
channel()->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate);
}
-void ChannelProxy::RegisterLegacyCodecs() {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- channel()->RegisterLegacyCodecs();
-}
-
Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index b808096..685a168 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -82,7 +82,6 @@
virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
virtual void SetRecPayloadType(int payload_type,
const SdpAudioFormat& format);
- virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
virtual void SetInputMute(bool muted);
virtual void RegisterExternalTransport(Transport* transport);
@@ -120,7 +119,6 @@
virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
virtual void OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate);
- virtual void RegisterLegacyCodecs();
private:
Channel* channel() const;
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index f1eee9c..b15d72f 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -15,8 +15,6 @@
#include "webrtc/base/byteorder.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/voice_engine/channel_proxy.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
namespace {
static const unsigned int kReflectorSsrc = 0x0000;
@@ -64,9 +62,6 @@
EXPECT_EQ(0, local_base_->Init());
local_sender_ = local_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
- ->GetChannelProxy(local_sender_)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
EXPECT_EQ(0, local_rtp_rtcp_->
@@ -77,9 +72,6 @@
EXPECT_EQ(0, remote_base_->Init());
reflector_ = remote_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
- ->GetChannelProxy(reflector_)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
@@ -230,9 +222,6 @@
unsigned int ConferenceTransport::AddStream(std::string file_name,
webrtc::FileFormats format) {
const int new_sender = remote_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
- ->GetChannelProxy(new_sender)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
@@ -246,9 +235,6 @@
new_sender, file_name.c_str(), true, false, format, 1.0));
const int new_receiver = local_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
- ->GetChannelProxy(new_receiver)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc b/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
index 5582124..0fbe635 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
+++ b/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
@@ -16,6 +16,5 @@
webrtc::VoiceEngineImpl* voe_impl =
static_cast<webrtc::VoiceEngineImpl*>(voice_engine_);
channel_proxy_ = voe_impl->GetChannelProxy(channel_);
- channel_proxy_->RegisterLegacyCodecs();
ResumePlaying();
}
diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
index d9a5e2b..a733b12 100644
--- a/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
@@ -8,9 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
class CodecBeforeStreamingTest : public AfterInitializationFixture {
protected:
@@ -21,9 +19,6 @@
codec_instance_.pacsize = 480;
channel_ = voe_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(voice_engine_)
- ->GetChannelProxy(channel_)
- ->RegisterLegacyCodecs();
}
void TearDown() {