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/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 3777b3d..2d41ecd 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -31,7 +31,6 @@
#include "webrtc/test/gtest.h"
#include "webrtc/voice_engine/transmit_mixer.h"
-using testing::ContainerEq;
using testing::Return;
using testing::StrictMock;
@@ -796,12 +795,26 @@
parameters.codecs[2].id = 126;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {106, {"ISAC", 16000, 1}},
- {126, {"telephone-event", 8000, 1}},
- {107, {"telephone-event", 32000, 1}}})));
+ int channel_num = voe_.GetLastChannel();
+
+ webrtc::CodecInst gcodec;
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
+ gcodec.plfreq = 16000;
+ gcodec.channels = 1;
+ 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.
@@ -832,11 +845,16 @@
parameters.codecs.push_back(kOpusCodec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {103, {"ISAC", 16000, 1}},
- {111, {"opus", 48000, 2}}})));
+ int channel_num = voe_.GetLastChannel();
+ webrtc::CodecInst opus;
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
+ // Even without stereo parameters, recv codecs still specify channels = 2.
+ EXPECT_EQ(2, opus.channels);
+ EXPECT_EQ(111, opus.pltype);
+ EXPECT_STREQ("opus", opus.plname);
+ opus.pltype = 0;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, opus));
+ EXPECT_EQ(111, opus.pltype);
}
// Test that we can decode OPUS with stereo = 0.
@@ -849,11 +867,16 @@
parameters.codecs[2].params["stereo"] = "0";
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {103, {"ISAC", 16000, 1}},
- {111, {"opus", 48000, 2, {{"stereo", "0"}}}}})));
+ int channel_num2 = voe_.GetLastChannel();
+ webrtc::CodecInst opus;
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
+ // Even when stereo is off, recv codecs still specify channels = 2.
+ EXPECT_EQ(2, opus.channels);
+ EXPECT_EQ(111, opus.pltype);
+ EXPECT_STREQ("opus", opus.plname);
+ opus.pltype = 0;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
+ EXPECT_EQ(111, opus.pltype);
}
// Test that we can decode OPUS with stereo = 1.
@@ -866,11 +889,15 @@
parameters.codecs[2].params["stereo"] = "1";
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {103, {"ISAC", 16000, 1}},
- {111, {"opus", 48000, 2, {{"stereo", "1"}}}}})));
+ int channel_num2 = voe_.GetLastChannel();
+ webrtc::CodecInst opus;
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
+ EXPECT_EQ(2, opus.channels);
+ EXPECT_EQ(111, opus.pltype);
+ EXPECT_STREQ("opus", opus.plname);
+ opus.pltype = 0;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
+ EXPECT_EQ(111, opus.pltype);
}
// Test that changes to recv codecs are applied to all streams.
@@ -884,15 +911,28 @@
parameters.codecs[0].id = 106; // collide with existing CN 32k
parameters.codecs[2].id = 126;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
- for (const auto& ssrc : {kSsrcX, kSsrcY}) {
- EXPECT_TRUE(AddRecvStream(ssrc));
- EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {106, {"ISAC", 16000, 1}},
- {126, {"telephone-event", 8000, 1}},
- {107, {"telephone-event", 32000, 1}}})));
- }
+ EXPECT_TRUE(AddRecvStream(kSsrcX));
+ int channel_num2 = voe_.GetLastChannel();
+
+ webrtc::CodecInst gcodec;
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
+ gcodec.plfreq = 16000;
+ gcodec.channels = 1;
+ 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) {
@@ -2921,9 +2961,12 @@
parameters.codecs.push_back(kPcmuCodec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}})));
+ int channel_num2 = voe_.GetLastChannel();
+ webrtc::CodecInst gcodec;
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "opus");
+ gcodec.plfreq = 48000;
+ gcodec.channels = 2;
+ EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec));
}
// Test that we properly clean up any streams that were added, even if