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Add SSRC to RtpEncodingParameters for audio.

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Was added for video initially, but not for audio.

BUG=webrtc:6862

Review-Url: https://codereview.webrtc.org/2568553002
Cr-Commit-Position: refs/heads/master@{#15552}
This commit is contained in:
deadbeef 2016-12-12 11:12:36 -08:00 committed by Commit bot
parent ccecdd4560
commit cb44343006
2 changed files with 19 additions and 0 deletions
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2
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1217,6 +1217,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
config_.voe_channel_id = ch;
config_.rtp.extensions = extensions;
config_.audio_network_adaptor_config = audio_network_adaptor_config;
rtp_parameters_.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
RecreateAudioSendStream(send_codec_spec);
}
@ -1742,6 +1743,7 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
for (const AudioCodec& codec : recv_codecs_) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
rtp_params.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
return rtp_params;
}

17
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -1113,6 +1113,14 @@ TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
}
// Test that GetRtpSendParameters returns an SSRC.
TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
EXPECT_TRUE(SetupSendStream());
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc1);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_EQ(kSsrc1, rtp_parameters.encodings[0].ssrc);
}
// Test that if we set/get parameters multiple times, we get the same results.
TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
EXPECT_TRUE(SetupSendStream());
@ -1146,6 +1154,15 @@ TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) {
EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
}
// Test that GetRtpReceiveParameters returns an SSRC.
TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) {
EXPECT_TRUE(SetupRecvStream());
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpReceiveParameters(kSsrc1);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_EQ(kSsrc1, rtp_parameters.encodings[0].ssrc);
}
// Test that if we set/get parameters multiple times, we get the same results.
TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) {
EXPECT_TRUE(SetupRecvStream());