Exposing rtcp report interval setting in objc api
Bug: webrtc:8789
Change-Id: I75d8cac70de00b067cbbcbe7faa3d3ccb0318453
Reviewed-on: https://webrtc-review.googlesource.com/c/110846
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25643}diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.h b/sdk/objc/api/peerconnection/RTCConfiguration.h
index f140081..f9e6edf 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -201,6 +201,16 @@
*/
@property(nonatomic, nullable) RTCCryptoOptions *cryptoOptions;
+/**
+ * Time interval between audio RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
+
+/**
+ * Time interval between video RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
+
- (instancetype)init;
@end
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.mm b/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 55bf26f..4041f83 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -53,6 +53,8 @@
@synthesize useMediaTransport = _useMediaTransport;
@synthesize useMediaTransportForDataChannels = _useMediaTransportForDataChannels;
@synthesize cryptoOptions = _cryptoOptions;
+@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
+@synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
- (instancetype)init {
// Copy defaults.
@@ -130,6 +132,8 @@
sframeRequireFrameEncryption:config.crypto_options->sframe
.require_frame_encryption];
}
+ _rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
+ _rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
}
return self;
}
@@ -261,7 +265,8 @@
_cryptoOptions.sframeRequireFrameEncryption ? true : false;
nativeConfig->crypto_options = absl::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
}
-
+ nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
+ nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
return nativeConfig.release();
}
diff --git a/sdk/objc/unittests/RTCConfigurationTest.mm b/sdk/objc/unittests/RTCConfigurationTest.mm
index 67a303e..f31fcfd 100644
--- a/sdk/objc/unittests/RTCConfigurationTest.mm
+++ b/sdk/objc/unittests/RTCConfigurationTest.mm
@@ -54,6 +54,8 @@
srtpEnableAes128Sha1_32CryptoCipher:YES
srtpEnableEncryptedRtpHeaderExtensions:YES
sframeRequireFrameEncryption:YES];
+ config.rtcpAudioReportIntervalMs = 2500;
+ config.rtcpVideoReportIntervalMs = 3750;
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
nativeConfig([config createNativeConfiguration]);
@@ -86,6 +88,8 @@
EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_aes128_sha1_32_crypto_cipher);
EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_encrypted_rtp_header_extensions);
EXPECT_EQ(true, nativeConfig->crypto_options->sframe.require_frame_encryption);
+ EXPECT_EQ(2500, nativeConfig->audio_rtcp_report_interval_ms());
+ EXPECT_EQ(3750, nativeConfig->video_rtcp_report_interval_ms());
}
- (void)testNativeConversionToConfiguration {
@@ -115,6 +119,8 @@
srtpEnableAes128Sha1_32CryptoCipher:NO
srtpEnableEncryptedRtpHeaderExtensions:NO
sframeRequireFrameEncryption:NO];
+ config.rtcpAudioReportIntervalMs = 1500;
+ config.rtcpVideoReportIntervalMs = 2150;
webrtc::PeerConnectionInterface::RTCConfiguration *nativeConfig =
[config createNativeConfiguration];
@@ -150,6 +156,8 @@
newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
newConfig.cryptoOptions.sframeRequireFrameEncryption);
+ EXPECT_EQ(config.rtcpAudioReportIntervalMs, newConfig.rtcpAudioReportIntervalMs);
+ EXPECT_EQ(config.rtcpVideoReportIntervalMs, newConfig.rtcpVideoReportIntervalMs);
}
- (void)testDefaultValues {