Index: webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java |
diff --git a/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java b/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java |
index 418b9337f1e11b5de81dcec25144b59577927d8c..8849edd2da1f1ceb97ad849bb5f7d08d66eb6e62 100644 |
--- a/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java |
+++ b/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java |
@@ -716,14 +716,20 @@ public class PeerConnectionTest { |
assertEquals(PeerConnection.SignalingState.STABLE, offeringPC.signalingState()); |
assertEquals(PeerConnection.SignalingState.STABLE, answeringPC.signalingState()); |
- // Set a bitrate limit for the outgoing video stream for the offerer. |
+ // Test some of the RtpSender API. |
RtpSender videoSender = null; |
+ RtpSender audioSender = null; |
for (RtpSender sender : offeringPC.getSenders()) { |
if (sender.track().kind().equals("video")) { |
videoSender = sender; |
+ } else { |
+ audioSender = sender; |
} |
} |
assertNotNull(videoSender); |
+ assertNotNull(audioSender); |
+ |
+ // Set a bitrate limit for the outgoing video stream for the offerer. |
RtpParameters rtpParameters = videoSender.getParameters(); |
assertNotNull(rtpParameters); |
assertEquals(1, rtpParameters.encodings.size()); |
@@ -732,6 +738,12 @@ public class PeerConnectionTest { |
rtpParameters.encodings.get(0).maxBitrateBps = 300000; |
assertTrue(videoSender.setParameters(rtpParameters)); |
+ // Create a DTMF sender. |
+ DtmfSender dtmfSender = audioSender.dtmf(); |
+ assertNotNull(dtmfSender); |
+ assertTrue(dtmfSender.canInsertDtmf()); |
+ assertTrue(dtmfSender.insertDtmf("123", 300, 100)); |
+ |
// Verify that we can read back the updated value. |
rtpParameters = videoSender.getParameters(); |
assertEquals(300000, (int) rtpParameters.encodings.get(0).maxBitrateBps); |