| 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);
|
|
|