Returns RTCError for setting unimplemented RtpParameters.
We have a number of RtpParameters that aren't implemented. If a client
is setting these values it creates unexpected results when the value
doesn't do anything for them. This change incorporates returning the
correct error if the parameter is unimplemented.
It also changes the scale_resolution_down_by and scale_framerate_down_by
RtpEncodingParameters to rtc::Optionals because they aren't implemented.
This change is part of the effort to ship get/setParameters in Chrome.
Bug: webrtc:8772
Change-Id: I9797695e5116e6aeb3c02afddbf460b2a0d7d5ab
Reviewed-on: https://webrtc-review.googlesource.com/75421
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23314}diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index b084b01..be0e975 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -15,6 +15,7 @@
#include "api/rtpparameters.h"
#include "media/base/fakemediaengine.h"
#include "media/base/rtpdataengine.h"
+#include "media/base/testutils.h"
#include "media/engine/fakewebrtccall.h"
#include "p2p/base/fakedtlstransport.h"
#include "pc/audiotrack.h"
@@ -162,16 +163,20 @@
void OnAudioSenderDestroyed() { audio_sender_destroyed_signal_fired_ = true; }
+ void CreateVideoRtpSender(uint32_t ssrc) {
+ CreateVideoRtpSender(false, ssrc);
+ }
+
void CreateVideoRtpSender() { CreateVideoRtpSender(false); }
- void CreateVideoRtpSender(bool is_screencast) {
+ void CreateVideoRtpSender(bool is_screencast, uint32_t ssrc = kVideoSsrc) {
AddVideoTrack(is_screencast);
video_rtp_sender_ =
new VideoRtpSender(worker_thread_, local_stream_->GetVideoTracks()[0],
{local_stream_->id()});
video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
- video_rtp_sender_->SetSsrc(kVideoSsrc);
- VerifyVideoChannelInput();
+ video_rtp_sender_->SetSsrc(ssrc);
+ VerifyVideoChannelInput(ssrc);
}
void CreateVideoRtpSenderWithNoTrack() {
@@ -661,6 +666,86 @@
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
+ DestroyAudioRtpSender();
+}
+
+TEST_F(RtpSenderReceiverTest, AudioSenderCantSetUnimplementedRtpParameters) {
+ CreateAudioRtpSender();
+ RtpParameters params = audio_rtp_sender_->GetParameters();
+ EXPECT_EQ(1u, params.encodings.size());
+
+ // Unimplemented RtpParameters: mid, header_extensions,
+ // degredation_preference.
+ params.mid = "dummy_mid";
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.header_extensions.emplace_back();
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
+ params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+
+ DestroyAudioRtpSender();
+}
+
+TEST_F(RtpSenderReceiverTest,
+ AudioSenderCantSetUnimplementedRtpEncodingParameters) {
+ CreateAudioRtpSender();
+ RtpParameters params = audio_rtp_sender_->GetParameters();
+ EXPECT_EQ(1u, params.encodings.size());
+
+ // Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
+ // max_framerate, scale_resolution_down_by, scale_framerate_down_by, rid,
+ // dependency_rids.
+ params.encodings[0].codec_payload_type = 1;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].fec = RtpFecParameters();
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].rtx = RtpRtxParameters();
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].dtx = DtxStatus::ENABLED;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].ptime = 1;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].max_framerate = 1;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].scale_resolution_down_by = 2.0;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].rid = "dummy_rid";
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
+ params = audio_rtp_sender_->GetParameters();
+
+ params.encodings[0].dependency_rids.push_back("dummy_rid");
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ audio_rtp_sender_->SetParameters(params).type());
DestroyAudioRtpSender();
}
@@ -782,6 +867,115 @@
DestroyVideoRtpSender();
}
+TEST_F(RtpSenderReceiverTest, VideoSenderCantSetUnimplementedRtpParameters) {
+ CreateVideoRtpSender();
+ RtpParameters params = video_rtp_sender_->GetParameters();
+ EXPECT_EQ(1u, params.encodings.size());
+
+ // Unimplemented RtpParameters: mid, header_extensions,
+ // degredation_preference.
+ params.mid = "dummy_mid";
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.header_extensions.emplace_back();
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
+ params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+
+ DestroyVideoRtpSender();
+}
+
+TEST_F(RtpSenderReceiverTest,
+ VideoSenderCantSetUnimplementedEncodingParameters) {
+ CreateVideoRtpSender();
+ RtpParameters params = video_rtp_sender_->GetParameters();
+ EXPECT_EQ(1u, params.encodings.size());
+
+ // Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
+ // max_framerate, scale_resolution_down_by, scale_framerate_down_by, rid,
+ // dependency_rids.
+ params.encodings[0].codec_payload_type = 1;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].fec = RtpFecParameters();
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].rtx = RtpRtxParameters();
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].dtx = DtxStatus::ENABLED;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].ptime = 1;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].max_framerate = 1;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].scale_resolution_down_by = 2.0;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].rid = "dummy_rid";
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[0].dependency_rids.push_back("dummy_rid");
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+
+ DestroyVideoRtpSender();
+}
+
+// A video sender can have multiple simulcast layers, in which case it will
+// contain multiple RtpEncodingParameters. This tests that if this is the case
+// (simulcast), then we can't set the bitrate_priority, or max_bitrate_bps
+// for any encodings besides at index 0, because these are both implemented
+// "per-sender."
+TEST_F(RtpSenderReceiverTest, VideoSenderCantSetPerSenderEncodingParameters) {
+ // Add a simulcast specific send stream that contains 2 encoding parameters.
+ std::vector<uint32_t> ssrcs({1, 2});
+ cricket::StreamParams stream_params =
+ cricket::CreateSimStreamParams("cname", ssrcs);
+ video_media_channel_->AddSendStream(stream_params);
+ uint32_t primary_ssrc = stream_params.first_ssrc();
+ CreateVideoRtpSender(primary_ssrc);
+ RtpParameters params = video_rtp_sender_->GetParameters();
+ EXPECT_EQ(ssrcs.size(), params.encodings.size());
+
+ params.encodings[1].bitrate_priority = 2.0;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+ params = video_rtp_sender_->GetParameters();
+
+ params.encodings[1].max_bitrate_bps = 200000;
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+ video_rtp_sender_->SetParameters(params).type());
+
+ DestroyVideoRtpSender();
+}
+
TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) {
CreateVideoRtpSender();
