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Add video send SSRC to RtpParameters, and don't allow changing SSRC.

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With this, RtpSender and RtpReceiver will always return an SSRC if one
is available. Also, attempts to change the SSRC with SetParameters will
fail, rather than silently doing nothing.

BUG=webrtc:6888

Review-Url: https://codereview.webrtc.org/2567333004
Cr-Commit-Position: refs/heads/master@{#15939}
This commit is contained in:
deadbeef 2017-01-06 23:05:37 -08:00 committed by Commit bot
parent 0aeaa91219
commit fb2aceded6
4 changed files with 62 additions and 30 deletions
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10
webrtc/media/engine/webrtcvideoengine2.cc
View File

@ -1560,6 +1560,11 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
// ValidateStreamParams should prevent this from happening.
RTC_CHECK(!parameters_.config.rtp.ssrcs.empty());
rtp_parameters_.encodings[0].ssrc =
rtc::Optional<uint32_t>(parameters_.config.rtp.ssrcs[0]);
// RTX.
sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
&parameters_.config.rtp.rtx.ssrcs);
@ -1862,11 +1867,16 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::GetRtpParameters() const {
bool WebRtcVideoChannel2::WebRtcVideoSendStream::ValidateRtpParameters(
const webrtc::RtpParameters& rtp_parameters) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (rtp_parameters.encodings.size() != 1) {
LOG(LS_ERROR)
<< "Attempted to set RtpParameters without exactly one encoding";
return false;
}
if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
return false;
}
return true;
}

31
webrtc/media/engine/webrtcvideoengine2_unittest.cc
View File

@ -3661,16 +3661,21 @@ TEST_F(WebRtcVideoChannel2Test,
// for each encoding individually.
AddSendStream();
// Setting RtpParameters with no encoding is expected to fail.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings.clear();
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
// Setting RtpParameters with exactly one encoding should succeed.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
// Two or more encodings should result in failure.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
// Zero encodings should also fail.
parameters.encodings.clear();
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
}
// Changing the SSRC through RtpParameters is not allowed.
TEST_F(WebRtcVideoChannel2Test, CannotSetSsrcInRtpSendParameters) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings[0].ssrc = rtc::Optional<uint32_t>(0xdeadbeef);
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
}
// Test that a stream will not be sending if its encoding is made inactive
@ -3753,6 +3758,18 @@ TEST_F(WebRtcVideoChannel2Test, GetRtpSendParametersCodecs) {
rtp_parameters.codecs[1]);
}
// Test that RtpParameters for send stream has one encoding and it has
// the correct SSRC.
TEST_F(WebRtcVideoChannel2Test, GetRtpSendParametersSsrc) {
AddSendStream();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_EQ(rtc::Optional<uint32_t>(last_ssrc_),
rtp_parameters.encodings[0].ssrc);
}
// Test that if we set/get parameters multiple times, we get the same results.
TEST_F(WebRtcVideoChannel2Test, SetAndGetRtpSendParameters) {
AddSendStream();
@ -3826,7 +3843,7 @@ TEST_F(WebRtcVideoChannel2Test, DISABLED_GetRtpReceiveFmtpSprop) {
// Test that RtpParameters for receive stream has one encoding and it has
// the correct SSRC.
TEST_F(WebRtcVideoChannel2Test, RtpEncodingParametersSsrcIsSet) {
TEST_F(WebRtcVideoChannel2Test, GetRtpReceiveParametersSsrc) {
AddRecvStream();
webrtc::RtpParameters rtp_parameters =

33
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1384,8 +1384,23 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
return rtp_parameters_;
}
bool ValidateRtpParameters(const webrtc::RtpParameters& rtp_parameters) {
if (rtp_parameters.encodings.size() != 1) {
LOG(LS_ERROR)
<< "Attempted to set RtpParameters without exactly one encoding";
return false;
}
if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
return false;
}
return true;
}
bool SetRtpParameters(const webrtc::RtpParameters& parameters) {
RTC_CHECK_EQ(1UL, parameters.encodings.size());
if (!ValidateRtpParameters(parameters)) {
return false;
}
auto send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
parameters.encodings[0].max_bitrate_bps,
send_codec_spec_.codec_inst);
@ -1704,9 +1719,6 @@ bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!ValidateRtpParameters(parameters)) {
return false;
}
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
@ -1760,9 +1772,6 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!ValidateRtpParameters(parameters)) {
return false;
}
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
@ -1779,16 +1788,6 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
return true;
}
bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
const webrtc::RtpParameters& rtp_parameters) {
if (rtp_parameters.encodings.size() != 1) {
LOG(LS_ERROR)
<< "Attempted to set RtpParameters without exactly one encoding";
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "Setting voice channel options: "

18
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -1039,15 +1039,21 @@ TEST_F(WebRtcVoiceEngineTestFake,
// for each encoding individually.
EXPECT_TRUE(SetupSendStream());
// Setting RtpParameters with no encoding is expected to fail.
webrtc::RtpParameters parameters;
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
// Setting RtpParameters with exactly one encoding should succeed.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters));
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrc1);
// Two or more encodings should result in failure.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
// Zero encodings should also fail.
parameters.encodings.clear();
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
}
// Changing the SSRC through RtpParameters is not allowed.
TEST_F(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) {
EXPECT_TRUE(SetupSendStream());
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrc1);
parameters.encodings[0].ssrc = rtc::Optional<uint32_t>(0xdeadbeef);
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
}
// Test that a stream will not be sending if its encoding is made