Removing redundant argument for SSRCs from ctor of RtpVideoSender.
SSRCs are specified twice in calls to the RtpVideoSender constructor.
Once in the first argument of ssrcs, and then again in the RtpConfig
ssrcs variable. Resolving to reference the variable in the RtpConfig.
Bug: None
TBR: stefan@webrtc.org
Change-Id: I53528140166a53f3558f950d5662b7d3d6b8c822
Reviewed-on: https://webrtc-review.googlesource.com/c/114910
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26094}diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index 47477a9..f667ca3 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -152,7 +152,6 @@
}
RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
- const std::vector<uint32_t>& ssrcs,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
@@ -163,7 +162,7 @@
std::unique_ptr<FecController> fec_controller,
const RtpSenderFrameEncryptionConfig& frame_encryption_config) {
video_rtp_senders_.push_back(absl::make_unique<RtpVideoSender>(
- ssrcs, suspended_ssrcs, states, rtp_config, rtcp_report_interval_ms,
+ suspended_ssrcs, states, rtp_config, rtcp_report_interval_ms,
send_transport, observers,
// TODO(holmer): Remove this circular dependency by injecting
// the parts of RtpTransportControllerSendInterface that are really used.
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 45836a0..d644e74 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -52,7 +52,6 @@
~RtpTransportControllerSend() override;
RtpVideoSenderInterface* CreateRtpVideoSender(
- const std::vector<uint32_t>& ssrcs,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>&
states, // move states into RtpTransportControllerSend
diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index 9868585..19c1dad 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -100,7 +100,6 @@
virtual PacketRouter* packet_router() = 0;
virtual RtpVideoSenderInterface* CreateRtpVideoSender(
- const std::vector<uint32_t>& ssrcs,
std::map<uint32_t, RtpState> suspended_ssrcs,
// TODO(holmer): Move states into RtpTransportControllerSend.
const std::map<uint32_t, RtpPayloadState>& states,
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 9dcc0f6..1bb2439 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -39,7 +39,6 @@
static const size_t kPathMTU = 1500;
std::vector<std::unique_ptr<RtpRtcp>> CreateRtpRtcpModules(
- const std::vector<uint32_t>& ssrcs,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
@@ -59,7 +58,7 @@
RtpKeepAliveConfig keepalive_config,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options) {
- RTC_DCHECK_GT(ssrcs.size(), 0);
+ RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
RtpRtcp::Configuration configuration;
configuration.audio = false;
@@ -91,7 +90,7 @@
std::vector<std::unique_ptr<RtpRtcp>> modules;
const std::vector<uint32_t>& flexfec_protected_ssrcs =
rtp_config.flexfec.protected_media_ssrcs;
- for (uint32_t ssrc : ssrcs) {
+ for (uint32_t ssrc : rtp_config.ssrcs) {
bool enable_flexfec = flexfec_sender != nullptr &&
std::find(flexfec_protected_ssrcs.begin(),
flexfec_protected_ssrcs.end(),
@@ -179,7 +178,6 @@
} // namespace
RtpVideoSender::RtpVideoSender(
- const std::vector<uint32_t>& ssrcs,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
@@ -199,8 +197,7 @@
suspended_ssrcs_(std::move(suspended_ssrcs)),
flexfec_sender_(MaybeCreateFlexfecSender(rtp_config, suspended_ssrcs_)),
fec_controller_(std::move(fec_controller)),
- rtp_modules_(CreateRtpRtcpModules(ssrcs,
- rtp_config,
+ rtp_modules_(CreateRtpRtcpModules(rtp_config,
rtcp_report_interval_ms,
send_transport,
observers.intra_frame_callback,
@@ -224,13 +221,10 @@
transport_overhead_bytes_per_packet_(0),
overhead_bytes_per_packet_(0),
encoder_target_rate_bps_(0) {
- RTC_DCHECK_EQ(ssrcs.size(), rtp_modules_.size());
- // The same argument for SSRCs is given to this method twice.
- // The SSRCs are also accessed in this method through both variables.
- RTC_DCHECK(ssrcs == rtp_config.ssrcs);
+ RTC_DCHECK_EQ(rtp_config.ssrcs.size(), rtp_modules_.size());
module_process_thread_checker_.DetachFromThread();
// SSRCs are assumed to be sorted in the same order as |rtp_modules|.
- for (uint32_t ssrc : ssrcs) {
+ for (uint32_t ssrc : rtp_config.ssrcs) {
// Restore state if it previously existed.
const RtpPayloadState* state = nullptr;
auto it = states.find(ssrc);
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index aff2067..dac43b1 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -49,7 +49,6 @@
public:
// Rtp modules are assumed to be sorted in simulcast index order.
RtpVideoSender(
- const std::vector<uint32_t>& ssrcs,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index d77fe58..958228d 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -92,8 +92,8 @@
config_.rtp.payload_type = payload_type;
std::map<uint32_t, RtpState> suspended_ssrcs;
router_ = absl::make_unique<RtpVideoSender>(
- config_.rtp.ssrcs, suspended_ssrcs, suspended_payload_states,
- config_.rtp, config_.rtcp_report_interval_ms, &transport_,
+ suspended_ssrcs, suspended_payload_states, config_.rtp,
+ config_.rtcp_report_interval_ms, &transport_,
CreateObservers(&call_stats_, &encoder_feedback_, &stats_proxy_,
&stats_proxy_, &stats_proxy_, &stats_proxy_,
&stats_proxy_, &stats_proxy_, &send_delay_stats_),
diff --git a/call/test/mock_rtp_transport_controller_send.h b/call/test/mock_rtp_transport_controller_send.h
index 4d34d06..c236041 100644
--- a/call/test/mock_rtp_transport_controller_send.h
+++ b/call/test/mock_rtp_transport_controller_send.h
@@ -32,10 +32,9 @@
class MockRtpTransportControllerSend
: public RtpTransportControllerSendInterface {
public:
- MOCK_METHOD10(
+ MOCK_METHOD9(
CreateRtpVideoSender,
- RtpVideoSenderInterface*(const std::vector<uint32_t>&,
- std::map<uint32_t, RtpState>,
+ RtpVideoSenderInterface*(std::map<uint32_t, RtpState>,
const std::map<uint32_t, RtpPayloadState>&,
const RtpConfig&,
int rtcp_report_interval_ms,
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index c6d19d0..64ab6e1 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -268,7 +268,6 @@
video_stream_encoder),
bandwidth_observer_(transport->GetBandwidthObserver()),
rtp_video_sender_(transport_->CreateRtpVideoSender(
- config_->rtp.ssrcs,
suspended_ssrcs,
suspended_payload_states,
config_->rtp,
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 32d0df1..2ee375a 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -99,7 +99,7 @@
EXPECT_CALL(transport_controller_, packet_router())
.WillRepeatedly(Return(&packet_router_));
EXPECT_CALL(transport_controller_,
- CreateRtpVideoSender(_, _, _, _, _, _, _, _, _, _))
+ CreateRtpVideoSender(_, _, _, _, _, _, _, _, _))
.WillRepeatedly(Return(&rtp_video_sender_));
EXPECT_CALL(rtp_video_sender_, SetActive(_))
.WillRepeatedly(testing::Invoke(
