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Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc

Browse Source 
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.

The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.

Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
This commit is contained in:
Erik Språng
2019-08-20 17:22:36 +02:00
committed by Commit Bot
parent e8ef87bdad
commit 54d5d2c75b
16 changed files with 47 additions and 39 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
audio/channel_send.cc
View File

@ -701,7 +701,7 @@ ChannelSend::ChannelSend(Clock* clock,
configuration.extmap_allow_mixed = extmap_allow_mixed;
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
configuration.media_send_ssrc = ssrc;
configuration.local_media_ssrc = ssrc;
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);

2
call/flexfec_receive_stream_impl.cc
View File

@ -131,7 +131,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
configuration.receive_statistics = receive_statistics;
configuration.outgoing_transport = config.rtcp_send_transport;
configuration.rtt_stats = rtt_stats;
configuration.media_send_ssrc = config.local_ssrc;
configuration.local_media_ssrc = config.local_ssrc;
return RtpRtcp::Create(configuration);
}

4
call/rtp_video_sender.cc
View File

@ -113,11 +113,11 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
RTC_DCHECK(rtp_config.rtx.ssrcs.empty() ||
rtp_config.rtx.ssrcs.size() == rtp_config.rtx.ssrcs.size());
for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
configuration.media_send_ssrc = rtp_config.ssrcs[i];
configuration.local_media_ssrc = rtp_config.ssrcs[i];
bool enable_flexfec = flexfec_sender != nullptr &&
std::find(flexfec_protected_ssrcs.begin(),
flexfec_protected_ssrcs.end(),
*configuration.media_send_ssrc) !=
*configuration.local_media_ssrc) !=
flexfec_protected_ssrcs.end();
configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
auto playout_delay_oracle = absl::make_unique<PlayoutDelayOracle>();

9
modules/rtp_rtcp/include/rtp_rtcp.h
View File

@ -122,11 +122,18 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
// defaults to webrtc::FieldTrialBasedConfig.
const WebRtcKeyValueConfig* field_trials = nullptr;
// SSRCs for sending media and retransmission, respectively.
// SSRCs for media and retransmission, respectively.
// FlexFec SSRC is fetched from |flexfec_sender|.
// |media_send_ssrc| has been deprecated, use local_media_ssrc instead.
absl::optional<uint32_t> media_send_ssrc;
absl::optional<uint32_t> local_media_ssrc;
absl::optional<uint32_t> rtx_send_ssrc;
// TODO(bugs.webrtc.org/10774): Remove this fallback.
absl::optional<uint32_t> get_local_media_ssrc() const {
return local_media_ssrc ? local_media_ssrc : media_send_ssrc;
}
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
};

2
modules/rtp_rtcp/source/nack_rtx_unittest.cc
View File

@ -135,7 +135,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
configuration.receive_statistics = receive_statistics_.get();
configuration.outgoing_transport = &transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
configuration.media_send_ssrc = kTestSsrc;
configuration.local_media_ssrc = kTestSsrc;
rtp_rtcp_module_ = RtpRtcp::Create(configuration);
rtp_sender_video_ = absl::make_unique<RTPSenderVideo>(
&fake_clock, rtp_rtcp_module_->RtpSender(), nullptr,

7
modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -137,7 +137,8 @@ RTCPReceiver::RTCPReceiver(const RtpRtcp::Configuration& config,
? config.rtcp_report_interval_ms
: (config.audio ? kDefaultAudioReportInterval
: kDefaultVideoReportInterval)),
main_ssrc_(config.media_send_ssrc.value_or(0)),
// TODO(bugs.webrtc.org/10774): Remove fallback.
main_ssrc_(config.get_local_media_ssrc().value_or(0)),
remote_ssrc_(0),
remote_sender_rtp_time_(0),
xr_rrtr_status_(false),
@ -152,8 +153,8 @@ RTCPReceiver::RTCPReceiver(const RtpRtcp::Configuration& config,
num_skipped_packets_(0),
last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
RTC_DCHECK(owner);
if (config.media_send_ssrc) {
registered_ssrcs_.insert(*config.media_send_ssrc);
if (config.get_local_media_ssrc()) {
registered_ssrcs_.insert(*config.get_local_media_ssrc());
}
if (config.rtx_send_ssrc) {
registered_ssrcs_.insert(*config.rtx_send_ssrc);

2
modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
View File

@ -152,7 +152,7 @@ class RtcpReceiverTest : public ::testing::Test {
config.bitrate_allocation_observer =
&bitrate_allocation_observer_;
config.rtcp_report_interval_ms = kRtcpIntervalMs;
config.media_send_ssrc = kReceiverMainSsrc;
config.local_media_ssrc = kReceiverMainSsrc;
config.rtx_send_ssrc = kReceiverExtraSsrc;
return config;
}(),

2
modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -130,7 +130,7 @@ RTCPSender::RTCPSender(const RtpRtcp::Configuration& config)
timestamp_offset_(0),
last_rtp_timestamp_(0),
last_frame_capture_time_ms_(-1),
ssrc_(config.media_send_ssrc.value_or(0)),
ssrc_(config.get_local_media_ssrc().value_or(0)),
remote_ssrc_(0),
receive_statistics_(config.receive_statistics),

10
modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -92,7 +92,7 @@ class RtcpSenderTest : public ::testing::Test {
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
configuration.rtcp_report_interval_ms = 1000;
configuration.receive_statistics = receive_statistics_.get();
configuration.media_send_ssrc = kSenderSsrc;
configuration.local_media_ssrc = kSenderSsrc;
return configuration;
}
@ -195,7 +195,7 @@ TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
config.receive_statistics = receive_statistics_.get();
config.outgoing_transport = &test_transport_;
config.rtcp_report_interval_ms = 1000;
config.media_send_ssrc = kSenderSsrc;
config.local_media_ssrc = kSenderSsrc;
rtcp_sender_.reset(new RTCPSender(config));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
@ -217,7 +217,7 @@ TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
config.receive_statistics = receive_statistics_.get();
config.outgoing_transport = &test_transport_;
config.rtcp_report_interval_ms = 1000;
config.media_send_ssrc = kSenderSsrc;
config.local_media_ssrc = kSenderSsrc;
rtcp_sender_.reset(new RTCPSender(config));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
@ -695,7 +695,7 @@ TEST_F(RtcpSenderTest, ByeMustBeLast) {
config.receive_statistics = receive_statistics_.get();
config.outgoing_transport = &mock_transport;
config.rtcp_report_interval_ms = 1000;
config.media_send_ssrc = kSenderSsrc;
config.local_media_ssrc = kSenderSsrc;
rtcp_sender_.reset(new RTCPSender(config));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
@ -827,7 +827,7 @@ TEST_F(RtcpSenderTest, DoesntSchedulesInitialReportWhenSsrcSetOnConstruction) {
TEST_F(RtcpSenderTest, DoesntSchedulesInitialReportOnFirstSetSsrc) {
// Set up without first SSRC not set at construction.
RtpRtcp::Configuration configuration = GetDefaultConfig();
configuration.media_send_ssrc = absl::nullopt;
configuration.local_media_ssrc = absl::nullopt;
rtcp_sender_.reset(new RTCPSender(configuration));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);

2
modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -162,7 +162,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
config.rtcp_packet_type_counter_observer = this;
config.rtt_stats = &rtt_stats_;
config.rtcp_report_interval_ms = rtcp_report_interval_ms_;
config.media_send_ssrc = kSenderSsrc;
config.local_media_ssrc = kSenderSsrc;
impl_.reset(new ModuleRtpRtcpImpl(config));
impl_->SetRTCPStatus(RtcpMode::kCompound);

2
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -175,7 +175,7 @@ RTPSender::RTPSender(const RtpRtcp::Configuration& config)
bitrate_callback_(config.send_bitrate_observer),
// RTP variables
sequence_number_forced_(false),
ssrc_(config.media_send_ssrc),
ssrc_(config.get_local_media_ssrc()),
ssrc_has_acked_(false),
rtx_ssrc_has_acked_(false),
last_rtp_timestamp_(0),

2
modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
View File

@ -70,7 +70,7 @@ class RtpSenderAudioTest : public ::testing::Test {
config.audio = true;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
return config;
}()),
rtp_sender_audio_(&fake_clock_, &rtp_sender_) {

34
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -245,7 +245,7 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.flexfec_sender = &flexfec_sender_;
config.transport_sequence_number_allocator = &seq_num_allocator_;
@ -426,7 +426,7 @@ TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
config.clock = &fake_clock_;
config.outgoing_transport = &transport;
config.paced_sender = &mock_paced_sender_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.event_log = &mock_rtc_event_log_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_ = absl::make_unique<RTPSender>(config);
@ -476,7 +476,7 @@ TEST_P(RtpSenderTestWithoutPacer,
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.transport_sequence_number_allocator = &seq_num_allocator_;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
@ -515,7 +515,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.transport_sequence_number_allocator = &seq_num_allocator_;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
@ -557,7 +557,7 @@ TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.transport_sequence_number_allocator = &seq_num_allocator_;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
@ -617,7 +617,7 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.send_side_delay_observer = &send_side_delay_observer_;
config.event_log = &mock_rtc_event_log_;
rtp_sender_ = absl::make_unique<RTPSender>(config);
@ -707,7 +707,7 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.paced_sender = &mock_paced_sender_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.transport_sequence_number_allocator = &seq_num_allocator_;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
@ -1244,7 +1244,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.paced_sender = &mock_paced_sender_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_ = absl::make_unique<RTPSender>(config);
@ -1280,7 +1280,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
config.clock = &fake_clock_;
config.outgoing_transport = &transport;
config.paced_sender = &mock_paced_sender_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.event_log = &mock_rtc_event_log_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
@ -1448,7 +1448,7 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.paced_sender = &mock_paced_sender_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.flexfec_sender = &flexfec_sender_;
config.transport_sequence_number_allocator = &seq_num_allocator_;
config.event_log = &mock_rtc_event_log_;
@ -1562,7 +1562,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
rtp_sender_ = absl::make_unique<RTPSender>(config);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetStorePacketsStatus(true, 10);
@ -1723,7 +1723,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.flexfec_sender = &flexfec_sender;
config.transport_sequence_number_allocator = &seq_num_allocator_;
config.event_log = &mock_rtc_event_log_;
@ -1992,7 +1992,7 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.paced_sender = &mock_paced_sender_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.flexfec_sender = &flexfec_sender;
config.transport_sequence_number_allocator = &seq_num_allocator_;
config.event_log = &mock_rtc_event_log_;
@ -2075,7 +2075,7 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.send_bitrate_observer = &callback;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_ = absl::make_unique<RTPSender>(config);
@ -2314,7 +2314,7 @@ TEST_P(RtpSenderTest, OnOverheadChanged) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.overhead_observer = &mock_overhead_observer;
rtp_sender_ = absl::make_unique<RTPSender>(config);
@ -2337,7 +2337,7 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.overhead_observer = &mock_overhead_observer;
rtp_sender_ = absl::make_unique<RTPSender>(config);
@ -2560,7 +2560,7 @@ TEST_P(RtpSenderTest, TrySendPacketUpdatesStats) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.flexfec_sender = &flexfec_sender_;
config.send_side_delay_observer = &send_side_delay_observer;

2
modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
View File

@ -147,7 +147,7 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
config.outgoing_transport = &transport_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.field_trials = &field_trials_;
config.media_send_ssrc = kSsrc;
config.local_media_ssrc = kSsrc;
return config;
}()),
rtp_sender_video_(&fake_clock_, &rtp_sender_, nullptr, field_trials_) {

2
video/end_to_end_tests/bandwidth_tests.cc
View File

@ -201,7 +201,7 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
config.clock = clock_;
config.outgoing_transport = receive_transport_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.media_send_ssrc = (*receive_configs)[0].rtp.local_ssrc;
config.local_media_ssrc = (*receive_configs)[0].rtp.local_ssrc;
rtp_rtcp_ = RtpRtcp::Create(config);
rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);

2
video/rtp_video_stream_receiver.cc
View File

@ -67,7 +67,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
configuration.rtt_stats = rtt_stats;
configuration.rtcp_packet_type_counter_observer =
rtcp_packet_type_counter_observer;
configuration.media_send_ssrc = local_ssrc;
configuration.local_media_ssrc = local_ssrc;
std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);