Refactor voe::Channel to not use RtpReceiver.
Analogous to https://webrtc-review.googlesource.com/c/src/+/92398, for
RtpVideoStreamReceiver.
Bug: webrtc:7135
Change-Id: I0639f9982da2ed80edbcf900cf14f8ae982ef80c
Reviewed-on: https://webrtc-review.googlesource.com/93820
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24309}diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 3be50fd..b507afc 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -20,7 +20,6 @@
#include "audio/conversion.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_receiver.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -73,9 +72,9 @@
static_cast<internal::AudioState*>(audio_state);
return absl::make_unique<voe::ChannelProxy>(absl::make_unique<voe::Channel>(
module_process_thread, internal_audio_state->audio_device_module(),
- nullptr /* RtcpRttStats */, event_log, config.jitter_buffer_max_packets,
- config.jitter_buffer_fast_accelerate, config.decoder_factory,
- config.codec_pair_id));
+ nullptr /* RtcpRttStats */, event_log, config.rtp.remote_ssrc,
+ config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate,
+ config.decoder_factory, config.codec_pair_id));
}
} // namespace
@@ -345,9 +344,7 @@
channel_proxy->SetLocalSSRC(new_config.rtp.local_ssrc);
}
- if (first_time) {
- channel_proxy->SetRemoteSSRC(new_config.rtp.remote_ssrc);
- } else {
+ if (!first_time) {
// Remote ssrc can't be changed mid-stream.
RTC_DCHECK_EQ(old_config.rtp.remote_ssrc, new_config.rtp.remote_ssrc);
}
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 0c3189d..eb93bd0 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -83,7 +83,6 @@
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
- EXPECT_CALL(*channel_proxy_, SetRemoteSSRC(kRemoteSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
EXPECT_CALL(*channel_proxy_,
RegisterReceiverCongestionControlObjects(&packet_router_))
diff --git a/audio/channel.cc b/audio/channel.cc
index c96184d..4f9a0f1 100644
--- a/audio/channel.cc
+++ b/audio/channel.cc
@@ -30,7 +30,6 @@
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
-#include "modules/rtp_rtcp/include/rtp_receiver.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "modules/utility/include/process_thread.h"
@@ -344,8 +343,7 @@
}
int64_t round_trip_time = 0;
- _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, NULL, NULL,
- NULL);
+ _rtpRtcpModule->RTT(remote_ssrc_, &round_trip_time, NULL, NULL, NULL);
std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time);
if (!nack_list.empty()) {
@@ -479,6 +477,7 @@
rtcp_rtt_stats,
rtc_event_log,
0,
+ 0,
false,
rtc::scoped_refptr<AudioDecoderFactory>(),
absl::nullopt) {
@@ -490,6 +489,7 @@
AudioDeviceModule* audio_device_module,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
+ uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
@@ -498,10 +498,7 @@
rtp_payload_registry_(new RTPPayloadRegistry()),
rtp_receive_statistics_(
ReceiveStatistics::Create(Clock::GetRealTimeClock())),
- rtp_receiver_(
- RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
- this,
- rtp_payload_registry_.get())),
+ remote_ssrc_(remote_ssrc),
_outputAudioLevel(),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
@@ -561,7 +558,7 @@
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
_rtpRtcpModule->SetSendingMediaStatus(false);
-
+ _rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
Init();
}
@@ -785,6 +782,22 @@
});
}
+std::vector<webrtc::RtpSource> Channel::GetSources() const {
+ int64_t now_ms = rtc::TimeMillis();
+ std::vector<RtpSource> sources;
+ {
+ rtc::CritScope cs(&rtp_sources_lock_);
+ sources = contributing_sources_.GetSources(now_ms);
+ if (last_received_rtp_system_time_ms_ >=
+ now_ms - ContributingSources::kHistoryMs) {
+ sources.emplace_back(*last_received_rtp_system_time_ms_, remote_ssrc_,
+ RtpSourceType::SSRC);
+ sources.back().set_audio_level(last_received_rtp_audio_level_);
+ }
+ }
+ return sources;
+}
+
void Channel::OnUplinkPacketLossRate(float packet_loss_rate) {
if (use_twcc_plr_for_ana_)
return;
@@ -833,7 +846,24 @@
_transportPtr = transport;
}
+// TODO(nisse): Move receive logic up to AudioReceiveStream.
void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
+ int64_t now_ms = rtc::TimeMillis();
+ uint8_t audio_level;
+ bool voice_activity;
+ bool has_audio_level =
+ packet.GetExtension<::webrtc::AudioLevel>(&voice_activity, &audio_level);
+
+ {
+ rtc::CritScope cs(&rtp_sources_lock_);
+ last_received_rtp_timestamp_ = packet.Timestamp();
+ last_received_rtp_system_time_ms_ = now_ms;
+ if (has_audio_level)
+ last_received_rtp_audio_level_ = audio_level;
+ std::vector<uint32_t> csrcs = packet.Csrcs();
+ contributing_sources_.Update(now_ms, csrcs);
+ }
+
RTPHeader header;
packet.GetHeader(&header);
@@ -861,8 +891,16 @@
if (!pl) {
return false;
}
- return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
- pl->typeSpecific);
+ WebRtcRTPHeader webrtc_rtp_header = {};
+ webrtc_rtp_header.header = header;
+
+ const size_t payload_data_length = payload_length - header.paddingLength;
+ if (payload_data_length == 0) {
+ webrtc_rtp_header.frameType = kEmptyFrame;
+ return OnReceivedPayloadData(nullptr, 0, &webrtc_rtp_header);
+ }
+ return OnReceivedPayloadData(payload, payload_data_length,
+ &webrtc_rtp_header);
}
bool Channel::IsPacketRetransmitted(const RTPHeader& header) const {
@@ -989,19 +1027,15 @@
return 0;
}
-void Channel::SetRemoteSSRC(uint32_t ssrc) {
- // Update ssrc so that NTP for AV sync can be updated.
- _rtpRtcpModule->SetRemoteSSRC(ssrc);
-}
-
void Channel::SetMid(const std::string& mid, int extension_id) {
int ret = SetSendRtpHeaderExtension(true, kRtpExtensionMid, extension_id);
RTC_DCHECK_EQ(0, ret);
_rtpRtcpModule->SetMid(mid);
}
+// TODO(nisse): Pass ssrc in return value instead.
int Channel::GetRemoteSSRC(unsigned int& ssrc) {
- ssrc = rtp_receiver_->SSRC();
+ ssrc = remote_ssrc_;
return 0;
}
@@ -1119,7 +1153,7 @@
// each received RTP packet.
RtcpStatistics statistics;
StreamStatistician* statistician =
- rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
+ rtp_receive_statistics_->GetStatistician(remote_ssrc_);
if (statistician) {
// Recompute |fraction_lost| only if RTCP is off. If it's on, then
// |fraction_lost| should only be recomputed when an RTCP SR or RR is sent.
@@ -1317,17 +1351,19 @@
absl::optional<Syncable::Info> Channel::GetSyncInfo() const {
Syncable::Info info;
- if (!rtp_receiver_->GetLatestTimestamps(
- &info.latest_received_capture_timestamp,
- &info.latest_receive_time_ms)) {
- return absl::nullopt;
- }
if (_rtpRtcpModule->RemoteNTP(&info.capture_time_ntp_secs,
&info.capture_time_ntp_frac, nullptr, nullptr,
&info.capture_time_source_clock) != 0) {
return absl::nullopt;
}
-
+ {
+ rtc::CritScope cs(&rtp_sources_lock_);
+ if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
+ return absl::nullopt;
+ }
+ info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
+ info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
+ }
return info;
}
@@ -1408,25 +1444,23 @@
return rtt;
}
- uint32_t remoteSSRC = rtp_receiver_->SSRC();
std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin();
for (; it != report_blocks.end(); ++it) {
- if (it->sender_ssrc == remoteSSRC)
+ if (it->sender_ssrc == remote_ssrc_)
break;
}
- if (it == report_blocks.end()) {
- // We have not received packets with SSRC matching the report blocks.
- // To calculate RTT we try with the SSRC of the first report block.
- // This is very important for send-only channels where we don't know
- // the SSRC of the other end.
- remoteSSRC = report_blocks[0].sender_ssrc;
- }
+
+ // If we have not received packets with SSRC matching the report blocks, use
+ // the SSRC of the first report block for calculating the RTT. This is very
+ // important for send-only channels where we don't know the SSRC of the other
+ // end.
+ uint32_t ssrc =
+ (it == report_blocks.end()) ? report_blocks[0].sender_ssrc : remote_ssrc_;
int64_t avg_rtt = 0;
int64_t max_rtt = 0;
int64_t min_rtt = 0;
- if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
- 0) {
+ if (_rtpRtcpModule->RTT(ssrc, &rtt, &avg_rtt, &min_rtt, &max_rtt) != 0) {
return 0;
}
return rtt;
diff --git a/audio/channel.h b/audio/channel.h
index 670223c..0676dba 100644
--- a/audio/channel.h
+++ b/audio/channel.h
@@ -21,6 +21,7 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
+#include "api/rtpreceiverinterface.h"
#include "audio/audio_level.h"
#include "call/syncable.h"
#include "common_types.h" // NOLINT(build/include)
@@ -28,8 +29,8 @@
#include "modules/audio_processing/rms_level.h"
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "modules/rtp_rtcp/include/rtp_receiver.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/contributing_sources.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/event.h"
#include "rtc_base/task_queue.h"
@@ -156,6 +157,7 @@
AudioDeviceModule* audio_device_module,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
+ uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
@@ -223,7 +225,6 @@
// RTP+RTCP
int SetLocalSSRC(unsigned int ssrc);
- void SetRemoteSSRC(uint32_t ssrc);
void SetMid(const std::string& mid, int extension_id);
int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
@@ -299,9 +300,7 @@
void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
- std::vector<RtpSource> GetSources() const {
- return rtp_receiver_->GetSources();
- }
+ std::vector<RtpSource> GetSources() const;
private:
class ProcessAndEncodeAudioTask;
@@ -343,8 +342,20 @@
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
- std::unique_ptr<RtpReceiver> rtp_receiver_;
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
+ const uint32_t remote_ssrc_;
+
+ // Info for GetSources and GetSyncInfo is updated on network or worker thread,
+ // queried on the worker thread.
+ rtc::CriticalSection rtp_sources_lock_;
+ ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_);
+ absl::optional<uint32_t> last_received_rtp_timestamp_
+ RTC_GUARDED_BY(&rtp_sources_lock_);
+ absl::optional<int64_t> last_received_rtp_system_time_ms_
+ RTC_GUARDED_BY(&rtp_sources_lock_);
+ absl::optional<uint8_t> last_received_rtp_audio_level_
+ RTC_GUARDED_BY(&rtp_sources_lock_);
+
std::unique_ptr<AudioCodingModule> audio_coding_;
AudioSinkInterface* audio_sink_ = nullptr;
AudioLevel _outputAudioLevel;
diff --git a/audio/channel_proxy.cc b/audio/channel_proxy.cc
index 78cc2fc..5e8181a 100644
--- a/audio/channel_proxy.cc
+++ b/audio/channel_proxy.cc
@@ -53,11 +53,6 @@
RTC_DCHECK_EQ(0, error);
}
-void ChannelProxy::SetRemoteSSRC(uint32_t ssrc) {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- channel_->SetRemoteSSRC(ssrc);
-}
-
void ChannelProxy::SetMid(const std::string& mid, int extension_id) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel_->SetMid(mid, extension_id);
diff --git a/audio/channel_proxy.h b/audio/channel_proxy.h
index 253a196..f82c1fd 100644
--- a/audio/channel_proxy.h
+++ b/audio/channel_proxy.h
@@ -34,7 +34,6 @@
class RtcpRttStats;
class RtpPacketSender;
class RtpPacketReceived;
-class RtpReceiver;
class RtpRtcp;
class RtpTransportControllerSendInterface;
class Transport;
@@ -62,7 +61,6 @@
virtual void SetRTCPStatus(bool enable);
virtual void SetLocalSSRC(uint32_t ssrc);
- virtual void SetRemoteSSRC(uint32_t ssrc);
virtual void SetMid(const std::string& mid, int extension_id);
virtual void SetRTCP_CNAME(const std::string& c_name);
virtual void SetNACKStatus(bool enable, int max_packets);
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index ee16195..f6a2637 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -37,7 +37,6 @@
void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
MOCK_METHOD1(SetRTCPStatus, void(bool enable));
MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
- MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
diff --git a/modules/rtp_rtcp/source/contributing_sources.cc b/modules/rtp_rtcp/source/contributing_sources.cc
index 9b82de5..853706c 100644
--- a/modules/rtp_rtcp/source/contributing_sources.cc
+++ b/modules/rtp_rtcp/source/contributing_sources.cc
@@ -14,15 +14,13 @@
namespace {
-// Set by the spec, see
-// https://www.w3.org/TR/webrtc/#dom-rtcrtpreceiver-getcontributingsources
-constexpr int64_t kHistoryMs = 10 * rtc::kNumMillisecsPerSec;
-
// Allow some stale records to accumulate before cleaning.
constexpr int64_t kPruningIntervalMs = 15 * rtc::kNumMillisecsPerSec;
} // namespace
+constexpr int64_t ContributingSources::kHistoryMs;
+
ContributingSources::ContributingSources() = default;
ContributingSources::~ContributingSources() = default;
diff --git a/modules/rtp_rtcp/source/contributing_sources.h b/modules/rtp_rtcp/source/contributing_sources.h
index 64d7f52..1a4a572 100644
--- a/modules/rtp_rtcp/source/contributing_sources.h
+++ b/modules/rtp_rtcp/source/contributing_sources.h
@@ -24,6 +24,10 @@
class ContributingSources {
public:
+ // Set by the spec, see
+ // https://www.w3.org/TR/webrtc/#dom-rtcrtpreceiver-getcontributingsources
+ static constexpr int64_t kHistoryMs = 10 * rtc::kNumMillisecsPerSec;
+
ContributingSources();
~ContributingSources();
