Delete class SSRCDatabase, and its global ssrc registry,
and the method RTPSender::GenerateNewSSRC.
It's now mandatory for higher layers to call SetSSRC, RTPSender
no longer allocates any ssrc by default.
BUG=webrtc:4306,webrtc:6887
Review-Url: https://codereview.webrtc.org/2644303002
Cr-Original-Commit-Position: refs/heads/master@{#16670}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: b78d4d13835f628e722a57abae2bf06ba3655921
diff --git a/logging/rtc_event_log/rtc_event_log.proto b/logging/rtc_event_log/rtc_event_log.proto
index 0da910a..b1ece47 100644
--- a/logging/rtc_event_log/rtc_event_log.proto
+++ b/logging/rtc_event_log/rtc_event_log.proto
@@ -102,6 +102,8 @@
}
message AudioPlayoutEvent {
+ // TODO(ivoc): Rename, we currently use the "remote" ssrc, i.e. identifying
+ // the receive stream, while local_ssrc identifies the send stream, if any.
// required - The SSRC of the audio stream associated with the playout event.
optional uint32 local_ssrc = 2;
}
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index da742c5..f2e53bb 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -143,8 +143,6 @@
"source/rtp_sender_video.h",
"source/rtp_utility.cc",
"source/rtp_utility.h",
- "source/ssrc_database.cc",
- "source/ssrc_database.h",
"source/time_util.cc",
"source/time_util.h",
"source/tmmbr_help.cc",
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index f74ebba..482a741 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -100,7 +100,6 @@
this),
clock_(configuration.clock),
audio_(configuration.audio),
- collision_detected_(false),
last_process_time_(configuration.clock->TimeInMilliseconds()),
last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
@@ -112,11 +111,6 @@
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
- // Make sure that RTCP objects are aware of our SSRC.
- uint32_t SSRC = rtp_sender_.SSRC();
- rtcp_sender_.SetSSRC(SSRC);
- SetRtcpReceiverSsrcs(SSRC);
-
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_.TimestampOffset());
@@ -355,19 +349,6 @@
if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) {
LOG(LS_WARNING) << "Failed to send RTCP BYE";
}
-
- collision_detected_ = false;
-
- // Generate a new SSRC for the next "call" if false
- rtp_sender_.SetSendingStatus(sending);
-
- // Make sure that RTCP objects are aware of our SSRC (it could have changed
- // Due to collision)
- uint32_t SSRC = rtp_sender_.SSRC();
- rtcp_sender_.SetSSRC(SSRC);
- SetRtcpReceiverSsrcs(SSRC);
-
- return 0;
}
return 0;
}
@@ -794,24 +775,6 @@
// Inform about the incoming SSRC.
rtcp_sender_.SetRemoteSSRC(ssrc);
rtcp_receiver_.SetRemoteSSRC(ssrc);
-
- // Check for a SSRC collision.
- if (rtp_sender_.SSRC() == ssrc && !collision_detected_) {
- // If we detect a collision change the SSRC but only once.
- collision_detected_ = true;
- uint32_t new_ssrc = rtp_sender_.GenerateNewSSRC();
- if (new_ssrc == 0) {
- // Configured via API ignore.
- return;
- }
- if (RtcpMode::kOff != rtcp_sender_.Status()) {
- // Send RTCP bye on the current SSRC.
- SendRTCP(kRtcpBye);
- }
- // Change local SSRC and inform all objects about the new SSRC.
- rtcp_sender_.SetSSRC(new_ssrc);
- SetRtcpReceiverSsrcs(new_ssrc);
- }
}
void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index f1af359..c15515d 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -333,7 +333,6 @@
bool TimeToSendFullNackList(int64_t now) const;
const bool audio_;
- bool collision_detected_;
int64_t last_process_time_;
int64_t last_bitrate_process_time_;
int64_t last_rtt_process_time_;
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 47ec31b..7283663 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -111,10 +111,8 @@
send_packet_observer_(send_packet_observer),
bitrate_callback_(bitrate_callback),
// RTP variables
- ssrc_db_(SSRCDatabase::GetSSRCDatabase()),
remote_ssrc_(0),
sequence_number_forced_(false),
- ssrc_forced_(false),
last_rtp_timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
@@ -128,11 +126,6 @@
send_side_bwe_with_overhead_(
webrtc::field_trial::FindFullName(
"WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
- ssrc_ = ssrc_db_->CreateSSRC();
- RTC_DCHECK(ssrc_ != 0);
- ssrc_rtx_ = ssrc_db_->CreateSSRC();
- RTC_DCHECK(ssrc_rtx_ != 0);
-
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
@@ -157,12 +150,6 @@
// variables but we grab them in all other methods. (what's the design?)
// Start documenting what thread we're on in what method so that it's easier
// to understand performance attributes and possibly remove locks.
- if (remote_ssrc_ != 0) {
- ssrc_db_->ReturnSSRC(remote_ssrc_);
- }
- ssrc_db_->ReturnSSRC(ssrc_);
-
- SSRCDatabase::ReturnSSRCDatabase();
while (!payload_type_map_.empty()) {
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.begin();
@@ -210,7 +197,7 @@
return rtp_header_extension_map_.Register(type, id);
case kRtpExtensionNone:
case kRtpExtensionNumberOfExtensions:
- LOG(LS_ERROR) << "Invalid RTP extension type for registration";
+ LOG(LS_ERROR) << "Invalid RTP extension type for registration.";
return -1;
}
return -1;
@@ -334,12 +321,13 @@
void RTPSender::SetRtxSsrc(uint32_t ssrc) {
rtc::CritScope lock(&send_critsect_);
- ssrc_rtx_ = ssrc;
+ ssrc_rtx_.emplace(ssrc);
}
uint32_t RTPSender::RtxSsrc() const {
rtc::CritScope lock(&send_critsect_);
- return ssrc_rtx_;
+ RTC_DCHECK(ssrc_rtx_);
+ return *ssrc_rtx_;
}
void RTPSender::SetRtxPayloadType(int payload_type,
@@ -348,7 +336,7 @@
RTC_DCHECK_LE(payload_type, 127);
RTC_DCHECK_LE(associated_payload_type, 127);
if (payload_type < 0) {
- LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
+ LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type << ".";
return;
}
@@ -360,7 +348,7 @@
rtc::CritScope lock(&send_critsect_);
if (payload_type < 0) {
- LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
+ LOG(LS_ERROR) << "Invalid payload_type " << payload_type << ".";
return -1;
}
if (payload_type_ == payload_type) {
@@ -401,7 +389,9 @@
{
// Drop this packet if we're not sending media packets.
rtc::CritScope lock(&send_critsect_);
- ssrc = ssrc_;
+ RTC_DCHECK(ssrc_);
+
+ ssrc = *ssrc_;
sequence_number = sequence_number_;
rtp_timestamp = timestamp_offset_ + capture_timestamp;
if (transport_frame_id_out)
@@ -521,7 +511,14 @@
if (!audio_configured_ && !last_packet_marker_bit_) {
break;
}
- ssrc = ssrc_;
+ if (!ssrc_) {
+ LOG(LS_ERROR) << "SSRC unset.";
+ return 0;
+ }
+
+ RTC_DCHECK(ssrc_);
+ ssrc = *ssrc_;
+
sequence_number = sequence_number_;
++sequence_number_;
payload_type = payload_type_;
@@ -545,7 +542,12 @@
(now_ms - last_timestamp_time_ms_) * kTimestampTicksPerMs;
capture_time_ms += (now_ms - last_timestamp_time_ms_);
}
- ssrc = ssrc_rtx_;
+ if (!ssrc_rtx_) {
+ LOG(LS_ERROR) << "RTX SSRC unset.";
+ return 0;
+ }
+ RTC_DCHECK(ssrc_rtx_);
+ ssrc = *ssrc_rtx_;
sequence_number = sequence_number_rtx_;
++sequence_number_rtx_;
payload_type = rtx_payload_type_map_.begin()->second;
@@ -645,7 +647,7 @@
"sent", bytes_sent);
// TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
if (bytes_sent <= 0) {
- LOG(LS_WARNING) << "Transport failed to send packet";
+ LOG(LS_WARNING) << "Transport failed to send packet.";
return false;
}
return true;
@@ -675,7 +677,7 @@
if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
- << ", Discard rest of packets";
+ << ", Discard rest of packets.";
break;
}
}
@@ -919,7 +921,9 @@
int max_delay_ms = 0;
{
rtc::CritScope lock(&send_critsect_);
- ssrc = ssrc_;
+ if (!ssrc_)
+ return;
+ ssrc = *ssrc_;
}
{
rtc::CritScope cs(&statistics_crit_);
@@ -959,7 +963,9 @@
uint32_t ssrc;
{
rtc::CritScope lock(&send_critsect_);
- ssrc = ssrc_;
+ if (!ssrc_)
+ return;
+ ssrc = *ssrc_;
}
rtc::CritScope lock(&statistics_crit_);
@@ -993,7 +999,8 @@
rtc::CritScope lock(&send_critsect_);
std::unique_ptr<RtpPacketToSend> packet(
new RtpPacketToSend(&rtp_header_extension_map_, max_packet_size_));
- packet->SetSsrc(ssrc_);
+ RTC_DCHECK(ssrc_);
+ packet->SetSsrc(*ssrc_);
packet->SetCsrcs(csrcs_);
// Reserve extensions, if registered, RtpSender set in SendToNetwork.
packet->ReserveExtension<AbsoluteSendTime>();
@@ -1010,7 +1017,7 @@
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return false;
- RTC_DCHECK_EQ(packet->Ssrc(), ssrc_);
+ RTC_DCHECK(packet->Ssrc() == ssrc_);
packet->SetSequenceNumber(sequence_number_++);
// Remember marker bit to determine if padding can be inserted with
@@ -1042,23 +1049,6 @@
return true;
}
-void RTPSender::SetSendingStatus(bool enabled) {
- if (!enabled) {
- rtc::CritScope lock(&send_critsect_);
- if (!ssrc_forced_) {
- // Generate a new SSRC.
- ssrc_db_->ReturnSSRC(ssrc_);
- ssrc_ = ssrc_db_->CreateSSRC();
- RTC_DCHECK(ssrc_ != 0);
- }
- // Don't initialize seq number if SSRC passed externally.
- if (!sequence_number_forced_ && !ssrc_forced_) {
- // Generate a new sequence number.
- sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
- }
- }
-}
-
void RTPSender::SetSendingMediaStatus(bool enabled) {
rtc::CritScope lock(&send_critsect_);
sending_media_ = enabled;
@@ -1079,29 +1069,14 @@
return timestamp_offset_;
}
-uint32_t RTPSender::GenerateNewSSRC() {
- // If configured via API, return 0.
- rtc::CritScope lock(&send_critsect_);
-
- if (ssrc_forced_) {
- return 0;
- }
- ssrc_ = ssrc_db_->CreateSSRC();
- RTC_DCHECK(ssrc_ != 0);
- return ssrc_;
-}
-
void RTPSender::SetSSRC(uint32_t ssrc) {
// This is configured via the API.
rtc::CritScope lock(&send_critsect_);
- if (ssrc_ == ssrc && ssrc_forced_) {
+ if (ssrc_ == ssrc) {
return; // Since it's same ssrc, don't reset anything.
}
- ssrc_forced_ = true;
- ssrc_db_->ReturnSSRC(ssrc_);
- ssrc_db_->RegisterSSRC(ssrc);
- ssrc_ = ssrc;
+ ssrc_.emplace(ssrc);
if (!sequence_number_forced_) {
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
@@ -1109,7 +1084,8 @@
uint32_t RTPSender::SSRC() const {
rtc::CritScope lock(&send_critsect_);
- return ssrc_;
+ RTC_DCHECK(ssrc_);
+ return *ssrc_;
}
rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const {
@@ -1189,6 +1165,8 @@
if (!sending_media_)
return nullptr;
+ RTC_DCHECK(ssrc_rtx_);
+
// Replace payload type.
auto kv = rtx_payload_type_map_.find(packet.PayloadType());
if (kv == rtx_payload_type_map_.end())
@@ -1199,7 +1177,7 @@
rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
// Replace SSRC.
- rtx_packet->SetSsrc(ssrc_rtx_);
+ rtx_packet->SetSsrc(*ssrc_rtx_);
}
uint8_t* rtx_payload =
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 09884b3..8f71b48 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -32,7 +32,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
namespace webrtc {
@@ -87,8 +86,6 @@
int8_t SendPayloadType() const;
- void SetSendingStatus(bool enabled);
-
void SetSendingMediaStatus(bool enabled);
bool SendingMedia() const;
@@ -98,7 +95,6 @@
uint32_t TimestampOffset() const;
void SetTimestampOffset(uint32_t timestamp);
- uint32_t GenerateNewSSRC();
void SetSSRC(uint32_t ssrc);
uint16_t SequenceNumber() const;
@@ -305,13 +301,13 @@
// RTP variables
uint32_t timestamp_offset_ GUARDED_BY(send_critsect_);
- SSRCDatabase* const ssrc_db_;
uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
bool sequence_number_forced_ GUARDED_BY(send_critsect_);
uint16_t sequence_number_ GUARDED_BY(send_critsect_);
uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
- bool ssrc_forced_ GUARDED_BY(send_critsect_);
- uint32_t ssrc_ GUARDED_BY(send_critsect_);
+ // Must be explicitly set by the application, use of rtc::Optional
+ // only to keep track of correct use.
+ rtc::Optional<uint32_t> ssrc_ GUARDED_BY(send_critsect_);
uint32_t last_rtp_timestamp_ GUARDED_BY(send_critsect_);
int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
@@ -319,7 +315,7 @@
bool last_packet_marker_bit_ GUARDED_BY(send_critsect_);
std::vector<uint32_t> csrcs_ GUARDED_BY(send_critsect_);
int rtx_ GUARDED_BY(send_critsect_);
- uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
+ rtc::Optional<uint32_t> ssrc_rtx_ GUARDED_BY(send_critsect_);
// Mapping rtx_payload_type_map_[associated] = rtx.
std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
size_t rtp_overhead_bytes_per_packet_ GUARDED_BY(send_critsect_);
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 5c0ff1f..22fe71e 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -343,6 +343,7 @@
false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
&feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr));
+ rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
@@ -960,7 +961,7 @@
new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
nullptr, nullptr, nullptr, nullptr, &callback, nullptr,
nullptr, nullptr, &retransmission_rate_limiter_, nullptr));
-
+ rtp_sender_->SetSSRC(kSsrc);
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
@@ -1022,6 +1023,7 @@
nullptr, nullptr, nullptr, &callback, nullptr,
nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, nullptr));
+ rtp_sender_->SetSSRC(kSsrc);
// Simulate kNumPackets sent with kPacketInterval ms intervals, with the
// number of packets selected so that we fill (but don't overflow) the one
@@ -1080,6 +1082,7 @@
nullptr, nullptr, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, nullptr));
+ rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
}
};
@@ -1440,6 +1443,7 @@
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer));
+ rtp_sender_->SetSSRC(kSsrc);
// RTP overhead is 12B.
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(12)).Times(1);
@@ -1460,6 +1464,7 @@
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer));
+ rtp_sender_->SetSSRC(kSsrc);
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
SendGenericPayload();
@@ -1475,6 +1480,7 @@
false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
&feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, &mock_overhead_observer));
+ rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
diff --git a/modules/rtp_rtcp/source/ssrc_database.cc b/modules/rtp_rtcp/source/ssrc_database.cc
deleted file mode 100644
index a96d05d..0000000
--- a/modules/rtp_rtcp/source/ssrc_database.cc
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-
-SSRCDatabase* SSRCDatabase::GetSSRCDatabase() {
- return GetStaticInstance<SSRCDatabase>(kAddRef);
-}
-
-void SSRCDatabase::ReturnSSRCDatabase() {
- GetStaticInstance<SSRCDatabase>(kRelease);
-}
-
-uint32_t SSRCDatabase::CreateSSRC() {
- rtc::CritScope lock(&crit_);
-
- while (true) { // Try until get a new ssrc.
- // 0 and 0xffffffff are invalid values for SSRC.
- uint32_t ssrc = random_.Rand(1u, 0xfffffffe);
- if (ssrcs_.insert(ssrc).second) {
- return ssrc;
- }
- }
-}
-
-void SSRCDatabase::RegisterSSRC(uint32_t ssrc) {
- rtc::CritScope lock(&crit_);
- ssrcs_.insert(ssrc);
-}
-
-void SSRCDatabase::ReturnSSRC(uint32_t ssrc) {
- rtc::CritScope lock(&crit_);
- ssrcs_.erase(ssrc);
-}
-
-SSRCDatabase::SSRCDatabase() : random_(rtc::TimeMicros()) {}
-
-SSRCDatabase::~SSRCDatabase() {}
-
-} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/ssrc_database.h b/modules/rtp_rtcp/source/ssrc_database.h
deleted file mode 100644
index 2f6357a..0000000
--- a/modules/rtp_rtcp/source/ssrc_database.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
-
-#include <set>
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/random.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/system_wrappers/include/static_instance.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// TODO(tommi, holmer): Look into whether we can eliminate locking in this
-// class or the class itself completely once voice engine doesn't rely on it.
-// At the moment voe_auto_test requires locking, but it's not clear if that's
-// an issue with the test code or if it reflects real world usage or if that's
-// the best design performance wise.
-// If we do decide to keep the class, we should at least get rid of using
-// StaticInstance.
-class SSRCDatabase {
- public:
- static SSRCDatabase* GetSSRCDatabase();
- static void ReturnSSRCDatabase();
-
- uint32_t CreateSSRC();
- void RegisterSSRC(uint32_t ssrc);
- void ReturnSSRC(uint32_t ssrc);
-
- protected:
- SSRCDatabase();
- ~SSRCDatabase();
-
- static SSRCDatabase* CreateInstance() { return new SSRCDatabase(); }
-
- // Friend function to allow the SSRC destructor to be accessed from the
- // template class.
- friend SSRCDatabase* GetStaticInstance<SSRCDatabase>(
- CountOperation count_operation);
-
- private:
- rtc::CriticalSection crit_;
- Random random_ GUARDED_BY(crit_);
- std::set<uint32_t> ssrcs_ GUARDED_BY(crit_);
- // TODO(tommi): Use a thread checker to ensure the object is created and
- // deleted on the same thread. At the moment this isn't possible due to
- // voe::ChannelOwner in voice engine. To reproduce, run:
- // voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index b0aa654..2302acc 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -583,7 +583,7 @@
int32_t id,
AudioFrame* audioFrame) {
unsigned int ssrc;
- RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+ RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0);
event_log_proxy_->LogAudioPlayout(ssrc);
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
diff --git a/voice_engine/channel_manager.cc b/voice_engine/channel_manager.cc
index b8ffe6e..b81c8a6 100644
--- a/voice_engine/channel_manager.cc
+++ b/voice_engine/channel_manager.cc
@@ -10,6 +10,7 @@
#include "webrtc/voice_engine/channel_manager.h"
+#include "webrtc/base/timeutils.h"
#include "webrtc/voice_engine/channel.h"
namespace webrtc {
@@ -45,12 +46,18 @@
: channel(channel), ref_count(1) {}
ChannelManager::ChannelManager(uint32_t instance_id)
- : instance_id_(instance_id), last_channel_id_(-1) {}
+ : instance_id_(instance_id),
+ last_channel_id_(-1),
+ random_(rtc::TimeNanos()) {}
ChannelOwner ChannelManager::CreateChannel(
const VoEBase::ChannelConfig& config) {
Channel* channel;
Channel::CreateChannel(channel, ++last_channel_id_, instance_id_, config);
+ // TODO(solenberg): Delete this, users should configure ssrc
+ // explicitly.
+ channel->SetLocalSSRC(random_.Rand<uint32_t>());
+
ChannelOwner channel_owner(channel);
rtc::CritScope crit(&lock_);
diff --git a/voice_engine/channel_manager.h b/voice_engine/channel_manager.h
index ae16aca..65cf994 100644
--- a/voice_engine/channel_manager.h
+++ b/voice_engine/channel_manager.h
@@ -16,6 +16,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/random.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/typedefs.h"
@@ -116,6 +117,9 @@
rtc::CriticalSection lock_;
std::vector<ChannelOwner> channels_;
+ // For generation of random ssrc:s.
+ webrtc::Random random_;
+
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelManager);
};
} // namespace voe
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index b736aac..0cb9990 100644
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -91,9 +91,10 @@
return;
}
- // We need to sleep a bit here for the name to propagate. For instance,
- // 200 milliseconds is not enough, so we'll go with one second here.
- Sleep(1000);
+ // We need to sleep a bit here for the name to propagate. For
+ // instance, 200 milliseconds is not enough, 1 second still flaky,
+ // so we'll go with five seconds here.
+ Sleep(5000);
char char_buffer[256];
voe_rtp_rtcp_->GetRemoteRTCP_CNAME(channel_, char_buffer);
