Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
Review URL: https://codereview.webrtc.org/1877253002
Cr-Commit-Position: refs/heads/master@{#12359}
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index fae8641..6ce6434 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -13,6 +13,7 @@
#include <map>
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
@@ -122,7 +123,7 @@
const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
void ResetLastReceivedPayloadTypes() {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
last_received_payload_type_ = -1;
last_received_media_payload_type_ = -1;
}
@@ -136,34 +137,34 @@
bool ReportMediaPayloadType(uint8_t media_payload_type);
int8_t red_payload_type() const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return red_payload_type_;
}
int8_t ulpfec_payload_type() const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return ulpfec_payload_type_;
}
int8_t last_received_payload_type() const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return last_received_payload_type_;
}
void set_last_received_payload_type(int8_t last_received_payload_type) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
last_received_payload_type_ = last_received_payload_type;
}
int8_t last_received_media_payload_type() const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return last_received_media_payload_type_;
}
bool use_rtx_payload_mapping_on_restore() const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return use_rtx_payload_mapping_on_restore_;
}
void set_use_rtx_payload_mapping_on_restore(bool val) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
use_rtx_payload_mapping_on_restore_ = val;
}
@@ -178,7 +179,7 @@
bool IsRtxInternal(const RTPHeader& header) const;
- rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ rtc::CriticalSection crit_sect_;
RtpUtility::PayloadTypeMap payload_type_map_;
rtc::scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
int8_t red_payload_type_;
diff --git a/webrtc/modules/rtp_rtcp/source/bitrate.cc b/webrtc/modules/rtp_rtcp/source/bitrate.cc
index 4e9fc72..49a2359 100644
--- a/webrtc/modules/rtp_rtcp/source/bitrate.cc
+++ b/webrtc/modules/rtp_rtcp/source/bitrate.cc
@@ -11,13 +11,11 @@
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
Bitrate::Bitrate(Clock* clock, Observer* observer)
: clock_(clock),
- crit_(CriticalSectionWrapper::CreateCriticalSection()),
packet_rate_(0),
bitrate_(0),
bitrate_next_idx_(0),
@@ -33,23 +31,23 @@
Bitrate::~Bitrate() {}
void Bitrate::Update(const size_t bytes) {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
bytes_count_ += bytes;
packet_count_++;
}
uint32_t Bitrate::PacketRate() const {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
return packet_rate_;
}
uint32_t Bitrate::BitrateLast() const {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
return bitrate_;
}
uint32_t Bitrate::BitrateNow() const {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
int64_t now = clock_->TimeInMilliseconds();
int64_t diff_ms = now - time_last_rate_update_;
@@ -67,7 +65,7 @@
}
int64_t Bitrate::time_last_rate_update() const {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
return time_last_rate_update_;
}
@@ -75,7 +73,7 @@
void Bitrate::Process() {
BitrateStatistics stats;
{
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
int64_t now = clock_->CurrentNtpInMilliseconds();
int64_t diff_ms = now - time_last_rate_update_;
diff --git a/webrtc/modules/rtp_rtcp/source/bitrate.h b/webrtc/modules/rtp_rtcp/source/bitrate.h
index 393d05d..7aaaead 100644
--- a/webrtc/modules/rtp_rtcp/source/bitrate.h
+++ b/webrtc/modules/rtp_rtcp/source/bitrate.h
@@ -15,7 +15,7 @@
#include <list>
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/typedefs.h"
@@ -23,7 +23,6 @@
namespace webrtc {
class Clock;
-class CriticalSectionWrapper;
class Bitrate {
public:
@@ -60,7 +59,7 @@
Clock* clock_;
private:
- rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+ rtc::CriticalSection crit_;
uint32_t packet_rate_;
uint32_t bitrate_;
uint8_t bitrate_next_idx_;
diff --git a/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc b/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
index ab21b87..81e8b59 100644
--- a/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
+++ b/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -13,20 +13,16 @@
#include <string.h>
namespace webrtc {
-DTMFqueue::DTMFqueue()
- : dtmf_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
- next_empty_index_(0) {
+DTMFqueue::DTMFqueue() : next_empty_index_(0) {
memset(dtmf_key_, 0, sizeof(dtmf_key_));
memset(dtmf_length, 0, sizeof(dtmf_length));
memset(dtmf_level_, 0, sizeof(dtmf_level_));
}
-DTMFqueue::~DTMFqueue() {
- delete dtmf_critsect_;
-}
+DTMFqueue::~DTMFqueue() {}
int32_t DTMFqueue::AddDTMF(uint8_t key, uint16_t len, uint8_t level) {
- CriticalSectionScoped lock(dtmf_critsect_);
+ rtc::CritScope lock(&dtmf_critsect_);
if (next_empty_index_ >= DTMF_OUTBAND_MAX) {
return -1;
@@ -40,7 +36,7 @@
}
int8_t DTMFqueue::NextDTMF(uint8_t* dtmf_key, uint16_t* len, uint8_t* level) {
- CriticalSectionScoped lock(dtmf_critsect_);
+ rtc::CritScope lock(&dtmf_critsect_);
if (next_empty_index_ == 0)
return -1;
@@ -60,12 +56,12 @@
}
bool DTMFqueue::PendingDTMF() {
- CriticalSectionScoped lock(dtmf_critsect_);
+ rtc::CritScope lock(&dtmf_critsect_);
return next_empty_index_ > 0;
}
void DTMFqueue::ResetDTMF() {
- CriticalSectionScoped lock(dtmf_critsect_);
+ rtc::CritScope lock(&dtmf_critsect_);
next_empty_index_ = 0;
}
} // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/dtmf_queue.h b/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
index d1b3f56..c0e616f 100644
--- a/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
+++ b/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
+#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -27,7 +27,7 @@
void ResetDTMF();
private:
- CriticalSectionWrapper* dtmf_critsect_;
+ rtc::CriticalSection dtmf_critsect_;
uint8_t next_empty_index_;
uint8_t dtmf_key_[DTMF_OUTBAND_MAX];
uint16_t dtmf_length[DTMF_OUTBAND_MAX];
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
index 2109574..28e98ba 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
@@ -16,7 +16,6 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
// RFC 5109
namespace webrtc {
@@ -26,8 +25,7 @@
}
FecReceiverImpl::FecReceiverImpl(RtpData* callback)
- : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- recovered_packet_callback_(callback),
+ : recovered_packet_callback_(callback),
fec_(new ForwardErrorCorrection()) {}
FecReceiverImpl::~FecReceiverImpl() {
@@ -42,7 +40,7 @@
}
FecPacketCounter FecReceiverImpl::GetPacketCounter() const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return packet_counter_;
}
@@ -77,7 +75,7 @@
int32_t FecReceiverImpl::AddReceivedRedPacket(
const RTPHeader& header, const uint8_t* incoming_rtp_packet,
size_t packet_length, uint8_t ulpfec_payload_type) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
uint8_t REDHeaderLength = 1;
size_t payload_data_length = packet_length - header.headerLength;
@@ -219,21 +217,21 @@
}
int32_t FecReceiverImpl::ProcessReceivedFec() {
- crit_sect_->Enter();
+ crit_sect_.Enter();
if (!received_packet_list_.empty()) {
// Send received media packet to VCM.
if (!received_packet_list_.front()->is_fec) {
ForwardErrorCorrection::Packet* packet =
received_packet_list_.front()->pkt;
- crit_sect_->Leave();
+ crit_sect_.Leave();
if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
packet->length)) {
return -1;
}
- crit_sect_->Enter();
+ crit_sect_.Enter();
}
if (fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_) != 0) {
- crit_sect_->Leave();
+ crit_sect_.Leave();
return -1;
}
assert(received_packet_list_.empty());
@@ -246,15 +244,15 @@
continue;
ForwardErrorCorrection::Packet* packet = (*it)->pkt;
++packet_counter_.num_recovered_packets;
- crit_sect_->Leave();
+ crit_sect_.Leave();
if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
packet->length)) {
return -1;
}
- crit_sect_->Enter();
+ crit_sect_.Enter();
(*it)->returned = true;
}
- crit_sect_->Leave();
+ crit_sect_.Leave();
return 0;
}
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index 6a63813..0ebca9b 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -13,7 +13,7 @@
// This header is included to get the nested declaration of Packet structure.
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
@@ -21,8 +21,6 @@
namespace webrtc {
-class CriticalSectionWrapper;
-
class FecReceiverImpl : public FecReceiver {
public:
explicit FecReceiverImpl(RtpData* callback);
@@ -38,7 +36,7 @@
FecPacketCounter GetPacketCounter() const override;
private:
- rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ rtc::CriticalSection crit_sect_;
RtpData* recovered_packet_callback_;
ForwardErrorCorrection* fec_;
// TODO(holmer): In the current version received_packet_list_ is never more
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 022fc96..1e7b355 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -12,10 +12,8 @@
#include <math.h>
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/time_util.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
@@ -29,7 +27,6 @@
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback)
: clock_(clock),
- stream_lock_(CriticalSectionWrapper::CreateCriticalSection()),
incoming_bitrate_(clock, NULL),
ssrc_(0),
max_reordering_threshold_(kDefaultMaxReorderingThreshold),
@@ -59,7 +56,7 @@
void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
size_t packet_length,
bool retransmitted) {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
bool in_order = InOrderPacketInternal(header.sequenceNumber);
ssrc_ = header.ssrc;
incoming_bitrate_.Update(packet_length);
@@ -150,7 +147,7 @@
StreamDataCounters data;
uint32_t ssrc;
{
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
data = receive_counters_;
ssrc = ssrc_;
}
@@ -161,7 +158,7 @@
RtcpStatistics data;
uint32_t ssrc;
{
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
data = last_reported_statistics_;
ssrc = ssrc_;
}
@@ -171,7 +168,7 @@
void StreamStatisticianImpl::FecPacketReceived(const RTPHeader& header,
size_t packet_length) {
{
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
receive_counters_.fec.AddPacket(packet_length, header);
}
NotifyRtpCallback();
@@ -179,14 +176,14 @@
void StreamStatisticianImpl::SetMaxReorderingThreshold(
int max_reordering_threshold) {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
max_reordering_threshold_ = max_reordering_threshold;
}
bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
bool reset) {
{
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
if (received_seq_first_ == 0 &&
receive_counters_.transmitted.payload_bytes == 0) {
// We have not received anything.
@@ -282,7 +279,7 @@
void StreamStatisticianImpl::GetDataCounters(
size_t* bytes_received, uint32_t* packets_received) const {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
if (bytes_received) {
*bytes_received = receive_counters_.transmitted.payload_bytes +
receive_counters_.transmitted.header_bytes +
@@ -295,30 +292,30 @@
void StreamStatisticianImpl::GetReceiveStreamDataCounters(
StreamDataCounters* data_counters) const {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
*data_counters = receive_counters_;
}
uint32_t StreamStatisticianImpl::BitrateReceived() const {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
return incoming_bitrate_.BitrateNow();
}
void StreamStatisticianImpl::ProcessBitrate() {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
incoming_bitrate_.Process();
}
void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
uint32_t* frac) const {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
*secs = last_receive_time_ntp_.seconds();
*frac = last_receive_time_ntp_.fractions();
}
bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
const RTPHeader& header, int64_t min_rtt) const {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
if (InOrderPacketInternal(header.sequenceNumber)) {
return false;
}
@@ -352,7 +349,7 @@
}
bool StreamStatisticianImpl::IsPacketInOrder(uint16_t sequence_number) const {
- CriticalSectionScoped cs(stream_lock_.get());
+ rtc::CritScope cs(&stream_lock_);
return InOrderPacketInternal(sequence_number);
}
@@ -377,7 +374,6 @@
ReceiveStatisticsImpl::ReceiveStatisticsImpl(Clock* clock)
: clock_(clock),
- receive_statistics_lock_(CriticalSectionWrapper::CreateCriticalSection()),
last_rate_update_ms_(0),
rtcp_stats_callback_(NULL),
rtp_stats_callback_(NULL) {}
@@ -394,7 +390,7 @@
bool retransmitted) {
StreamStatisticianImpl* impl;
{
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
if (it != statisticians_.end()) {
impl = it->second;
@@ -412,7 +408,7 @@
void ReceiveStatisticsImpl::FecPacketReceived(const RTPHeader& header,
size_t packet_length) {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
// Ignore FEC if it is the first packet.
if (it != statisticians_.end()) {
@@ -421,7 +417,7 @@
}
StatisticianMap ReceiveStatisticsImpl::GetActiveStatisticians() const {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
StatisticianMap active_statisticians;
for (StatisticianImplMap::const_iterator it = statisticians_.begin();
it != statisticians_.end(); ++it) {
@@ -438,7 +434,7 @@
StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
uint32_t ssrc) const {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
StatisticianImplMap::const_iterator it = statisticians_.find(ssrc);
if (it == statisticians_.end())
return NULL;
@@ -447,7 +443,7 @@
void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
int max_reordering_threshold) {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
for (StatisticianImplMap::iterator it = statisticians_.begin();
it != statisticians_.end(); ++it) {
it->second->SetMaxReorderingThreshold(max_reordering_threshold);
@@ -455,7 +451,7 @@
}
void ReceiveStatisticsImpl::Process() {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
for (StatisticianImplMap::iterator it = statisticians_.begin();
it != statisticians_.end(); ++it) {
it->second->ProcessBitrate();
@@ -464,7 +460,7 @@
}
int64_t ReceiveStatisticsImpl::TimeUntilNextProcess() {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
int64_t time_since_last_update = clock_->TimeInMilliseconds() -
last_rate_update_ms_;
return std::max<int64_t>(
@@ -473,7 +469,7 @@
void ReceiveStatisticsImpl::RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
if (callback != NULL)
assert(rtcp_stats_callback_ == NULL);
rtcp_stats_callback_ = callback;
@@ -481,20 +477,20 @@
void ReceiveStatisticsImpl::StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
if (rtcp_stats_callback_)
rtcp_stats_callback_->StatisticsUpdated(statistics, ssrc);
}
void ReceiveStatisticsImpl::CNameChanged(const char* cname, uint32_t ssrc) {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
if (rtcp_stats_callback_)
rtcp_stats_callback_->CNameChanged(cname, ssrc);
}
void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
if (callback != NULL)
assert(rtp_stats_callback_ == NULL);
rtp_stats_callback_ = callback;
@@ -502,7 +498,7 @@
void ReceiveStatisticsImpl::DataCountersUpdated(const StreamDataCounters& stats,
uint32_t ssrc) {
- CriticalSectionScoped cs(receive_statistics_lock_.get());
+ rtc::CritScope cs(&receive_statistics_lock_);
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(stats, ssrc);
}
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index 6da8334..3967967 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -16,15 +16,12 @@
#include <algorithm>
#include <map>
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/ntp_time.h"
namespace webrtc {
-class CriticalSectionWrapper;
-
class StreamStatisticianImpl : public StreamStatistician {
public:
StreamStatisticianImpl(Clock* clock,
@@ -57,11 +54,11 @@
void UpdateCounters(const RTPHeader& rtp_header,
size_t packet_length,
bool retransmitted);
- void NotifyRtpCallback() LOCKS_EXCLUDED(stream_lock_.get());
- void NotifyRtcpCallback() LOCKS_EXCLUDED(stream_lock_.get());
+ void NotifyRtpCallback() LOCKS_EXCLUDED(stream_lock_);
+ void NotifyRtcpCallback() LOCKS_EXCLUDED(stream_lock_);
Clock* clock_;
- rtc::scoped_ptr<CriticalSectionWrapper> stream_lock_;
+ rtc::CriticalSection stream_lock_;
Bitrate incoming_bitrate_;
uint32_t ssrc_;
int max_reordering_threshold_; // In number of packets or sequence numbers.
@@ -131,7 +128,7 @@
typedef std::map<uint32_t, StreamStatisticianImpl*> StatisticianImplMap;
Clock* clock_;
- rtc::scoped_ptr<CriticalSectionWrapper> receive_statistics_lock_;
+ rtc::CriticalSection receive_statistics_lock_;
int64_t last_rate_update_ms_;
StatisticianImplMap statisticians_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 0faf2a4..3e8e47f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -49,13 +49,9 @@
receiver_only_(receiver_only),
_lastReceived(0),
_rtpRtcp(*owner),
- _criticalSectionFeedbacks(
- CriticalSectionWrapper::CreateCriticalSection()),
_cbRtcpBandwidthObserver(rtcp_bandwidth_observer),
_cbRtcpIntraFrameObserver(rtcp_intra_frame_observer),
_cbTransportFeedbackObserver(transport_feedback_observer),
- _criticalSectionRTCPReceiver(
- CriticalSectionWrapper::CreateCriticalSection()),
main_ssrc_(0),
_remoteSSRC(0),
_remoteSenderInfo(),
@@ -76,9 +72,6 @@
}
RTCPReceiver::~RTCPReceiver() {
- delete _criticalSectionRTCPReceiver;
- delete _criticalSectionFeedbacks;
-
ReportBlockMap::iterator it = _receivedReportBlockMap.begin();
for (; it != _receivedReportBlockMap.end(); ++it) {
ReportBlockInfoMap* info_map = &(it->second);
@@ -103,12 +96,12 @@
}
int64_t RTCPReceiver::LastReceived() {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
return _lastReceived;
}
int64_t RTCPReceiver::LastReceivedReceiverReport() const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
int64_t last_received_rr = -1;
for (ReceivedInfoMap::const_iterator it = _receivedInfoMap.begin();
it != _receivedInfoMap.end(); ++it) {
@@ -120,7 +113,7 @@
}
void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
// new SSRC reset old reports
memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
@@ -131,7 +124,7 @@
}
uint32_t RTCPReceiver::RemoteSSRC() const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
return _remoteSSRC;
}
@@ -139,7 +132,7 @@
const std::set<uint32_t>& registered_ssrcs) {
uint32_t old_ssrc = 0;
{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
old_ssrc = main_ssrc_;
main_ssrc_ = main_ssrc;
registered_ssrcs_ = registered_ssrcs;
@@ -156,7 +149,7 @@
int64_t* avgRTT,
int64_t* minRTT,
int64_t* maxRTT) const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
RTCPReportBlockInformation* reportBlock =
GetReportBlockInformation(remoteSSRC, main_ssrc_);
@@ -180,13 +173,13 @@
}
void RTCPReceiver::SetRtcpXrRrtrStatus(bool enable) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
xr_rrtr_status_ = enable;
}
bool RTCPReceiver::GetAndResetXrRrRtt(int64_t* rtt_ms) {
assert(rtt_ms);
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
if (xr_rr_rtt_ms_ == 0) {
return false;
}
@@ -202,7 +195,7 @@
uint32_t* RTCPArrivalTimeFrac,
uint32_t* rtcp_timestamp) const
{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
if(ReceivedNTPsecs)
{
*ReceivedNTPsecs = _remoteSenderInfo.NTPseconds; // NTP from incoming SendReport
@@ -228,7 +221,7 @@
bool RTCPReceiver::LastReceivedXrReferenceTimeInfo(
RtcpReceiveTimeInfo* info) const {
assert(info);
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
if (_lastReceivedXRNTPsecs == 0 && _lastReceivedXRNTPfrac == 0) {
return false;
}
@@ -251,7 +244,7 @@
int32_t RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const {
assert(senderInfo);
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
if (_lastReceivedSRNTPsecs == 0) {
return -1;
}
@@ -264,7 +257,7 @@
int32_t RTCPReceiver::StatisticsReceived(
std::vector<RTCPReportBlock>* receiveBlocks) const {
assert(receiveBlocks);
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
ReportBlockMap::const_iterator it = _receivedReportBlockMap.begin();
for (; it != _receivedReportBlockMap.end(); ++it) {
const ReportBlockInfoMap* info_map = &(it->second);
@@ -280,7 +273,7 @@
RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
RTCPUtility::RTCPParserV2* rtcpParser)
{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
_lastReceived = _clock->TimeInMilliseconds();
@@ -590,7 +583,7 @@
RTCPCnameInformation*
RTCPReceiver::CreateCnameInformation(uint32_t remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
std::map<uint32_t, RTCPCnameInformation*>::iterator it =
_receivedCnameMap.find(remoteSSRC);
@@ -606,7 +599,7 @@
RTCPCnameInformation*
RTCPReceiver::GetCnameInformation(uint32_t remoteSSRC) const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
std::map<uint32_t, RTCPCnameInformation*>::const_iterator it =
_receivedCnameMap.find(remoteSSRC);
@@ -619,7 +612,7 @@
RTCPReceiveInformation*
RTCPReceiver::CreateReceiveInformation(uint32_t remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
std::map<uint32_t, RTCPReceiveInformation*>::iterator it =
_receivedInfoMap.find(remoteSSRC);
@@ -634,7 +627,7 @@
RTCPReceiveInformation*
RTCPReceiver::GetReceiveInformation(uint32_t remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
std::map<uint32_t, RTCPReceiveInformation*>::iterator it =
_receivedInfoMap.find(remoteSSRC);
@@ -651,7 +644,7 @@
}
bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
if (_lastReceivedRrMs == 0)
return false;
@@ -665,7 +658,7 @@
}
bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
if (_lastIncreasedSequenceNumberMs == 0)
return false;
@@ -680,7 +673,7 @@
}
bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
bool updateBoundingSet = false;
int64_t timeNow = _clock->TimeInMilliseconds();
@@ -724,7 +717,7 @@
}
int32_t RTCPReceiver::BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
std::map<uint32_t, RTCPReceiveInformation*>::iterator receiveInfoIt =
_receivedInfoMap.find(_remoteSSRC);
@@ -773,7 +766,7 @@
cnameInfo->name[RTCP_CNAME_SIZE - 1] = 0;
strncpy(cnameInfo->name, rtcpPacket.CName.CName, RTCP_CNAME_SIZE - 1);
{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
+ rtc::CritScope lock(&_criticalSectionFeedbacks);
if (stats_callback_ != NULL) {
stats_callback_->CNameChanged(rtcpPacket.CName.CName,
rtcpPacket.CName.SenderSSRC);
@@ -1283,12 +1276,12 @@
void RTCPReceiver::RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {
- CriticalSectionScoped cs(_criticalSectionFeedbacks);
+ rtc::CritScope cs(&_criticalSectionFeedbacks);
stats_callback_ = callback;
}
RtcpStatisticsCallback* RTCPReceiver::GetRtcpStatisticsCallback() {
- CriticalSectionScoped cs(_criticalSectionFeedbacks);
+ rtc::CritScope cs(&_criticalSectionFeedbacks);
return stats_callback_;
}
@@ -1305,7 +1298,7 @@
std::set<uint32_t> registered_ssrcs;
{
// We don't want to hold this critsect when triggering the callbacks below.
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
local_ssrc = main_ssrc_;
registered_ssrcs = registered_ssrcs_;
}
@@ -1378,7 +1371,7 @@
}
if (!receiver_only_) {
- CriticalSectionScoped cs(_criticalSectionFeedbacks);
+ rtc::CritScope cs(&_criticalSectionFeedbacks);
if (stats_callback_) {
for (ReportBlockList::const_iterator it =
rtcpPacketInformation.report_blocks.begin();
@@ -1400,7 +1393,7 @@
char cName[RTCP_CNAME_SIZE]) const {
assert(cName);
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC);
if (cnameInfo == NULL) {
return -1;
@@ -1414,7 +1407,7 @@
int32_t RTCPReceiver::TMMBRReceived(uint32_t size,
uint32_t accNumCandidates,
TMMBRSet* candidateSet) const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ rtc::CritScope lock(&_criticalSectionRTCPReceiver);
std::map<uint32_t, RTCPReceiveInformation*>::const_iterator
receiveInfoIt = _receivedInfoMap.begin();
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 475ab1e..28c28cb6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -15,6 +15,7 @@
#include <set>
#include <vector>
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
@@ -267,12 +268,12 @@
int64_t _lastReceived;
ModuleRtpRtcpImpl& _rtpRtcp;
- CriticalSectionWrapper* _criticalSectionFeedbacks;
+ rtc::CriticalSection _criticalSectionFeedbacks;
RtcpBandwidthObserver* const _cbRtcpBandwidthObserver;
RtcpIntraFrameObserver* const _cbRtcpIntraFrameObserver;
TransportFeedbackObserver* const _cbTransportFeedbackObserver;
- CriticalSectionWrapper* _criticalSectionRTCPReceiver;
+ rtc::CriticalSection _criticalSectionRTCPReceiver;
uint32_t main_ssrc_ GUARDED_BY(_criticalSectionRTCPReceiver);
uint32_t _remoteSSRC GUARDED_BY(_criticalSectionRTCPReceiver);
std::set<uint32_t> registered_ssrcs_ GUARDED_BY(_criticalSectionRTCPReceiver);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index d4cbe54..2cec8a3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
@@ -9,10 +9,9 @@
*/
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
@@ -30,7 +29,7 @@
bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
private:
- rtc::scoped_ptr<CriticalSectionWrapper> critical_section_;
+ rtc::CriticalSection critical_section_;
RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(critical_section_);
};
@@ -38,8 +37,7 @@
return new RtpHeaderParserImpl;
}
-RtpHeaderParserImpl::RtpHeaderParserImpl()
- : critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {}
+RtpHeaderParserImpl::RtpHeaderParserImpl() {}
bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
@@ -54,7 +52,7 @@
RtpHeaderExtensionMap map;
{
- CriticalSectionScoped cs(critical_section_.get());
+ rtc::CritScope cs(&critical_section_);
rtp_header_extension_map_.GetCopy(&map);
}
@@ -67,12 +65,12 @@
bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
- CriticalSectionScoped cs(critical_section_.get());
+ rtc::CritScope cs(&critical_section_);
return rtp_header_extension_map_.Register(type, id) == 0;
}
bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
- CriticalSectionScoped cs(critical_section_.get());
+ rtc::CritScope cs(&critical_section_);
return rtp_header_extension_map_.Deregister(type) == 0;
}
} // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index 49f9d85..713fba8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -21,7 +21,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
@@ -29,7 +28,6 @@
RTPPacketHistory::RTPPacketHistory(Clock* clock)
: clock_(clock),
- critsect_(CriticalSectionWrapper::CreateCriticalSection()),
store_(false),
prev_index_(0) {}
@@ -38,7 +36,7 @@
void RTPPacketHistory::SetStorePacketsStatus(bool enable,
uint16_t number_to_store) {
- CriticalSectionScoped cs(critsect_.get());
+ rtc::CritScope cs(&critsect_);
if (enable) {
if (store_) {
LOG(LS_WARNING) << "Purging packet history in order to re-set status.";
@@ -70,7 +68,7 @@
}
bool RTPPacketHistory::StorePackets() const {
- CriticalSectionScoped cs(critsect_.get());
+ rtc::CritScope cs(&critsect_);
return store_;
}
@@ -78,7 +76,7 @@
size_t packet_length,
int64_t capture_time_ms,
StorageType type) {
- CriticalSectionScoped cs(critsect_.get());
+ rtc::CritScope cs(&critsect_);
if (!store_) {
return 0;
}
@@ -131,7 +129,7 @@
}
bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const {
- CriticalSectionScoped cs(critsect_.get());
+ rtc::CritScope cs(&critsect_);
if (!store_) {
return false;
}
@@ -150,7 +148,7 @@
}
bool RTPPacketHistory::SetSent(uint16_t sequence_number) {
- CriticalSectionScoped cs(critsect_.get());
+ rtc::CritScope cs(&critsect_);
if (!store_) {
return false;
}
@@ -176,7 +174,7 @@
uint8_t* packet,
size_t* packet_length,
int64_t* stored_time_ms) {
- CriticalSectionScoped cs(critsect_.get());
+ rtc::CritScope cs(&critsect_);
RTC_CHECK_GE(*packet_length, static_cast<size_t>(IP_PACKET_SIZE));
if (!store_)
return false;
@@ -232,7 +230,7 @@
bool RTPPacketHistory::GetBestFittingPacket(uint8_t* packet,
size_t* packet_length,
int64_t* stored_time_ms) {
- CriticalSectionScoped cs(critsect_.get());
+ rtc::CritScope cs(&critsect_);
if (!store_)
return false;
int index = FindBestFittingPacket(*packet_length);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index 8e1a732..b4d48aa 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -15,6 +15,7 @@
#include <vector>
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -23,7 +24,6 @@
namespace webrtc {
class Clock;
-class CriticalSectionWrapper;
static const size_t kMaxHistoryCapacity = 9600;
@@ -71,19 +71,19 @@
uint8_t* packet,
size_t* packet_length,
int64_t* stored_time_ms) const
- EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
- void Allocate(size_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
- void Free() EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+ EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+ void Allocate(size_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+ void Free() EXCLUSIVE_LOCKS_REQUIRED(critsect_);
void VerifyAndAllocatePacketLength(size_t packet_length, uint32_t start_index)
- EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+ EXCLUSIVE_LOCKS_REQUIRED(critsect_);
bool FindSeqNum(uint16_t sequence_number, int32_t* index) const
- EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+ EXCLUSIVE_LOCKS_REQUIRED(critsect_);
int FindBestFittingPacket(size_t size) const
- EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+ EXCLUSIVE_LOCKS_REQUIRED(critsect_);
private:
Clock* clock_;
- rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
+ rtc::CriticalSection critsect_;
bool store_ GUARDED_BY(critsect_);
uint32_t prev_index_ GUARDED_BY(critsect_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index f3793d0..e379580 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -16,8 +16,7 @@
namespace webrtc {
RTPPayloadRegistry::RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
- : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- rtp_payload_strategy_(rtp_payload_strategy),
+ : rtp_payload_strategy_(rtp_payload_strategy),
red_payload_type_(-1),
ulpfec_payload_type_(-1),
incoming_payload_type_(-1),
@@ -67,7 +66,7 @@
size_t payload_name_length = strlen(payload_name);
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
RtpUtility::PayloadTypeMap::iterator it =
payload_type_map_.find(payload_type);
@@ -122,7 +121,7 @@
int32_t RTPPayloadRegistry::DeRegisterReceivePayload(
const int8_t payload_type) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
RtpUtility::PayloadTypeMap::iterator it =
payload_type_map_.find(payload_type);
assert(it != payload_type_map_.end());
@@ -176,7 +175,7 @@
assert(payload_type);
size_t payload_name_length = strlen(payload_name);
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
RtpUtility::PayloadTypeMap::const_iterator it = payload_type_map_.begin();
@@ -218,12 +217,12 @@
}
bool RTPPayloadRegistry::RtxEnabled() const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return rtx_;
}
bool RTPPayloadRegistry::IsRtx(const RTPHeader& header) const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return IsRtxInternal(header);
}
@@ -264,7 +263,7 @@
original_sequence_number);
ByteWriter<uint32_t>::WriteBigEndian(restored_packet + 8, original_ssrc);
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
if (!rtx_)
return true;
@@ -290,20 +289,20 @@
}
void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
ssrc_rtx_ = ssrc;
rtx_ = true;
}
bool RTPPayloadRegistry::GetRtxSsrc(uint32_t* ssrc) const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
*ssrc = ssrc_rtx_;
return rtx_;
}
void RTPPayloadRegistry::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
if (payload_type < 0) {
LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
return;
@@ -315,7 +314,7 @@
}
bool RTPPayloadRegistry::IsRed(const RTPHeader& header) const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return red_payload_type_ == header.payloadType;
}
@@ -325,7 +324,7 @@
bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type,
PayloadUnion* payload) const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
RtpUtility::PayloadTypeMap::const_iterator it =
payload_type_map_.find(payload_type);
@@ -343,13 +342,13 @@
if (!payload) {
return -1;
}
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
return rtp_payload_strategy_->GetPayloadTypeFrequency(*payload);
}
const RtpUtility::Payload* RTPPayloadRegistry::PayloadTypeToPayload(
uint8_t payload_type) const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
RtpUtility::PayloadTypeMap::const_iterator it =
payload_type_map_.find(payload_type);
@@ -363,13 +362,13 @@
}
void RTPPayloadRegistry::SetIncomingPayloadType(const RTPHeader& header) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
if (!IsRtxInternal(header))
incoming_payload_type_ = header.payloadType;
}
bool RTPPayloadRegistry::ReportMediaPayloadType(uint8_t media_payload_type) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
if (last_received_media_payload_type_ == media_payload_type) {
// Media type unchanged.
return true;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 9de65ab..38b2830 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -16,7 +16,6 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
@@ -46,26 +45,26 @@
// Outband TelephoneEvent(DTMF) detection
void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
bool forward_to_decoder) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
telephone_event_forward_to_decoder_ = forward_to_decoder;
}
// Is forwarding of outband telephone events turned on/off?
bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return telephone_event_forward_to_decoder_;
}
bool RTPReceiverAudio::TelephoneEventPayloadType(
int8_t payload_type) const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return telephone_event_payload_type_ == payload_type;
}
bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type,
uint32_t* frequency,
bool* cng_payload_type_has_changed) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
*cng_payload_type_has_changed = false;
// We can have four CNG on 8000Hz, 16000Hz, 32000Hz and 48000Hz.
@@ -152,7 +151,7 @@
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_type,
uint32_t frequency) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) {
telephone_event_payload_type_ = payload_type;
@@ -206,7 +205,7 @@
}
int RTPReceiverAudio::GetPayloadTypeFrequency() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (last_received_g722_) {
return 8000;
}
@@ -249,7 +248,7 @@
}
int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
assert(num_energy_ <= kRtpCsrcSize);
@@ -291,7 +290,7 @@
bool telephone_event_packet =
TelephoneEventPayloadType(rtp_header->header.payloadType);
if (telephone_event_packet) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
// RFC 4733 2.3
// 0 1 2 3
@@ -336,7 +335,7 @@
}
{
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (!telephone_event_packet) {
last_received_frequency_ = audio_specific.frequency;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index f10613b..bec1578 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -14,7 +14,6 @@
#include <set>
#include "webrtc/base/onetimeevent.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -23,8 +22,6 @@
namespace webrtc {
-class CriticalSectionWrapper;
-
// Handles audio RTP packets. This class is thread-safe.
class RTPReceiverAudio : public RTPReceiverStrategy,
public TelephoneEventHandler {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 6f2efe7..04725ae 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -61,8 +61,6 @@
rtp_payload_registry_(rtp_payload_registry),
rtp_media_receiver_(rtp_media_receiver),
cb_rtp_feedback_(incoming_messages_callback),
- critical_section_rtp_receiver_(
- CriticalSectionWrapper::CreateCriticalSection()),
last_receive_time_(0),
last_received_payload_length_(0),
ssrc_(0),
@@ -89,7 +87,7 @@
const uint32_t frequency,
const size_t channels,
const uint32_t rate) {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
// TODO(phoglund): Try to streamline handling of the RED codec and some other
// cases which makes it necessary to keep track of whether we created a
@@ -111,29 +109,29 @@
int32_t RtpReceiverImpl::DeRegisterReceivePayload(
const int8_t payload_type) {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
}
NACKMethod RtpReceiverImpl::NACK() const {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
return nack_method_;
}
// Turn negative acknowledgment requests on/off.
void RtpReceiverImpl::SetNACKStatus(const NACKMethod method) {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
nack_method_ = method;
}
uint32_t RtpReceiverImpl::SSRC() const {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
return ssrc_;
}
// Get remote CSRC.
int32_t RtpReceiverImpl::CSRCs(uint32_t array_of_csrcs[kRtpCsrcSize]) const {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
assert(num_csrcs_ <= kRtpCsrcSize);
@@ -179,7 +177,7 @@
bool is_first_packet_in_frame = false;
{
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
if (HaveReceivedFrame()) {
is_first_packet_in_frame =
last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
@@ -198,7 +196,7 @@
}
{
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
last_receive_time_ = clock_->TimeInMilliseconds();
last_received_payload_length_ = payload_data_length;
@@ -219,7 +217,7 @@
}
bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
if (!HaveReceivedFrame())
return false;
*timestamp = last_received_timestamp_;
@@ -227,7 +225,7 @@
}
bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
if (!HaveReceivedFrame())
return false;
*receive_time_ms = last_received_frame_time_ms_;
@@ -247,7 +245,7 @@
uint32_t rate = 0;
{
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
int8_t last_received_payload_type =
rtp_payload_registry_->last_received_payload_type();
@@ -318,7 +316,7 @@
int8_t payload_type = rtp_header.payloadType;
{
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
int8_t last_received_payload_type =
rtp_payload_registry_->last_received_payload_type();
@@ -401,7 +399,7 @@
uint8_t old_num_csrcs = 0;
{
- CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+ rtc::CritScope lock(&critical_section_rtp_receiver_);
if (!rtp_media_receiver_->ShouldReportCsrcChanges(
rtp_header.header.payloadType)) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 63b65fe..7c6287c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -11,11 +11,11 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -79,7 +79,7 @@
RtpFeedback* cb_rtp_feedback_;
- rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
+ rtc::CriticalSection critical_section_rtp_receiver_;
int64_t last_receive_time_;
size_t last_received_payload_length_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
index 3797b1b..69d079f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
@@ -12,25 +12,22 @@
#include <stdlib.h>
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-
namespace webrtc {
RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback)
- : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- data_callback_(data_callback) {
+ : data_callback_(data_callback) {
memset(&last_payload_, 0, sizeof(last_payload_));
}
void RTPReceiverStrategy::GetLastMediaSpecificPayload(
PayloadUnion* payload) const {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
memcpy(payload, &last_payload_, sizeof(*payload));
}
void RTPReceiverStrategy::SetLastMediaSpecificPayload(
const PayloadUnion& payload) {
- CriticalSectionScoped cs(crit_sect_.get());
+ rtc::CritScope cs(&crit_sect_);
memcpy(&last_payload_, &payload, sizeof(last_payload_));
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index f2a60ff..663b883 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -11,11 +11,10 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -95,7 +94,7 @@
// packet.
explicit RTPReceiverStrategy(RtpData* data_callback);
- rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ rtc::CriticalSection crit_sect_;
PayloadUnion last_payload_;
RtpData* data_callback_;
};
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 72fb1b2..f53f55a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -21,7 +21,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 7500232..dc89b8f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -12,7 +12,6 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
#include "webrtc/base/onetimeevent.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 5875529..64f64d8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -112,7 +112,6 @@
key_frame_req_method_(kKeyFrameReqPliRtcp),
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_stats_(configuration.rtt_stats),
- critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()),
rtt_ms_(0) {
// Make sure that RTCP objects are aware of our SSRC.
uint32_t SSRC = rtp_sender_.SSRC();
@@ -982,12 +981,12 @@
}
void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
- CriticalSectionScoped cs(critical_section_rtt_.get());
+ rtc::CritScope cs(&critical_section_rtt_);
rtt_ms_ = rtt_ms;
}
int64_t ModuleRtpRtcpImpl::rtt_ms() const {
- CriticalSectionScoped cs(critical_section_rtt_.get());
+ rtc::CritScope cs(&critical_section_rtt_);
return rtt_ms_;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 76faca0..ec8e84a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -16,8 +16,8 @@
#include <utility>
#include <vector>
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/gtest_prod_util.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
@@ -362,7 +362,7 @@
PacketLossStats receive_loss_stats_;
// The processed RTT from RtcpRttStats.
- rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
+ rtc::CriticalSection critical_section_rtt_;
int64_t rtt_ms_;
};
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 3fbca7b..f7b72b8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -24,7 +24,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
#include "webrtc/modules/rtp_rtcp/source/time_util.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
namespace webrtc {
@@ -147,7 +146,6 @@
nack_bitrate_(clock, bitrates_.retransmit_bitrate_observer()),
packet_history_(clock),
// Statistics
- statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
rtp_stats_callback_(NULL),
frame_count_observer_(frame_count_observer),
send_side_delay_observer_(send_side_delay_observer),
@@ -166,7 +164,6 @@
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
- target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
target_bitrate_(0) {
memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
@@ -210,12 +207,12 @@
}
void RTPSender::SetTargetBitrate(uint32_t bitrate) {
- CriticalSectionScoped cs(target_bitrate_critsect_.get());
+ rtc::CritScope cs(&target_bitrate_critsect_);
target_bitrate_ = bitrate;
}
uint32_t RTPSender::GetTargetBitrate() {
- CriticalSectionScoped cs(target_bitrate_critsect_.get());
+ rtc::CritScope cs(&target_bitrate_critsect_);
return target_bitrate_;
}
@@ -532,7 +529,7 @@
payload_size, fragmentation, rtp_hdr);
}
- CriticalSectionScoped cs(statistics_crit_.get());
+ rtc::CritScope cs(&statistics_crit_);
// Note: This is currently only counting for video.
if (frame_type == kVideoFrameKey) {
++frame_counts_.key_frames;
@@ -966,7 +963,7 @@
// Get ssrc before taking statistics_crit_ to avoid possible deadlock.
uint32_t ssrc = is_rtx ? RtxSsrc() : SSRC();
- CriticalSectionScoped lock(statistics_crit_.get());
+ rtc::CritScope lock(&statistics_crit_);
if (is_rtx) {
counters = &rtx_rtp_stats_;
} else {
@@ -1109,7 +1106,7 @@
ssrc = ssrc_;
}
{
- CriticalSectionScoped cs(statistics_crit_.get());
+ rtc::CritScope cs(&statistics_crit_);
// TODO(holmer): Compute this iteratively instead.
send_delays_[now_ms] = now_ms - capture_time_ms;
send_delays_.erase(send_delays_.begin(),
@@ -1157,7 +1154,7 @@
void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
- CriticalSectionScoped lock(statistics_crit_.get());
+ rtc::CritScope lock(&statistics_crit_);
*rtp_stats = rtp_stats_;
*rtx_stats = rtx_rtp_stats_;
}
@@ -1858,12 +1855,12 @@
void RTPSender::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
- CriticalSectionScoped cs(statistics_crit_.get());
+ rtc::CritScope cs(&statistics_crit_);
rtp_stats_callback_ = callback;
}
StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
- CriticalSectionScoped cs(statistics_crit_.get());
+ rtc::CritScope cs(&statistics_crit_);
return rtp_stats_callback_;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 4344df6..df75f11 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -454,7 +454,7 @@
RTPPacketHistory packet_history_;
// Statistics
- rtc::scoped_ptr<CriticalSectionWrapper> statistics_crit_;
+ rtc::CriticalSection statistics_crit_;
SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
@@ -489,7 +489,7 @@
// SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
// that by the time the function returns there is no guarantee
// that the target bitrate is still valid.
- rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
+ rtc::CriticalSection target_bitrate_critsect_;
uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index c85a197..6d0f7a4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -25,7 +25,6 @@
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
: _clock(clock),
_rtpSender(rtpSender),
- _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_packetSizeSamples(160),
_dtmfEventIsOn(false),
_dtmfEventFirstPacketSent(false),
@@ -54,7 +53,7 @@
// set audio packet size, used to determine when it's time to send a DTMF packet
// in silence (CNG)
int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packetSizeSamples) {
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
_packetSizeSamples = packetSizeSamples;
return 0;
@@ -68,7 +67,7 @@
const uint32_t rate,
RtpUtility::Payload** payload) {
if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
// we can have multiple CNG payload types
switch (frequency) {
case 8000:
@@ -87,7 +86,7 @@
return -1;
}
} else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) {
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
// Don't add it to the list
// we dont want to allow send with a DTMF payloadtype
_dtmfPayloadType = payloadType;
@@ -105,7 +104,7 @@
}
bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
// for audio true for first packet in a speech burst
bool markerBit = false;
if (_lastPayloadType != payload_type) {
@@ -163,7 +162,7 @@
int8_t dtmf_payload_type;
uint16_t packet_size_samples;
{
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
red_payload_type = _REDPayloadType;
audio_level_dbov = _audioLevel_dBov;
dtmf_payload_type = _dtmfPayloadType;
@@ -336,7 +335,7 @@
}
{
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
_lastPayloadType = payloadType;
}
// Update audio level extension, if included.
@@ -365,7 +364,7 @@
if (level_dBov > 127) {
return -1;
}
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
_audioLevel_dBov = level_dBov;
return 0;
}
@@ -375,14 +374,14 @@
if (payloadType < -1) {
return -1;
}
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
_REDPayloadType = payloadType;
return 0;
}
// Get payload type for Redundant Audio Data RFC 2198
int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
- CriticalSectionScoped cs(_sendAudioCritsect.get());
+ rtc::CritScope cs(&_sendAudioCritsect);
if (_REDPayloadType == -1) {
// not configured
return -1;
@@ -396,7 +395,7 @@
uint16_t time_ms,
uint8_t level) {
{
- CriticalSectionScoped lock(_sendAudioCritsect.get());
+ rtc::CritScope lock(&_sendAudioCritsect);
if (_dtmfPayloadType < 0) {
// TelephoneEvent payloadtype not configured
return -1;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 2ba10d2..4bc0266 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -12,6 +12,7 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
#include "webrtc/common_types.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/onetimeevent.h"
#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
@@ -73,7 +74,7 @@
Clock* const _clock;
RTPSender* const _rtpSender;
- rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+ rtc::CriticalSection _sendAudioCritsect;
uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 87c975d..d617f10 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -24,14 +24,12 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
enum { REDForFECHeaderLength = 1 };
RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
: _rtpSender(*rtpSender),
- crit_(CriticalSectionWrapper::CreateCriticalSection()),
_videoType(kRtpVideoGeneric),
_retransmissionSettings(kRetransmitBaseLayer),
// Generic FEC
@@ -119,7 +117,7 @@
uint16_t next_fec_sequence_number = 0;
{
// Only protect while creating RED and FEC packets, not when sending.
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
red_packet.reset(producer_fec_.BuildRedPacket(
data_buffer, payload_length, rtp_header_length, red_payload_type_));
if (protect) {
@@ -170,7 +168,7 @@
void RTPSenderVideo::SetGenericFECStatus(const bool enable,
const uint8_t payloadTypeRED,
const uint8_t payloadTypeFEC) {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
fec_enabled_ = enable;
red_payload_type_ = payloadTypeRED;
fec_payload_type_ = payloadTypeFEC;
@@ -184,14 +182,14 @@
void RTPSenderVideo::GenericFECStatus(bool* enable,
uint8_t* payloadTypeRED,
uint8_t* payloadTypeFEC) const {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
*enable = fec_enabled_;
*payloadTypeRED = red_payload_type_;
*payloadTypeFEC = fec_payload_type_;
}
size_t RTPSenderVideo::FECPacketOverhead() const {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
if (fec_enabled_) {
// Overhead is FEC headers plus RED for FEC header plus anything in RTP
// header beyond the 12 bytes base header (CSRC list, extensions...)
@@ -206,7 +204,7 @@
void RTPSenderVideo::SetFecParameters(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params) {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
RTC_DCHECK(delta_params);
RTC_DCHECK(key_params);
delta_fec_params_ = *delta_params;
@@ -234,7 +232,7 @@
bool fec_enabled;
bool first_frame = first_frame_sent_();
{
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
FecProtectionParams* fec_params =
frameType == kVideoFrameKey ? &key_fec_params_ : &delta_fec_params_;
producer_fec_.SetFecParameters(fec_params, 0);
@@ -345,12 +343,12 @@
}
int RTPSenderVideo::SelectiveRetransmissions() const {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
return _retransmissionSettings;
}
void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
- CriticalSectionScoped cs(crit_.get());
+ rtc::CritScope cs(&crit_);
_retransmissionSettings = settings;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 3ea9168..8307b83 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -13,8 +13,8 @@
#include <list>
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/onetimeevent.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -28,7 +28,6 @@
#include "webrtc/typedefs.h"
namespace webrtc {
-class CriticalSectionWrapper;
class RTPSenderVideo {
public:
@@ -98,7 +97,7 @@
RTPSenderInterface& _rtpSender;
// Should never be held when calling out of this class.
- const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+ const rtc::CriticalSection crit_;
RtpVideoCodecTypes _videoType;
int32_t _retransmissionSettings GUARDED_BY(crit_);
diff --git a/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc b/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
index da43204..af95090 100644
--- a/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
+++ b/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -67,8 +67,7 @@
}
TMMBRHelp::TMMBRHelp()
- : _criticalSection(CriticalSectionWrapper::CreateCriticalSection()),
- _candidateSet(),
+ : _candidateSet(),
_boundingSet(),
_boundingSetToSend(),
_ptrIntersectionBoundingSet(NULL),
@@ -80,13 +79,12 @@
delete [] _ptrMaxPRBoundingSet;
_ptrIntersectionBoundingSet = 0;
_ptrMaxPRBoundingSet = 0;
- delete _criticalSection;
}
TMMBRSet*
TMMBRHelp::VerifyAndAllocateBoundingSet(uint32_t minimumSize)
{
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
if(minimumSize > _boundingSet.capacity())
{
@@ -110,7 +108,7 @@
int32_t
TMMBRHelp::SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend)
{
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
if (boundingSetToSend == NULL)
{
@@ -134,7 +132,7 @@
int32_t
TMMBRHelp::VerifyAndAllocateBoundingSetToSend(uint32_t minimumSize)
{
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
_boundingSetToSend.VerifyAndAllocateSet(minimumSize);
return 0;
@@ -143,7 +141,7 @@
TMMBRSet*
TMMBRHelp::VerifyAndAllocateCandidateSet(uint32_t minimumSize)
{
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
_candidateSet.VerifyAndAllocateSet(minimumSize);
return &_candidateSet;
@@ -164,7 +162,7 @@
int32_t
TMMBRHelp::FindTMMBRBoundingSet(TMMBRSet*& boundingSet)
{
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
// Work on local variable, will be modified
TMMBRSet candidateSet;
@@ -207,7 +205,7 @@
int32_t
TMMBRHelp::FindTMMBRBoundingSet(int32_t numCandidates, TMMBRSet& candidateSet)
{
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
uint32_t numBoundingSet = 0;
VerifyAndAllocateBoundingSet(candidateSet.capacity());
@@ -412,7 +410,7 @@
bool TMMBRHelp::IsOwner(const uint32_t ssrc,
const uint32_t length) const {
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
if (length == 0) {
// Empty bounding set.
@@ -428,7 +426,7 @@
}
bool TMMBRHelp::CalcMinBitRate( uint32_t* minBitrateKbit) const {
- CriticalSectionScoped lock(_criticalSection);
+ rtc::CritScope lock(&_criticalSection);
if (_candidateSet.size() == 0) {
// Empty bounding set.
diff --git a/webrtc/modules/rtp_rtcp/source/tmmbr_help.h b/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
index 6236d5d..c73d53b 100644
--- a/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
+++ b/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
@@ -12,8 +12,8 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
#include <vector>
+#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -80,7 +80,7 @@
int32_t FindTMMBRBoundingSet(int32_t numCandidates, TMMBRSet& candidateSet);
private:
- CriticalSectionWrapper* _criticalSection;
+ rtc::CriticalSection _criticalSection;
TMMBRSet _candidateSet;
TMMBRSet _boundingSet;
TMMBRSet _boundingSetToSend;