Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| index bd9314d5d768b6e2d30c5be2a4ef2287b9c35ef5..0a83944e45d1bffc8c967d5d8130b584c7119c8c 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| @@ -63,56 +63,46 @@ uint32_t ConvertMsTo24Bits(int64_t time_ms) { |
| } |
| } // namespace |
| -class BitrateAggregator { |
| - public: |
| - explicit BitrateAggregator(BitrateStatisticsObserver* bitrate_callback) |
| - : callback_(bitrate_callback), |
| - total_bitrate_observer_(*this), |
| - retransmit_bitrate_observer_(*this), |
| - ssrc_(0) {} |
| - |
| - void OnStatsUpdated() const { |
| - if (callback_) |
| - callback_->Notify(total_bitrate_observer_.statistics(), |
| - retransmit_bitrate_observer_.statistics(), |
| - ssrc_); |
| - } |
| - |
| - Bitrate::Observer* total_bitrate_observer() { |
| - return &total_bitrate_observer_; |
| - } |
| - Bitrate::Observer* retransmit_bitrate_observer() { |
| - return &retransmit_bitrate_observer_; |
| - } |
| - |
| - void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; } |
| - |
| - private: |
| - // We assume that these observers are called on the same thread, which is |
| - // true for RtpSender as they are called on the Process thread. |
| - class BitrateObserver : public Bitrate::Observer { |
| - public: |
| - explicit BitrateObserver(const BitrateAggregator& aggregator) |
| - : aggregator_(aggregator) {} |
| - |
| - // Implements Bitrate::Observer. |
| - void BitrateUpdated(const BitrateStatistics& stats) override { |
| - statistics_ = stats; |
| - aggregator_.OnStatsUpdated(); |
| - } |
| +RTPSender::BitrateAggregator::BitrateAggregator( |
| + BitrateStatisticsObserver* bitrate_callback) |
| + : callback_(bitrate_callback), |
| + total_bitrate_observer_(*this), |
| + retransmit_bitrate_observer_(*this), |
| + ssrc_(0) {} |
| - BitrateStatistics statistics() const { return statistics_; } |
| +void RTPSender::BitrateAggregator::OnStatsUpdated() const { |
| + if (callback_) { |
| + callback_->Notify(total_bitrate_observer_.statistics(), |
| + retransmit_bitrate_observer_.statistics(), ssrc_); |
| + } |
| +} |
| - private: |
| - BitrateStatistics statistics_; |
| - const BitrateAggregator& aggregator_; |
| - }; |
| +Bitrate::Observer* RTPSender::BitrateAggregator::total_bitrate_observer() { |
| + return &total_bitrate_observer_; |
| +} |
| +Bitrate::Observer* RTPSender::BitrateAggregator::retransmit_bitrate_observer() { |
| + return &retransmit_bitrate_observer_; |
| +} |
| - BitrateStatisticsObserver* const callback_; |
| - BitrateObserver total_bitrate_observer_; |
| - BitrateObserver retransmit_bitrate_observer_; |
| - uint32_t ssrc_; |
| -}; |
| +void RTPSender::BitrateAggregator::set_ssrc(uint32_t ssrc) { |
| + ssrc_ = ssrc; |
| +} |
| + |
| +RTPSender::BitrateAggregator::BitrateObserver::BitrateObserver( |
| + const BitrateAggregator& aggregator) |
| + : aggregator_(aggregator) {} |
| + |
| +// Implements Bitrate::Observer. |
| +void RTPSender::BitrateAggregator::BitrateObserver::BitrateUpdated( |
| + const BitrateStatistics& stats) { |
| + statistics_ = stats; |
| + aggregator_.OnStatsUpdated(); |
| +} |
| + |
| +const BitrateStatistics& |
| +RTPSender::BitrateAggregator::BitrateObserver::statistics() const { |
| + return statistics_; |
| +} |
| RTPSender::RTPSender( |
| bool audio, |
| @@ -132,8 +122,8 @@ RTPSender::RTPSender( |
| clock_delta_ms_(clock_->TimeInMilliseconds() - |
| TickTime::MillisecondTimestamp()), |
| random_(clock_->TimeInMicroseconds()), |
| - bitrates_(new BitrateAggregator(bitrate_callback)), |
| - total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()), |
| + bitrates_(bitrate_callback), |
| + total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()), |
| audio_configured_(audio), |
| audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr), |
| video_(audio ? nullptr : new RTPSenderVideo(clock, this)), |
| @@ -141,7 +131,6 @@ RTPSender::RTPSender( |
| transport_sequence_number_allocator_(sequence_number_allocator), |
| transport_feedback_observer_(transport_feedback_observer), |
| last_capture_time_ms_sent_(0), |
| - send_critsect_(CriticalSectionWrapper::CreateCriticalSection()), |
| transport_(transport), |
| sending_media_(true), // Default to sending media. |
| max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP. |
| @@ -157,7 +146,7 @@ RTPSender::RTPSender( |
| // NACK. |
| nack_byte_count_times_(), |
| nack_byte_count_(), |
| - nack_bitrate_(clock, bitrates_->retransmit_bitrate_observer()), |
| + nack_bitrate_(clock, bitrates_.retransmit_bitrate_observer()), |
| packet_history_(clock), |
| // Statistics |
| statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| @@ -168,7 +157,7 @@ RTPSender::RTPSender( |
| // RTP variables |
| start_timestamp_forced_(false), |
| start_timestamp_(0), |
| - ssrc_db_(*SSRCDatabase::GetSSRCDatabase()), |
| + ssrc_db_(SSRCDatabase::GetSSRCDatabase()), |
| remote_ssrc_(0), |
| sequence_number_forced_(false), |
| ssrc_forced_(false), |
| @@ -184,21 +173,35 @@ RTPSender::RTPSender( |
| target_bitrate_(0) { |
| memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_)); |
| memset(nack_byte_count_, 0, sizeof(nack_byte_count_)); |
| - // We need to seed the random generator. |
| + // We need to seed the random generator for BuildPaddingPacket() below. |
| + // TODO(holmer,tommi): Note that TimeInMilliseconds might return 0 on Mac |
| + // early on in the process. |
| srand(static_cast<uint32_t>(clock_->TimeInMilliseconds())); |
| - ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0. |
| - ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0. |
| - bitrates_->set_ssrc(ssrc_); |
| + ssrc_ = ssrc_db_->CreateSSRC(); |
| + RTC_DCHECK(ssrc_); |
| + ssrc_rtx_ = ssrc_db_->CreateSSRC(); |
| + RTC_DCHECK(ssrc_rtx_); |
| + |
| + bitrates_.set_ssrc(ssrc_); |
| // Random start, 16 bits. Can't be 0. |
| sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber); |
| sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber); |
| } |
| RTPSender::~RTPSender() { |
| + // 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 |
| + |
| + // TODO(tommi,holmer): We don't grab locks in the dtor before accessing member |
| + // 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(remote_ssrc_); |
| } |
| - ssrc_db_.ReturnSSRC(ssrc_); |
| + ssrc_db_->ReturnSSRC(ssrc_); |
| SSRCDatabase::ReturnSSRCDatabase(); |
| while (!payload_type_map_.empty()) { |
| @@ -246,7 +249,7 @@ int32_t RTPSender::SetTransmissionTimeOffset(int32_t transmission_time_offset) { |
| transmission_time_offset < -(0x800000 - 1)) { // Word24. |
| return -1; |
| } |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| transmission_time_offset_ = transmission_time_offset; |
| return 0; |
| } |
| @@ -255,25 +258,25 @@ int32_t RTPSender::SetAbsoluteSendTime(uint32_t absolute_send_time) { |
| if (absolute_send_time > 0xffffff) { // UWord24. |
| return -1; |
| } |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| absolute_send_time_ = absolute_send_time; |
| return 0; |
| } |
| void RTPSender::SetVideoRotation(VideoRotation rotation) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| rotation_ = rotation; |
| } |
| int32_t RTPSender::SetTransportSequenceNumber(uint16_t sequence_number) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| transport_sequence_number_ = sequence_number; |
| return 0; |
| } |
| int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type, |
| uint8_t id) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (type == kRtpExtensionVideoRotation) { |
| cvo_mode_ = kCVOInactive; |
| return rtp_header_extension_map_.RegisterInactive(type, id); |
| @@ -282,17 +285,17 @@ int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type, |
| } |
| bool RTPSender::IsRtpHeaderExtensionRegistered(RTPExtensionType type) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return rtp_header_extension_map_.IsRegistered(type); |
| } |
| int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return rtp_header_extension_map_.Deregister(type); |
| } |
| size_t RTPSender::RtpHeaderExtensionTotalLength() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return rtp_header_extension_map_.GetTotalLengthInBytes(); |
| } |
| @@ -303,7 +306,7 @@ int32_t RTPSender::RegisterPayload( |
| size_t channels, |
| uint32_t rate) { |
| assert(payload_name); |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| std::map<int8_t, RtpUtility::Payload*>::iterator it = |
| payload_type_map_.find(payload_number); |
| @@ -346,7 +349,7 @@ int32_t RTPSender::RegisterPayload( |
| } |
| int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| std::map<int8_t, RtpUtility::Payload*>::iterator it = |
| payload_type_map_.find(payload_type); |
| @@ -361,12 +364,12 @@ int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) { |
| } |
| void RTPSender::SetSendPayloadType(int8_t payload_type) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| payload_type_ = payload_type; |
| } |
| int8_t RTPSender::SendPayloadType() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return payload_type_; |
| } |
| @@ -379,7 +382,7 @@ int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length, |
| // Sanity check. |
| RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE) |
| << "Invalid max payload length: " << max_payload_length; |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| max_payload_length_ = max_payload_length; |
| packet_over_head_ = packet_over_head; |
| return 0; |
| @@ -388,7 +391,7 @@ int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length, |
| size_t RTPSender::MaxDataPayloadLength() const { |
| int rtx; |
| { |
| - CriticalSectionScoped rtx_lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| rtx = rtx_; |
| } |
| if (audio_configured_) { |
| @@ -407,28 +410,28 @@ size_t RTPSender::MaxPayloadLength() const { |
| uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; } |
| void RTPSender::SetRtxStatus(int mode) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| rtx_ = mode; |
| } |
| int RTPSender::RtxStatus() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return rtx_; |
| } |
| void RTPSender::SetRtxSsrc(uint32_t ssrc) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| ssrc_rtx_ = ssrc; |
| } |
| uint32_t RTPSender::RtxSsrc() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return ssrc_rtx_; |
| } |
| void RTPSender::SetRtxPayloadType(int payload_type, |
| int associated_payload_type) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| RTC_DCHECK_LE(payload_type, 127); |
| RTC_DCHECK_LE(associated_payload_type, 127); |
| if (payload_type < 0) { |
| @@ -441,7 +444,7 @@ void RTPSender::SetRtxPayloadType(int payload_type, |
| } |
| std::pair<int, int> RTPSender::RtxPayloadType() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| for (const auto& kv : rtx_payload_type_map_) { |
| if (kv.second == rtx_payload_type_) { |
| return std::make_pair(rtx_payload_type_, kv.first); |
| @@ -452,7 +455,7 @@ std::pair<int, int> RTPSender::RtxPayloadType() const { |
| int32_t RTPSender::CheckPayloadType(int8_t payload_type, |
| RtpVideoCodecTypes* video_type) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (payload_type < 0) { |
| LOG(LS_ERROR) << "Invalid payload_type " << payload_type; |
| @@ -494,7 +497,7 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type, |
| RTPSenderInterface::CVOMode RTPSender::ActivateCVORtpHeaderExtension() { |
| if (cvo_mode_ == kCVOInactive) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (rtp_header_extension_map_.SetActive(kRtpExtensionVideoRotation, true)) { |
| cvo_mode_ = kCVOActivated; |
| } |
| @@ -513,7 +516,7 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type, |
| uint32_t ssrc; |
| { |
| // Drop this packet if we're not sending media packets. |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| ssrc = ssrc_; |
| if (!sending_media_) { |
| return 0; |
| @@ -565,7 +568,7 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type, |
| size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) { |
| { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if ((rtx_ & kRtxRedundantPayloads) == 0) |
| return 0; |
| } |
| @@ -627,7 +630,7 @@ size_t RTPSender::SendPadData(size_t bytes, |
| int payload_type; |
| bool over_rtx; |
| { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (!timestamp_provided) { |
| timestamp = timestamp_; |
| capture_time_ms = capture_time_ms_; |
| @@ -741,7 +744,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) { |
| } |
| int rtx = kRtxOff; |
| { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| rtx = rtx_; |
| } |
| if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms, |
| @@ -839,7 +842,7 @@ bool RTPSender::ProcessNACKBitRate(uint32_t now) { |
| const uint32_t kAvgIntervalMs = 1000; |
| uint32_t target_bitrate = GetTargetBitrate(); |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (target_bitrate == 0) { |
| return true; |
| @@ -864,7 +867,7 @@ bool RTPSender::ProcessNACKBitRate(uint32_t now) { |
| } |
| void RTPSender::UpdateNACKBitRate(uint32_t bytes, int64_t now) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (bytes == 0) |
| return; |
| nack_bitrate_.Update(bytes); |
| @@ -900,7 +903,7 @@ bool RTPSender::TimeToSendPacket(uint16_t sequence_number, |
| } |
| int rtx; |
| { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| rtx = rtx_; |
| } |
| return PrepareAndSendPacket(data_buffer, |
| @@ -958,7 +961,7 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, |
| bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options); |
| if (ret) { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| media_has_been_sent_ = true; |
| } |
| UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx, |
| @@ -1018,7 +1021,7 @@ size_t RTPSender::TimeToSendPadding(size_t bytes) { |
| if (audio_configured_ || bytes == 0) |
| return 0; |
| { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (!sending_media_) |
| return 0; |
| } |
| @@ -1090,7 +1093,7 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer, |
| return -1; |
| { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| media_has_been_sent_ = true; |
| } |
| UpdateRtpStats(buffer, length, rtp_header, false, false); |
| @@ -1105,7 +1108,7 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) { |
| int avg_delay_ms = 0; |
| int max_delay_ms = 0; |
| { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| ssrc = ssrc_; |
| } |
| { |
| @@ -1131,7 +1134,7 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) { |
| } |
| void RTPSender::ProcessBitrate() { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| total_bitrate_sent_.Process(); |
| nack_bitrate_.Process(); |
| if (audio_configured_) { |
| @@ -1141,7 +1144,7 @@ void RTPSender::ProcessBitrate() { |
| } |
| size_t RTPSender::RTPHeaderLength() const { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| size_t rtp_header_length = kRtpHeaderLength; |
| rtp_header_length += sizeof(uint32_t) * csrcs_.size(); |
| rtp_header_length += RtpHeaderExtensionTotalLength(); |
| @@ -1149,7 +1152,7 @@ size_t RTPSender::RTPHeaderLength() const { |
| } |
| uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| uint16_t first_allocated_sequence_number = sequence_number_; |
| sequence_number_ += packets_to_send; |
| return first_allocated_sequence_number; |
| @@ -1208,7 +1211,7 @@ int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer, |
| bool timestamp_provided, |
| bool inc_sequence_number) { |
| assert(payload_type >= 0); |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (timestamp_provided) { |
| timestamp_ = start_timestamp_ + capture_timestamp; |
| @@ -1515,7 +1518,7 @@ void RTPSender::UpdateTransmissionTimeOffset(uint8_t* rtp_packet, |
| const RTPHeader& rtp_header, |
| int64_t time_diff_ms) const { |
| size_t offset; |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| switch (VerifyExtension(kRtpExtensionTransmissionTimeOffset, rtp_packet, |
| rtp_packet_length, rtp_header, |
| kTransmissionTimeOffsetLength, &offset)) { |
| @@ -1541,7 +1544,7 @@ bool RTPSender::UpdateAudioLevel(uint8_t* rtp_packet, |
| bool is_voiced, |
| uint8_t dBov) const { |
| size_t offset; |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| switch (VerifyExtension(kRtpExtensionAudioLevel, rtp_packet, |
| rtp_packet_length, rtp_header, kAudioLevelLength, |
| @@ -1566,7 +1569,7 @@ bool RTPSender::UpdateVideoRotation(uint8_t* rtp_packet, |
| const RTPHeader& rtp_header, |
| VideoRotation rotation) const { |
| size_t offset; |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| switch (VerifyExtension(kRtpExtensionVideoRotation, rtp_packet, |
| rtp_packet_length, rtp_header, kVideoRotationLength, |
| @@ -1591,7 +1594,7 @@ void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet, |
| const RTPHeader& rtp_header, |
| int64_t now_ms) const { |
| size_t offset; |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| switch (VerifyExtension(kRtpExtensionAbsoluteSendTime, rtp_packet, |
| rtp_packet_length, rtp_header, |
| @@ -1618,7 +1621,7 @@ uint16_t RTPSender::UpdateTransportSequenceNumber( |
| size_t rtp_packet_length, |
| const RTPHeader& rtp_header) const { |
| size_t offset; |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| switch (VerifyExtension(kRtpExtensionTransportSequenceNumber, rtp_packet, |
| rtp_packet_length, rtp_header, |
| @@ -1647,12 +1650,13 @@ void RTPSender::SetSendingStatus(bool enabled) { |
| // Will be ignored if it's already configured via API. |
| SetStartTimestamp(RTPtime, false); |
| } else { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (!ssrc_forced_) { |
| // Generate a new SSRC. |
| - ssrc_db_.ReturnSSRC(ssrc_); |
| - ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0. |
| - bitrates_->set_ssrc(ssrc_); |
| + ssrc_db_->ReturnSSRC(ssrc_); |
| + ssrc_ = ssrc_db_->CreateSSRC(); |
| + RTC_DCHECK(ssrc_); |
|
stefan-webrtc
2016/01/26 13:19:52
I prefer > 0 or != 0.
tommi
2016/01/30 10:53:41
Done.
|
| + bitrates_.set_ssrc(ssrc_); |
| } |
| // Don't initialize seq number if SSRC passed externally. |
| if (!sequence_number_forced_ && !ssrc_forced_) { |
| @@ -1663,22 +1667,22 @@ void RTPSender::SetSendingStatus(bool enabled) { |
| } |
| void RTPSender::SetSendingMediaStatus(bool enabled) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| sending_media_ = enabled; |
| } |
| bool RTPSender::SendingMedia() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return sending_media_; |
| } |
| uint32_t RTPSender::Timestamp() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return timestamp_; |
| } |
| void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (force) { |
| start_timestamp_forced_ = true; |
| start_timestamp_ = timestamp; |
| @@ -1690,58 +1694,59 @@ void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) { |
| } |
| uint32_t RTPSender::StartTimestamp() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return start_timestamp_; |
| } |
| uint32_t RTPSender::GenerateNewSSRC() { |
| // If configured via API, return 0. |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (ssrc_forced_) { |
| return 0; |
| } |
| - ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0. |
| - bitrates_->set_ssrc(ssrc_); |
| + ssrc_ = ssrc_db_->CreateSSRC(); |
| + RTC_DCHECK(ssrc_); |
|
stefan-webrtc
2016/01/26 13:19:52
same here
tommi
2016/01/30 10:53:41
Done.
|
| + bitrates_.set_ssrc(ssrc_); |
| return ssrc_; |
| } |
| void RTPSender::SetSSRC(uint32_t ssrc) { |
| // This is configured via the API. |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| if (ssrc_ == ssrc && ssrc_forced_) { |
| return; // Since it's same ssrc, don't reset anything. |
| } |
| ssrc_forced_ = true; |
| - ssrc_db_.ReturnSSRC(ssrc_); |
| - ssrc_db_.RegisterSSRC(ssrc); |
| + ssrc_db_->ReturnSSRC(ssrc_); |
| + ssrc_db_->RegisterSSRC(ssrc); |
| ssrc_ = ssrc; |
| - bitrates_->set_ssrc(ssrc_); |
| + bitrates_.set_ssrc(ssrc_); |
| if (!sequence_number_forced_) { |
| sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber); |
| } |
| } |
| uint32_t RTPSender::SSRC() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return ssrc_; |
| } |
| void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) { |
| assert(csrcs.size() <= kRtpCsrcSize); |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| csrcs_ = csrcs; |
| } |
| void RTPSender::SetSequenceNumber(uint16_t seq) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| sequence_number_forced_ = true; |
| sequence_number_ = seq; |
| } |
| uint16_t RTPSender::SequenceNumber() const { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| return sequence_number_; |
| } |
| @@ -1818,7 +1823,7 @@ int32_t RTPSender::SetFecParameters( |
| void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length, |
| uint8_t* buffer_rtx) { |
| - CriticalSectionScoped cs(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| uint8_t* data_buffer_rtx = buffer_rtx; |
| // Add RTX header. |
| RtpUtility::RtpHeaderParser rtp_parser( |
| @@ -1873,7 +1878,7 @@ uint32_t RTPSender::BitrateSent() const { |
| void RTPSender::SetRtpState(const RtpState& rtp_state) { |
| SetStartTimestamp(rtp_state.start_timestamp, true); |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| sequence_number_ = rtp_state.sequence_number; |
| sequence_number_forced_ = true; |
| timestamp_ = rtp_state.timestamp; |
| @@ -1883,7 +1888,7 @@ void RTPSender::SetRtpState(const RtpState& rtp_state) { |
| } |
| RtpState RTPSender::GetRtpState() const { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| RtpState state; |
| state.sequence_number = sequence_number_; |
| @@ -1897,12 +1902,12 @@ RtpState RTPSender::GetRtpState() const { |
| } |
| void RTPSender::SetRtxRtpState(const RtpState& rtp_state) { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| sequence_number_rtx_ = rtp_state.sequence_number; |
| } |
| RtpState RTPSender::GetRtxRtpState() const { |
| - CriticalSectionScoped lock(send_critsect_.get()); |
| + rtc::CritScope lock(&send_critsect_); |
| RtpState state; |
| state.sequence_number = sequence_number_rtx_; |