From e08a4c01b92a5d660237593c13107da298b695e1 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Niels=20M=C3=B6ller?= Date: Fri, 23 Mar 2018 15:56:57 +0100 Subject: [PATCH] Delete dead code in RtpReceiverImpl and RTPPayloadRegistry. Bug: webrtc:8995 Change-Id: I5460c699c2dc6cf17b2f88be74698b913d4c29b8 Reviewed-on: https://webrtc-review.googlesource.com/64447 Reviewed-by: Danil Chapovalov Commit-Queue: Niels Moller Cr-Commit-Position: refs/heads/master@{#22607} --- .../rtp_rtcp/include/rtp_payload_registry.h | 10 ----- .../rtp_rtcp/source/rtp_payload_registry.cc | 16 +------- .../source/rtp_payload_registry_unittest.cc | 7 ---- modules/rtp_rtcp/source/rtp_receiver_impl.cc | 39 ------------------- .../rtp_rtcp/source/rtp_receiver_strategy.cc | 14 ------- .../rtp_rtcp/source/rtp_receiver_strategy.h | 5 --- 6 files changed, 1 insertion(+), 90 deletions(-) diff --git a/modules/rtp_rtcp/include/rtp_payload_registry.h b/modules/rtp_rtcp/include/rtp_payload_registry.h index fc2b5fb403..928d158198 100644 --- a/modules/rtp_rtcp/include/rtp_payload_registry.h +++ b/modules/rtp_rtcp/include/rtp_payload_registry.h @@ -54,12 +54,8 @@ class RTPPayloadRegistry { void ResetLastReceivedPayloadTypes() { rtc::CritScope cs(&crit_sect_); last_received_payload_type_ = -1; - last_received_media_payload_type_ = -1; } - // Returns true if the new media payload type has not changed. - bool ReportMediaPayloadType(uint8_t media_payload_type); - int8_t last_received_payload_type() const { rtc::CritScope cs(&crit_sect_); return last_received_payload_type_; @@ -69,11 +65,6 @@ class RTPPayloadRegistry { last_received_payload_type_ = last_received_payload_type; } - int8_t last_received_media_payload_type() const { - rtc::CritScope cs(&crit_sect_); - return last_received_media_payload_type_; - } - private: // Prunes the payload type map of the specific payload type, if it exists. void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( @@ -82,7 +73,6 @@ class RTPPayloadRegistry { rtc::CriticalSection crit_sect_; std::map payload_type_map_; int8_t last_received_payload_type_; - int8_t last_received_media_payload_type_; // As a first step in splitting this class up in separate cases for audio and // video, DCHECK that no instance is used for both audio and video. diff --git a/modules/rtp_rtcp/source/rtp_payload_registry.cc b/modules/rtp_rtcp/source/rtp_payload_registry.cc index 5f9c1f940a..46a4ec17f1 100644 --- a/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/modules/rtp_rtcp/source/rtp_payload_registry.cc @@ -103,8 +103,7 @@ bool IsPayloadTypeValid(int8_t payload_type) { } // namespace RTPPayloadRegistry::RTPPayloadRegistry() - : last_received_payload_type_(-1), - last_received_media_payload_type_(-1) {} + : last_received_payload_type_(-1) {} RTPPayloadRegistry::~RTPPayloadRegistry() = default; @@ -129,7 +128,6 @@ void RTPPayloadRegistry::SetAudioReceivePayloads( // Clear the value of last received payload type since it might mean // something else now. last_received_payload_type_ = -1; - last_received_media_payload_type_ = -1; } int32_t RTPPayloadRegistry::RegisterReceivePayload( @@ -170,7 +168,6 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( // Successful set of payload type, clear the value of last received payload // type since it might mean something else. last_received_payload_type_ = -1; - last_received_media_payload_type_ = -1; return 0; } @@ -204,7 +201,6 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( // Successful set of payload type, clear the value of last received payload // type since it might mean something else. last_received_payload_type_ = -1; - last_received_media_payload_type_ = -1; return 0; } @@ -280,14 +276,4 @@ rtc::Optional RTPPayloadRegistry::PayloadTypeToPayload( : rtc::Optional(it->second); } -bool RTPPayloadRegistry::ReportMediaPayloadType(uint8_t media_payload_type) { - rtc::CritScope cs(&crit_sect_); - if (last_received_media_payload_type_ == media_payload_type) { - // Media type unchanged. - return true; - } - last_received_media_payload_type_ = media_payload_type; - return false; -} - } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc index 4e0434fc99..ac1afd6371 100644 --- a/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc @@ -155,11 +155,6 @@ TEST(RtpPayloadRegistryTest, rtp_payload_registry.set_last_received_payload_type(17); EXPECT_EQ(17, rtp_payload_registry.last_received_payload_type()); - bool media_type_unchanged = rtp_payload_registry.ReportMediaPayloadType(18); - EXPECT_FALSE(media_type_unchanged); - media_type_unchanged = rtp_payload_registry.ReportMediaPayloadType(18); - EXPECT_TRUE(media_type_unchanged); - bool ignored; constexpr int payload_type = 34; const SdpAudioFormat audio_format("name", 44000, 1); @@ -167,8 +162,6 @@ TEST(RtpPayloadRegistryTest, payload_type, audio_format, &ignored)); EXPECT_EQ(-1, rtp_payload_registry.last_received_payload_type()); - media_type_unchanged = rtp_payload_registry.ReportMediaPayloadType(18); - EXPECT_FALSE(media_type_unchanged); } class ParameterizedRtpPayloadRegistryTest diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc index 425973cfe2..1fd45bd3ff 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -272,8 +272,6 @@ bool RtpReceiverImpl::GetLatestTimestamps(uint32_t* timestamp, // Implementation note: must not hold critsect when called. void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) { bool new_ssrc = false; - // TODO(nisse): This is unused, delete any corresponding dead code. - rtc::Optional reinitialize_audio_payload; { rtc::CritScope lock(&critical_section_rtp_receiver_); @@ -288,23 +286,6 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) { last_received_timestamp_ = 0; last_received_frame_time_ms_ = -1; - // Do we have a SSRC? Then the stream is restarted. - if (ssrc_ != 0) { - // Do we have the same codec? Then re-initialize coder. - if (rtp_header.payloadType == last_received_payload_type) { - const auto payload = rtp_payload_registry_->PayloadTypeToPayload( - rtp_header.payloadType); - if (!payload) { - return; - } - if (payload->typeSpecific.is_audio()) { - reinitialize_audio_payload.emplace( - payload->typeSpecific.audio_payload()); - } else { - // OnInitializeDecoder() is only used for audio. - } - } - } ssrc_ = rtp_header.ssrc; } } @@ -326,10 +307,6 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) { int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header, const int8_t first_payload_byte, PayloadUnion* specific_payload) { - // TODO(nisse): re_initialize_decoder is unused, and most or all of - // this code can likely be deleted. - bool re_initialize_decoder = false; - int8_t payload_type = rtp_header.payloadType; { @@ -357,22 +334,6 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header, return -1; } rtp_payload_registry_->set_last_received_payload_type(payload_type); - - re_initialize_decoder = true; - - rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific); - rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload); - - if (!payload->typeSpecific.is_audio()) { - bool media_type_unchanged = - rtp_payload_registry_->ReportMediaPayloadType(payload_type); - if (media_type_unchanged) { - // Only reset the decoder if the media codec type has changed. - re_initialize_decoder = false; - } - } - } else { - rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload); } } // End critsect. diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/modules/rtp_rtcp/source/rtp_receiver_strategy.cc index b5a6356ed7..1404273865 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_strategy.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.cc @@ -19,20 +19,6 @@ RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback) RTPReceiverStrategy::~RTPReceiverStrategy() = default; -void RTPReceiverStrategy::GetLastMediaSpecificPayload( - PayloadUnion* payload) const { - rtc::CritScope cs(&crit_sect_); - if (last_payload_) { - *payload = *last_payload_; - } -} - -void RTPReceiverStrategy::SetLastMediaSpecificPayload( - const PayloadUnion& payload) { - rtc::CritScope cs(&crit_sect_); - last_payload_.emplace(payload); -} - void RTPReceiverStrategy::CheckPayloadChanged(int8_t payload_type, PayloadUnion* specific_payload, bool* should_discard_changes) { diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/modules/rtp_rtcp/source/rtp_receiver_strategy.h index 76bffff183..d5c51a278c 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_strategy.h +++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.h @@ -67,10 +67,6 @@ class RTPReceiverStrategy { virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const; - // Stores / retrieves the last media specific payload for later reference. - void GetLastMediaSpecificPayload(PayloadUnion* payload) const; - void SetLastMediaSpecificPayload(const PayloadUnion& payload); - protected: // The data callback is where we should send received payload data. // See ParseRtpPacket. This class does not claim ownership of the callback. @@ -83,7 +79,6 @@ class RTPReceiverStrategy { explicit RTPReceiverStrategy(RtpData* data_callback); rtc::CriticalSection crit_sect_; - rtc::Optional last_payload_; RtpData* data_callback_; }; } // namespace webrtc