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Delete dead code in RtpReceiverImpl and RTPPayloadRegistry.

Browse Source 
Bug: webrtc:8995
Change-Id: I5460c699c2dc6cf17b2f88be74698b913d4c29b8
Reviewed-on: https://webrtc-review.googlesource.com/64447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22607}
This commit is contained in:
Niels Möller 2018-03-23 15:56:57 +01:00 committed by Commit Bot
parent 250155d0db
commit e08a4c01b9
6 changed files with 1 additions and 90 deletions
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10
modules/rtp_rtcp/include/rtp_payload_registry.h
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@ -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<int, RtpUtility::Payload> 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.

16
modules/rtp_rtcp/source/rtp_payload_registry.cc
View File

@ -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<RtpUtility::Payload> RTPPayloadRegistry::PayloadTypeToPayload(
: rtc::Optional<RtpUtility::Payload>(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

7
modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
View File

@ -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

39
modules/rtp_rtcp/source/rtp_receiver_impl.cc
View File

@ -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<AudioPayload> 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.

14
modules/rtp_rtcp/source/rtp_receiver_strategy.cc
View File

@ -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) {

5
modules/rtp_rtcp/source/rtp_receiver_strategy.h
View File

@ -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<PayloadUnion> last_payload_;
RtpData* data_callback_;
};
} // namespace webrtc