From 3ed46bd83b29157a66f53c26eaa39f7b216d65d2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Niels=20M=C3=B6ller?= Date: Mon, 6 Aug 2018 17:12:27 +0200 Subject: [PATCH] Delete RTPReceiverStrategy::OnNewPayloadTypeCreated and related code. Bug: webrtc:7135 Change-Id: Ic20d98cbfb8154f5abbc2501cbcccb950148e732 Reviewed-on: https://webrtc-review.googlesource.com/92396 Reviewed-by: Danil Chapovalov Reviewed-by: Oskar Sundbom Commit-Queue: Niels Moller Cr-Commit-Position: refs/heads/master@{#24219} --- modules/rtp_rtcp/include/rtp_rtcp_defines.h | 2 - modules/rtp_rtcp/source/rtp_receiver_audio.cc | 57 +------------------ modules/rtp_rtcp/source/rtp_receiver_audio.h | 18 ------ modules/rtp_rtcp/source/rtp_receiver_impl.cc | 8 --- .../rtp_rtcp/source/rtp_receiver_strategy.h | 10 ---- modules/rtp_rtcp/source/rtp_receiver_video.cc | 12 ---- modules/rtp_rtcp/source/rtp_receiver_video.h | 5 -- 7 files changed, 1 insertion(+), 111 deletions(-) diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h index 69e800e313..260e46170b 100644 --- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h +++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h @@ -89,8 +89,6 @@ class PayloadUnion { absl::variant payload_; }; -enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 }; - enum ProtectionType { kUnprotectedPacket, kProtectedPacket }; enum StorageType { kDontRetransmit, kAllowRetransmission }; diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/modules/rtp_rtcp/source/rtp_receiver_audio.cc index eac50e2877..030c79f2ca 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -25,28 +25,10 @@ RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy( } RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback) - : RTPReceiverStrategy(data_callback), - telephone_event_payload_type_(-1), - cng_nb_payload_type_(-1), - cng_wb_payload_type_(-1), - cng_swb_payload_type_(-1), - cng_fb_payload_type_(-1) {} + : RTPReceiverStrategy(data_callback) {} RTPReceiverAudio::~RTPReceiverAudio() = default; -bool RTPReceiverAudio::TelephoneEventPayloadType(int8_t payload_type) const { - rtc::CritScope lock(&crit_sect_); - return telephone_event_payload_type_ == payload_type; -} - -bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type) { - rtc::CritScope lock(&crit_sect_); - return payload_type == cng_nb_payload_type_ || - payload_type == cng_wb_payload_type_ || - payload_type == cng_swb_payload_type_ || - payload_type == cng_fb_payload_type_; -} - // - Sample based or frame based codecs based on RFC 3551 // - // - NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples. @@ -79,32 +61,6 @@ bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type) { // - MPA frame N/A var. var. // - // - G7221 frame N/A -int32_t RTPReceiverAudio::OnNewPayloadTypeCreated( - int payload_type, - const SdpAudioFormat& audio_format) { - rtc::CritScope lock(&crit_sect_); - - if (RtpUtility::StringCompare(audio_format.name.c_str(), "telephone-event", - 15)) { - telephone_event_payload_type_ = payload_type; - } - if (RtpUtility::StringCompare(audio_format.name.c_str(), "cn", 2)) { - // We support comfort noise at four different frequencies. - if (audio_format.clockrate_hz == 8000) { - cng_nb_payload_type_ = payload_type; - } else if (audio_format.clockrate_hz == 16000) { - cng_wb_payload_type_ = payload_type; - } else if (audio_format.clockrate_hz == 32000) { - cng_swb_payload_type_ = payload_type; - } else if (audio_format.clockrate_hz == 48000) { - cng_fb_payload_type_ = payload_type; - } else { - assert(false); - return -1; - } - } - return 0; -} int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, const PayloadUnion& specific_payload, @@ -119,17 +75,6 @@ int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, specific_payload.audio_payload()); } -RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive( - uint16_t last_payload_length) const { - // Our CNG is 9 bytes; if it's a likely CNG the receiver needs to check - // kRtpNoRtp against NetEq speech_type kOutputPLCtoCNG. - if (last_payload_length < 10) { // our CNG is 9 bytes - return kRtpNoRtp; - } else { - return kRtpDead; - } -} - // We are not allowed to have any critsects when calling data_callback. int32_t RTPReceiverAudio::ParseAudioCodecSpecific( WebRtcRTPHeader* rtp_header, diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.h b/modules/rtp_rtcp/source/rtp_receiver_audio.h index b4304bc307..5d97a1f3a9 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/modules/rtp_rtcp/source/rtp_receiver_audio.h @@ -27,36 +27,18 @@ class RTPReceiverAudio : public RTPReceiverStrategy { explicit RTPReceiverAudio(RtpData* data_callback); ~RTPReceiverAudio() override; - // Returns true if CNG is configured with |payload_type|. - bool CNGPayloadType(const int8_t payload_type); - int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, const PayloadUnion& specific_payload, const uint8_t* packet, size_t payload_length, int64_t timestamp_ms) override; - RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override; - - int32_t OnNewPayloadTypeCreated(int payload_type, - const SdpAudioFormat& audio_format) override; - private: int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_length, const AudioPayload& audio_specific); - // Is TelephoneEvent configured with |payload_type|. - bool TelephoneEventPayloadType(const int8_t payload_type) const; - - int8_t telephone_event_payload_type_; - - int8_t cng_nb_payload_type_; - int8_t cng_wb_payload_type_; - int8_t cng_swb_payload_type_; - int8_t cng_fb_payload_type_; - ThreadUnsafeOneTimeEvent first_packet_received_; }; } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc index 54ac6ea854..1646444bb6 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -105,14 +105,6 @@ int32_t RtpReceiverImpl::RegisterReceivePayload( bool created_new_payload = false; int32_t result = rtp_payload_registry_->RegisterReceivePayload( payload_type, audio_format, &created_new_payload); - if (created_new_payload) { - if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_type, - audio_format) != 0) { - RTC_LOG(LS_ERROR) << "Failed to register payload: " << audio_format.name - << "/" << payload_type; - return -1; - } - } return result; } diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/modules/rtp_rtcp/source/rtp_receiver_strategy.h index fc3ee8d22d..987af7c41a 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_strategy.h +++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.h @@ -40,16 +40,6 @@ class RTPReceiverStrategy { size_t payload_length, int64_t timestamp_ms) = 0; - // Computes the current dead-or-alive state. - virtual RTPAliveType ProcessDeadOrAlive( - uint16_t last_payload_length) const = 0; - - // Notifies the strategy that we have created a new non-RED audio payload type - // in the payload registry. - virtual int32_t OnNewPayloadTypeCreated( - int payload_type, - const SdpAudioFormat& audio_format) = 0; - protected: // The data callback is where we should send received payload data. // See ParseRtpPacket. This class does not claim ownership of the callback. diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.cc b/modules/rtp_rtcp/source/rtp_receiver_video.cc index 62e93e9e13..1101becfb1 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_video.cc @@ -36,13 +36,6 @@ RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback) RTPReceiverVideo::~RTPReceiverVideo() {} -int32_t RTPReceiverVideo::OnNewPayloadTypeCreated( - int payload_type, - const SdpAudioFormat& audio_format) { - RTC_NOTREACHED(); - return 0; -} - int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, const PayloadUnion& specific_payload, const uint8_t* payload, @@ -108,9 +101,4 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, : -1; } -RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive( - uint16_t last_payload_length) const { - return kRtpDead; -} - } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.h b/modules/rtp_rtcp/source/rtp_receiver_video.h index 58ba3d0b0c..46b97f5033 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_video.h +++ b/modules/rtp_rtcp/source/rtp_receiver_video.h @@ -30,11 +30,6 @@ class RTPReceiverVideo : public RTPReceiverStrategy { size_t packet_length, int64_t timestamp) override; - RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override; - - int32_t OnNewPayloadTypeCreated(int payload_type, - const SdpAudioFormat& audio_format) override; - void SetPacketOverHead(uint16_t packet_over_head); private: