From dc6e68b4a7b9caa635a1046ec88f5fc50c962b85 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Niels=20M=C3=B6ller?= Date: Thu, 2 Aug 2018 15:13:29 +0200 Subject: [PATCH] Delete class TelephoneEventHandler and related code. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Followup to https://webrtc-review.googlesource.com/91125. Bug: webrtc:7135 Change-Id: I7011cc65ac756931d8134763da57ec1bc9c584d6 Reviewed-on: https://webrtc-review.googlesource.com/91163 Reviewed-by: Åsa Persson Commit-Queue: Niels Moller Cr-Commit-Position: refs/heads/master@{#24174} --- modules/rtp_rtcp/include/rtp_receiver.h | 11 ----------- modules/rtp_rtcp/source/rtp_receiver_audio.cc | 5 ----- modules/rtp_rtcp/source/rtp_receiver_audio.h | 11 ++++------- modules/rtp_rtcp/source/rtp_receiver_impl.cc | 4 ---- modules/rtp_rtcp/source/rtp_receiver_impl.h | 2 -- modules/rtp_rtcp/source/rtp_receiver_strategy.h | 4 ---- modules/rtp_rtcp/source/rtp_receiver_video.cc | 4 ---- modules/rtp_rtcp/source/rtp_receiver_video.h | 2 -- 8 files changed, 4 insertions(+), 39 deletions(-) diff --git a/modules/rtp_rtcp/include/rtp_receiver.h b/modules/rtp_rtcp/include/rtp_receiver.h index 3868c61eb5..8e2737d93c 100644 --- a/modules/rtp_rtcp/include/rtp_receiver.h +++ b/modules/rtp_rtcp/include/rtp_receiver.h @@ -21,14 +21,6 @@ namespace webrtc { class RTPPayloadRegistry; class VideoCodec; -class TelephoneEventHandler { - public: - virtual ~TelephoneEventHandler() {} - - // Is TelephoneEvent configured with payload type payload_type - virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0; -}; - class RtpReceiver { public: // Creates a video-enabled RTP receiver. @@ -45,9 +37,6 @@ class RtpReceiver { virtual ~RtpReceiver() {} - // Returns a TelephoneEventHandler if available. - virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; - // Registers a receive payload in the payload registry and notifies the media // receiver strategy. virtual int32_t RegisterReceivePayload( diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/modules/rtp_rtcp/source/rtp_receiver_audio.cc index 22d5255fa8..434a53e0e6 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -26,7 +26,6 @@ RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy( RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback) : RTPReceiverStrategy(data_callback), - TelephoneEventHandler(), telephone_event_payload_type_(-1), cng_nb_payload_type_(-1), cng_wb_payload_type_(-1), @@ -40,10 +39,6 @@ bool RTPReceiverAudio::TelephoneEventPayloadType(int8_t payload_type) const { return telephone_event_payload_type_ == payload_type; } -TelephoneEventHandler* RTPReceiverAudio::GetTelephoneEventHandler() { - return this; -} - bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type) { rtc::CritScope lock(&crit_sect_); return payload_type == cng_nb_payload_type_ || diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.h b/modules/rtp_rtcp/source/rtp_receiver_audio.h index 3f779f6f0f..e582446ae8 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/modules/rtp_rtcp/source/rtp_receiver_audio.h @@ -22,17 +22,11 @@ namespace webrtc { // Handles audio RTP packets. This class is thread-safe. -class RTPReceiverAudio : public RTPReceiverStrategy, - public TelephoneEventHandler { +class RTPReceiverAudio : public RTPReceiverStrategy { public: explicit RTPReceiverAudio(RtpData* data_callback); ~RTPReceiverAudio() override; - // Is TelephoneEvent configured with |payload_type|. - bool TelephoneEventPayloadType(const int8_t payload_type) const override; - - TelephoneEventHandler* GetTelephoneEventHandler() override; - // Returns true if CNG is configured with |payload_type|. bool CNGPayloadType(const int8_t payload_type); @@ -59,6 +53,9 @@ class RTPReceiverAudio : public RTPReceiverStrategy, 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_; diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc index ac18688d2d..1cdbfdf447 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -195,10 +195,6 @@ bool RtpReceiverImpl::IncomingRtpPacket(const RTPHeader& rtp_header, return true; } -TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() { - return rtp_media_receiver_->GetTelephoneEventHandler(); -} - std::vector RtpReceiverImpl::GetSources() const { rtc::CritScope lock(&critical_section_rtp_receiver_); diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.h b/modules/rtp_rtcp/source/rtp_receiver_impl.h index 878f499e4c..55a57eaa8d 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_impl.h +++ b/modules/rtp_rtcp/source/rtp_receiver_impl.h @@ -53,8 +53,6 @@ class RtpReceiverImpl : public RtpReceiver { int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override; - TelephoneEventHandler* GetTelephoneEventHandler() override; - std::vector GetSources() const override; const std::vector& ssrc_sources_for_testing() const { diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/modules/rtp_rtcp/source/rtp_receiver_strategy.h index bc6378fded..3a8d200624 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_strategy.h +++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.h @@ -20,8 +20,6 @@ namespace webrtc { struct CodecInst; -class TelephoneEventHandler; - // This strategy deals with media-specific RTP packet processing. // This class is not thread-safe and must be protected by its caller. class RTPReceiverStrategy { @@ -42,8 +40,6 @@ class RTPReceiverStrategy { size_t payload_length, int64_t timestamp_ms) = 0; - virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; - // Computes the current dead-or-alive state. virtual RTPAliveType ProcessDeadOrAlive( uint16_t last_payload_length) const = 0; diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.cc b/modules/rtp_rtcp/source/rtp_receiver_video.cc index 5e6bf3eacf..62e93e9e13 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_video.cc @@ -108,10 +108,6 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, : -1; } -TelephoneEventHandler* RTPReceiverVideo::GetTelephoneEventHandler() { - return nullptr; -} - RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive( uint16_t last_payload_length) const { return kRtpDead; diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.h b/modules/rtp_rtcp/source/rtp_receiver_video.h index adaab9cddd..58ba3d0b0c 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_video.h +++ b/modules/rtp_rtcp/source/rtp_receiver_video.h @@ -30,8 +30,6 @@ class RTPReceiverVideo : public RTPReceiverStrategy { size_t packet_length, int64_t timestamp) override; - TelephoneEventHandler* GetTelephoneEventHandler() override; - RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override; int32_t OnNewPayloadTypeCreated(int payload_type,