From 2beb42983ca24e1326a9a7f2c06b3ad740eea2c3 Mon Sep 17 00:00:00 2001 From: solenberg Date: Thu, 22 Sep 2016 01:46:03 -0700 Subject: [PATCH] Remove unnecessary interface TelephoneEventHandler. BUG=webrtc:2795 Review-Url: https://codereview.webrtc.org/2357583002 Cr-Commit-Position: refs/heads/master@{#14346} --- .../modules/rtp_rtcp/include/rtp_receiver.h | 18 ------------------ .../rtp_rtcp/source/rtp_receiver_audio.cc | 19 ------------------- .../rtp_rtcp/source/rtp_receiver_audio.h | 14 ++------------ .../rtp_rtcp/source/rtp_receiver_impl.cc | 4 ---- .../rtp_rtcp/source/rtp_receiver_impl.h | 2 -- .../rtp_rtcp/source/rtp_receiver_strategy.h | 2 -- .../rtp_rtcp/source/rtp_receiver_video.h | 2 -- .../rtp_rtcp/test/testAPI/test_api_audio.cc | 3 --- webrtc/voice_engine/channel.cc | 2 -- webrtc/voice_engine/channel.h | 1 - 10 files changed, 2 insertions(+), 65 deletions(-) diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h index 9db1c63da7..c04a173ff4 100644 --- a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h +++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h @@ -18,21 +18,6 @@ namespace webrtc { class RTPPayloadRegistry; -class TelephoneEventHandler { - public: - virtual ~TelephoneEventHandler() {} - - // The following three methods implement the TelephoneEventHandler interface. - // Forward DTMFs to decoder for playout. - virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0; - - // Is forwarding of outband telephone events turned on/off? - virtual bool TelephoneEventForwardToDecoder() const = 0; - - // 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. @@ -51,9 +36,6 @@ 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/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc index 38b2830b79..cd25d9ef87 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -25,9 +25,7 @@ RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy( RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback) : RTPReceiverStrategy(data_callback), - TelephoneEventHandler(), last_received_frequency_(8000), - telephone_event_forward_to_decoder_(false), telephone_event_payload_type_(-1), cng_nb_payload_type_(-1), cng_wb_payload_type_(-1), @@ -42,19 +40,6 @@ RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback) memset(current_remote_energy_, 0, sizeof(current_remote_energy_)); } -// Outband TelephoneEvent(DTMF) detection -void RTPReceiverAudio::SetTelephoneEventForwardToDecoder( - bool forward_to_decoder) { - rtc::CritScope lock(&crit_sect_); - telephone_event_forward_to_decoder_ = forward_to_decoder; -} - -// Is forwarding of outband telephone events turned on/off? -bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const { - rtc::CritScope lock(&crit_sect_); - return telephone_event_forward_to_decoder_; -} - bool RTPReceiverAudio::TelephoneEventPayloadType( int8_t payload_type) const { rtc::CritScope lock(&crit_sect_); @@ -356,10 +341,6 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific( // check if it's a DTMF event, hence something we can playout if (telephone_event_packet) { - if (!telephone_event_forward_to_decoder_) { - // don't forward event to decoder - return 0; - } std::set::iterator first = telephone_event_reported_.begin(); if (first != telephone_event_reported_.end() && *first > 15) { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h index d5d89bae2d..c1bccc2087 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h @@ -23,23 +23,13 @@ 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); virtual ~RTPReceiverAudio() {} - // The following three methods implement the TelephoneEventHandler interface. - // Forward DTMFs to decoder for playout. - void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override; - - // Is forwarding of outband telephone events turned on/off? - bool TelephoneEventForwardToDecoder() const override; - // Is TelephoneEvent configured with payload type payload_type - bool TelephoneEventPayloadType(const int8_t payload_type) const override; - - TelephoneEventHandler* GetTelephoneEventHandler() override { return this; } + bool TelephoneEventPayloadType(const int8_t payload_type) const; // Returns true if CNG is configured with payload type payload_type. If so, // the frequency and cng_payload_type_has_changed are filled in. diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc index 190449b3dd..babad1c069 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -200,10 +200,6 @@ bool RtpReceiverImpl::IncomingRtpPacket( return true; } -TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() { - return rtp_media_receiver_->GetTelephoneEventHandler(); -} - bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const { rtc::CritScope lock(&critical_section_rtp_receiver_); if (!HaveReceivedFrame()) diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h index 1ae1c9168a..7ac81d9231 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h @@ -57,8 +57,6 @@ class RtpReceiverImpl : public RtpReceiver { int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override; - TelephoneEventHandler* GetTelephoneEventHandler() override; - private: bool HaveReceivedFrame() const; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h index 663b883295..02ec8f836a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h @@ -44,8 +44,6 @@ class RTPReceiverStrategy { int64_t timestamp_ms, bool is_first_packet) = 0; - virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; - // Retrieves the last known applicable frequency. virtual int GetPayloadTypeFrequency() const = 0; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h index a8aaf5da18..1a3b41b9d9 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h @@ -33,8 +33,6 @@ class RTPReceiverVideo : public RTPReceiverStrategy { int64_t timestamp, bool is_first_packet) override; - TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; } - int GetPayloadTypeFrequency() const override; RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override; diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc index 291dded3b2..bb93f57e93 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc @@ -162,9 +162,6 @@ TEST_F(RtpRtcpAudioTest, Basic) { module1->SetSSRC(test_ssrc); module1->SetStartTimestamp(test_timestamp); - // Test detection at the end of a DTMF tone. - // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true)); - EXPECT_EQ(0, module1->SetSendingStatus(true)); // Start basic RTP test. diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc index 582bde5f26..7895e9b9e1 100644 --- a/webrtc/voice_engine/channel.cc +++ b/webrtc/voice_engine/channel.cc @@ -809,7 +809,6 @@ Channel::Channel(int32_t channelId, this, this, rtp_payload_registry_.get())), - telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()), _outputAudioLevel(), _externalTransport(false), // Avoid conflict with other channels by adding 1024 - 1026, @@ -979,7 +978,6 @@ int32_t Channel::Init() { // disabled by the user. // After StopListen (when no sockets exists), RTCP packets will no longer // be transmitted since the Transport object will then be invalid. - telephone_event_handler_->SetTelephoneEventForwardToDecoder(true); // RTCP is enabled by default. _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); // --- Register all permanent callbacks diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h index 4988e079d1..ab46d2e468 100644 --- a/webrtc/voice_engine/channel.h +++ b/webrtc/voice_engine/channel.h @@ -458,7 +458,6 @@ class Channel std::unique_ptr rtp_receive_statistics_; std::unique_ptr statistics_proxy_; std::unique_ptr rtp_receiver_; - TelephoneEventHandler* telephone_event_handler_; std::unique_ptr _rtpRtcpModule; std::unique_ptr audio_coding_; acm2::CodecManager codec_manager_;