Remove unnecessary interface TelephoneEventHandler.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2357583002
Cr-Original-Commit-Position: refs/heads/master@{#14346}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 2beb42983ca24e1326a9a7f2c06b3ad740eea2c3
diff --git a/modules/rtp_rtcp/include/rtp_receiver.h b/modules/rtp_rtcp/include/rtp_receiver.h
index 9db1c63..c04a173 100644
--- a/modules/rtp_rtcp/include/rtp_receiver.h
+++ b/modules/rtp_rtcp/include/rtp_receiver.h
@@ -18,21 +18,6 @@
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 @@
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 38b2830..cd25d9e 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -25,9 +25,7 @@
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 @@
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 @@
// 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<uint8_t>::iterator first =
telephone_event_reported_.begin();
if (first != telephone_event_reported_.end() && *first > 15) {
diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.h b/modules/rtp_rtcp/source/rtp_receiver_audio.h
index d5d89ba..c1bccc2 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/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/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 190449b..babad1c 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -200,10 +200,6 @@
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/modules/rtp_rtcp/source/rtp_receiver_impl.h b/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 1ae1c91..7ac81d9 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -57,8 +57,6 @@
int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
- TelephoneEventHandler* GetTelephoneEventHandler() override;
-
private:
bool HaveReceivedFrame() const;
diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index 663b883..02ec8f8 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -44,8 +44,6 @@
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/modules/rtp_rtcp/source/rtp_receiver_video.h b/modules/rtp_rtcp/source/rtp_receiver_video.h
index a8aaf5d..1a3b41b 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -33,8 +33,6 @@
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/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 291dded..bb93f57 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -162,9 +162,6 @@
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/voice_engine/channel.cc b/voice_engine/channel.cc
index 582bde5..7895e9b 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -809,7 +809,6 @@
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 @@
// 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/voice_engine/channel.h b/voice_engine/channel.h
index 4988e07..ab46d2e 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -458,7 +458,6 @@
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<StatisticsProxy> statistics_proxy_;
std::unique_ptr<RtpReceiver> rtp_receiver_;
- TelephoneEventHandler* telephone_event_handler_;
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
std::unique_ptr<AudioCodingModule> audio_coding_;
acm2::CodecManager codec_manager_;
