Move DTMF sender to RtpSender (as opposed to WebRtcSession).
Previously in the spec, there was a createDtmfSender method on
PeerConnection, but that's been replaced by a "dtmf" attribute
on RtpSender, which allows getting a DTMF sender without having
an audio track.
This also simplifies the code slightly, since tracks are now not
necessary for identification.
BUG=webrtc:4180
Review-Url: https://codereview.webrtc.org/2666853002
Cr-Original-Commit-Position: refs/heads/master@{#16409}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 20cb0c1c85d4571919106019a562bbcbd35041fd
diff --git a/api/rtpsenderinterface.h b/api/rtpsenderinterface.h
index 7129376..68547b0 100644
--- a/api/rtpsenderinterface.h
+++ b/api/rtpsenderinterface.h
@@ -17,6 +17,7 @@
#include <string>
#include <vector>
+#include "webrtc/api/dtmfsenderinterface.h"
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
@@ -51,6 +52,9 @@
virtual RtpParameters GetParameters() const = 0;
virtual bool SetParameters(const RtpParameters& parameters) = 0;
+ // Returns null for a video sender.
+ virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
+
protected:
virtual ~RtpSenderInterface() {}
};
@@ -66,6 +70,7 @@
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
+ PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
END_PROXY_MAP()
} // namespace webrtc
diff --git a/api/test/mock_rtpsender.h b/api/test/mock_rtpsender.h
index 7abc4e5..7458f45 100644
--- a/api/test/mock_rtpsender.h
+++ b/api/test/mock_rtpsender.h
@@ -29,6 +29,7 @@
MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
MOCK_CONST_METHOD0(GetParameters, RtpParameters());
MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
+ MOCK_CONST_METHOD0(GetDtmfSender, rtc::scoped_refptr<DtmfSenderInterface>());
};
} // namespace webrtc
diff --git a/pc/dtmfsender.cc b/pc/dtmfsender.cc
index 2ef921b..bbb100e 100644
--- a/pc/dtmfsender.cc
+++ b/pc/dtmfsender.cc
@@ -63,8 +63,8 @@
AudioTrackInterface* track,
rtc::Thread* signaling_thread,
DtmfProviderInterface* provider) {
- if (!track || !signaling_thread) {
- return NULL;
+ if (!signaling_thread) {
+ return nullptr;
}
rtc::scoped_refptr<DtmfSender> dtmf_sender(
new rtc::RefCountedObject<DtmfSender>(track, signaling_thread,
@@ -81,7 +81,6 @@
provider_(provider),
duration_(kDtmfDefaultDurationMs),
inter_tone_gap_(kDtmfDefaultGapMs) {
- RTC_DCHECK(track_ != NULL);
RTC_DCHECK(signaling_thread_ != NULL);
// TODO(deadbeef): Once we can use shared_ptr and weak_ptr,
// do that instead of relying on a "destroyed" signal.
@@ -109,7 +108,7 @@
if (!provider_) {
return false;
}
- return provider_->CanInsertDtmf(track_->id());
+ return provider_->CanInsertDtmf();
}
bool DtmfSender::InsertDtmf(const std::string& tones, int duration,
@@ -206,7 +205,7 @@
}
// The provider starts playout of the given tone on the
// associated RTP media stream, using the appropriate codec.
- if (!provider_->InsertDtmf(track_->id(), code, duration_)) {
+ if (!provider_->InsertDtmf(code, duration_)) {
LOG(LS_ERROR) << "The DtmfProvider can no longer send DTMF.";
return;
}
diff --git a/pc/dtmfsender.h b/pc/dtmfsender.h
index a019e78..5eab055 100644
--- a/pc/dtmfsender.h
+++ b/pc/dtmfsender.h
@@ -34,14 +34,13 @@
// to send DTMF.
class DtmfProviderInterface {
public:
- // Returns true if the audio track with given id (|track_id|) is capable
- // of sending DTMF. Otherwise returns false.
- virtual bool CanInsertDtmf(const std::string& track_id) = 0;
- // Sends DTMF |code| via the audio track with given id (|track_id|).
+ // Returns true if the audio sender is capable of sending DTMF. Otherwise
+ // returns false.
+ virtual bool CanInsertDtmf() = 0;
+ // Sends DTMF |code|.
// The |duration| indicates the length of the DTMF tone in ms.
// Returns true on success and false on failure.
- virtual bool InsertDtmf(const std::string& track_id,
- int code, int duration) = 0;
+ virtual bool InsertDtmf(int code, int duration) = 0;
// Returns a |sigslot::signal0<>| signal. The signal should fire before
// the provider is destroyed.
virtual sigslot::signal0<>* GetOnDestroyedSignal() = 0;
@@ -55,6 +54,8 @@
public sigslot::has_slots<>,
public rtc::MessageHandler {
public:
+ // |track| is only there for backwards compatibility, since there's a track
+ // accessor method.
static rtc::scoped_refptr<DtmfSender> Create(
AudioTrackInterface* track,
rtc::Thread* signaling_thread,
diff --git a/pc/dtmfsender_unittest.cc b/pc/dtmfsender_unittest.cc
index a58c1ec..109760c 100644
--- a/pc/dtmfsender_unittest.cc
+++ b/pc/dtmfsender_unittest.cc
@@ -78,13 +78,9 @@
}
// Implements DtmfProviderInterface.
- bool CanInsertDtmf(const std::string& track_label) override {
- return (can_insert_dtmf_tracks_.count(track_label) != 0);
- }
+ bool CanInsertDtmf() override { return can_insert_; }
- bool InsertDtmf(const std::string& track_label,
- int code,
- int duration) override {
+ bool InsertDtmf(int code, int duration) override {
int gap = 0;
// TODO(ronghuawu): Make the timer (basically the rtc::TimeNanos)
// mockable and use a fake timer in the unit tests.
@@ -110,15 +106,10 @@
}
// helper functions
- void AddCanInsertDtmfTrack(const std::string& label) {
- can_insert_dtmf_tracks_.insert(label);
- }
- void RemoveCanInsertDtmfTrack(const std::string& label) {
- can_insert_dtmf_tracks_.erase(label);
- }
+ void SetCanInsertDtmf(bool can_insert) { can_insert_ = can_insert; }
private:
- std::set<std::string> can_insert_dtmf_tracks_;
+ bool can_insert_ = false;
std::vector<DtmfInfo> dtmf_info_queue_;
int64_t last_insert_dtmf_call_;
sigslot::signal0<> SignalDestroyed;
@@ -130,7 +121,7 @@
: track_(AudioTrack::Create(kTestAudioLabel, NULL)),
observer_(new rtc::RefCountedObject<FakeDtmfObserver>()),
provider_(new FakeDtmfProvider()) {
- provider_->AddCanInsertDtmfTrack(kTestAudioLabel);
+ provider_->SetCanInsertDtmf(true);
dtmf_ = DtmfSender::Create(track_, rtc::Thread::Current(),
provider_.get());
dtmf_->RegisterObserver(observer_.get());
@@ -227,7 +218,7 @@
TEST_F(DtmfSenderTest, CanInsertDtmf) {
EXPECT_TRUE(dtmf_->CanInsertDtmf());
- provider_->RemoveCanInsertDtmfTrack(kTestAudioLabel);
+ provider_->SetCanInsertDtmf(false);
EXPECT_FALSE(dtmf_->CanInsertDtmf());
}
@@ -333,7 +324,7 @@
std::string tones = "3,4";
int duration = 100;
int inter_tone_gap = 50;
- provider_->RemoveCanInsertDtmfTrack(kTestAudioLabel);
+ provider_->SetCanInsertDtmf(false);
EXPECT_FALSE(dtmf_->InsertDtmf(tones, duration, inter_tone_gap));
}
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 14e1809..a38c986 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -949,20 +949,15 @@
}
if (!track) {
LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
- return NULL;
+ return nullptr;
}
- if (!local_streams_->FindAudioTrack(track->id())) {
- LOG(LS_ERROR) << "CreateDtmfSender is called with a non local audio track.";
- return NULL;
+ auto it = FindSenderForTrack(track);
+ if (it == senders_.end()) {
+ LOG(LS_ERROR) << "CreateDtmfSender called with a non-added track.";
+ return nullptr;
}
- rtc::scoped_refptr<DtmfSenderInterface> sender(
- DtmfSender::Create(track, signaling_thread(), session_.get()));
- if (!sender.get()) {
- LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
- return NULL;
- }
- return DtmfSenderProxy::Create(signaling_thread(), sender.get());
+ return (*it)->GetDtmfSender();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
diff --git a/pc/rtpsender.cc b/pc/rtpsender.cc
index 3e8c7e1..c8ac830 100644
--- a/pc/rtpsender.cc
+++ b/pc/rtpsender.cc
@@ -57,6 +57,7 @@
sink_adapter_(new LocalAudioSinkAdapter()) {
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
+ CreateDtmfSender();
}
AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
@@ -71,6 +72,7 @@
sink_adapter_(new LocalAudioSinkAdapter()) {
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
+ CreateDtmfSender();
}
AudioRtpSender::AudioRtpSender(cricket::VoiceChannel* channel,
@@ -79,12 +81,50 @@
stream_id_(rtc::CreateRandomUuid()),
channel_(channel),
stats_(stats),
- sink_adapter_(new LocalAudioSinkAdapter()) {}
+ sink_adapter_(new LocalAudioSinkAdapter()) {
+ CreateDtmfSender();
+}
AudioRtpSender::~AudioRtpSender() {
+ // For DtmfSender.
+ SignalDestroyed();
Stop();
}
+bool AudioRtpSender::CanInsertDtmf() {
+ if (!channel_) {
+ LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
+ return false;
+ }
+ // Check that this RTP sender is active (description has been applied that
+ // matches an SSRC to its ID).
+ if (!ssrc_) {
+ LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
+ return false;
+ }
+ return channel_->CanInsertDtmf();
+}
+
+bool AudioRtpSender::InsertDtmf(int code, int duration) {
+ if (!channel_) {
+ LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
+ return false;
+ }
+ if (!ssrc_) {
+ LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
+ return false;
+ }
+ if (!channel_->InsertDtmf(ssrc_, code, duration)) {
+ LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
+ return false;
+ }
+ return true;
+}
+
+sigslot::signal0<>* AudioRtpSender::GetOnDestroyedSignal() {
+ return &SignalDestroyed;
+}
+
void AudioRtpSender::OnChanged() {
TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
@@ -158,6 +198,10 @@
return channel_->SetRtpSendParameters(ssrc_, parameters);
}
+rtc::scoped_refptr<DtmfSenderInterface> AudioRtpSender::GetDtmfSender() const {
+ return dtmf_sender_proxy_;
+}
+
void AudioRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
@@ -237,6 +281,20 @@
}
}
+void AudioRtpSender::CreateDtmfSender() {
+ // Should be on signaling thread.
+ // TODO(deadbeef): Add thread checking to RtpSender/RtpReceiver
+ // implementations.
+ rtc::scoped_refptr<DtmfSenderInterface> sender(
+ DtmfSender::Create(track_, rtc::Thread::Current(), this));
+ if (!sender.get()) {
+ LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
+ RTC_NOTREACHED();
+ }
+ dtmf_sender_proxy_ =
+ DtmfSenderProxy::Create(rtc::Thread::Current(), sender.get());
+}
+
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
const std::string& stream_id,
cricket::VideoChannel* channel)
@@ -336,6 +394,11 @@
return channel_->SetRtpSendParameters(ssrc_, parameters);
}
+rtc::scoped_refptr<DtmfSenderInterface> VideoRtpSender::GetDtmfSender() const {
+ LOG(LS_ERROR) << "Tried to get DTMF sender from video sender.";
+ return nullptr;
+}
+
void VideoRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
diff --git a/pc/rtpsender.h b/pc/rtpsender.h
index ed244b4..0322399 100644
--- a/pc/rtpsender.h
+++ b/pc/rtpsender.h
@@ -24,6 +24,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/media/base/audiosource.h"
#include "webrtc/pc/channel.h"
+#include "webrtc/pc/dtmfsender.h"
#include "webrtc/pc/statscollector.h"
namespace webrtc {
@@ -68,7 +69,8 @@
rtc::CriticalSection lock_;
};
-class AudioRtpSender : public ObserverInterface,
+class AudioRtpSender : public DtmfProviderInterface,
+ public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
@@ -91,10 +93,15 @@
virtual ~AudioRtpSender();
- // ObserverInterface implementation
+ // DtmfSenderProvider implementation.
+ bool CanInsertDtmf() override;
+ bool InsertDtmf(int code, int duration) override;
+ sigslot::signal0<>* GetOnDestroyedSignal() override;
+
+ // ObserverInterface implementation.
void OnChanged() override;
- // RtpSenderInterface implementation
+ // RtpSenderInterface implementation.
bool SetTrack(MediaStreamTrackInterface* track) override;
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_;
@@ -116,6 +123,8 @@
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
+ rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
+
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
@@ -140,11 +149,16 @@
// Helper function to call SetAudioSend with "stop sending" parameters.
void ClearAudioSend();
+ void CreateDtmfSender();
+
+ sigslot::signal0<> SignalDestroyed;
+
std::string id_;
std::string stream_id_;
cricket::VoiceChannel* channel_ = nullptr;
StatsCollector* stats_;
rtc::scoped_refptr<AudioTrackInterface> track_;
+ rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
uint32_t ssrc_ = 0;
bool cached_track_enabled_ = false;
bool stopped_ = false;
@@ -197,6 +211,8 @@
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
+ rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
+
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index 86c0612..508a417 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -13,6 +13,7 @@
#include <utility>
#include "webrtc/base/gunit.h"
+#include "webrtc/base/sigslot.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/mediachannel.h"
@@ -38,6 +39,8 @@
using ::testing::InvokeWithoutArgs;
using ::testing::Return;
+namespace {
+
static const char kStreamLabel1[] = "local_stream_1";
static const char kVideoTrackId[] = "video_1";
static const char kAudioTrackId[] = "audio_1";
@@ -45,10 +48,14 @@
static const uint32_t kVideoSsrc2 = 100;
static const uint32_t kAudioSsrc = 99;
static const uint32_t kAudioSsrc2 = 101;
+static const int kDefaultTimeout = 10000; // 10 seconds.
+
+} // namespace
namespace webrtc {
-class RtpSenderReceiverTest : public testing::Test {
+class RtpSenderReceiverTest : public testing::Test,
+ public sigslot::has_slots<> {
public:
RtpSenderReceiverTest()
: // Create fake media engine/etc. so we can create channels to use to
@@ -75,6 +82,8 @@
&fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(),
cricket::CN_VIDEO, nullptr, rtcp_mux_required, srtp_required,
cricket::VideoOptions());
+ voice_channel_->Enable(true);
+ video_channel_->Enable(true);
voice_media_channel_ = media_engine_->GetVoiceChannel(0);
video_media_channel_ = media_engine_->GetVideoChannel(0);
RTC_CHECK(voice_channel_);
@@ -104,7 +113,14 @@
cricket::StreamParams::CreateLegacy(kVideoSsrc2));
}
- void TearDown() override { channel_manager_.Terminate(); }
+ // Needed to use DTMF sender.
+ void AddDtmfCodec() {
+ cricket::AudioSendParameters params;
+ const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000,
+ 0, 1);
+ params.codecs.push_back(kTelephoneEventCodec);
+ voice_media_channel_->SetSendParameters(params);
+ }
void AddVideoTrack() { AddVideoTrack(false); }
@@ -124,9 +140,13 @@
new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
voice_channel_, nullptr);
audio_rtp_sender_->SetSsrc(kAudioSsrc);
+ audio_rtp_sender_->GetOnDestroyedSignal()->connect(
+ this, &RtpSenderReceiverTest::OnAudioSenderDestroyed);
VerifyVoiceChannelInput();
}
+ void OnAudioSenderDestroyed() { audio_sender_destroyed_signal_fired_ = true; }
+
void CreateVideoRtpSender() { CreateVideoRtpSender(false); }
void CreateVideoRtpSender(bool is_screencast) {
@@ -247,6 +267,7 @@
rtc::scoped_refptr<MediaStreamInterface> stream_;
rtc::scoped_refptr<VideoTrackInterface> video_track_;
rtc::scoped_refptr<AudioTrackInterface> audio_track_;
+ bool audio_sender_destroyed_signal_fired_ = false;
};
// Test that |voice_channel_| is updated when an audio track is associated
@@ -721,4 +742,66 @@
DestroyVideoRtpSender();
}
+TEST_F(RtpSenderReceiverTest, AudioSenderHasDtmfSender) {
+ CreateAudioRtpSender();
+ EXPECT_NE(nullptr, audio_rtp_sender_->GetDtmfSender());
+}
+
+TEST_F(RtpSenderReceiverTest, VideoSenderDoesNotHaveDtmfSender) {
+ CreateVideoRtpSender();
+ EXPECT_EQ(nullptr, video_rtp_sender_->GetDtmfSender());
+}
+
+// Test that the DTMF sender is really using |voice_channel_|, and thus returns
+// true/false from CanSendDtmf based on what |voice_channel_| returns.
+TEST_F(RtpSenderReceiverTest, CanInsertDtmf) {
+ AddDtmfCodec();
+ CreateAudioRtpSender();
+ auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
+ ASSERT_NE(nullptr, dtmf_sender);
+ EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
+}
+
+TEST_F(RtpSenderReceiverTest, CanNotInsertDtmf) {
+ CreateAudioRtpSender();
+ auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
+ ASSERT_NE(nullptr, dtmf_sender);
+ // DTMF codec has not been added, as it was in the above test.
+ EXPECT_FALSE(dtmf_sender->CanInsertDtmf());
+}
+
+TEST_F(RtpSenderReceiverTest, InsertDtmf) {
+ AddDtmfCodec();
+ CreateAudioRtpSender();
+ auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
+ ASSERT_NE(nullptr, dtmf_sender);
+
+ EXPECT_EQ(0U, voice_media_channel_->dtmf_info_queue().size());
+
+ // Insert DTMF
+ const int expected_duration = 90;
+ dtmf_sender->InsertDtmf("012", expected_duration, 100);
+
+ // Verify
+ ASSERT_EQ_WAIT(3U, voice_media_channel_->dtmf_info_queue().size(),
+ kDefaultTimeout);
+ const uint32_t send_ssrc =
+ voice_media_channel_->send_streams()[0].first_ssrc();
+ EXPECT_TRUE(CompareDtmfInfo(voice_media_channel_->dtmf_info_queue()[0],
+ send_ssrc, 0, expected_duration));
+ EXPECT_TRUE(CompareDtmfInfo(voice_media_channel_->dtmf_info_queue()[1],
+ send_ssrc, 1, expected_duration));
+ EXPECT_TRUE(CompareDtmfInfo(voice_media_channel_->dtmf_info_queue()[2],
+ send_ssrc, 2, expected_duration));
+}
+
+// Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
+// destroyed, which is needed for the DTMF sender.
+TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
+ CreateAudioRtpSender();
+ EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
+ audio_rtp_sender_ = nullptr;
+ EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
+}
+
} // namespace webrtc
diff --git a/pc/webrtcsession.cc b/pc/webrtcsession.cc
index cfcc487..a02aa57 100644
--- a/pc/webrtcsession.cc
+++ b/pc/webrtcsession.cc
@@ -231,29 +231,6 @@
return true;
}
-static bool GetAudioSsrcByTrackId(const SessionDescription* session_description,
- const std::string& track_id,
- uint32_t* ssrc) {
- const cricket::ContentInfo* audio_info =
- cricket::GetFirstAudioContent(session_description);
- if (!audio_info) {
- LOG(LS_ERROR) << "Audio not used in this call";
- return false;
- }
-
- const cricket::MediaContentDescription* audio_content =
- static_cast<const cricket::MediaContentDescription*>(
- audio_info->description);
- const cricket::StreamParams* stream =
- cricket::GetStreamByIds(audio_content->streams(), "", track_id);
- if (!stream) {
- return false;
- }
-
- *ssrc = stream->first_ssrc();
- return true;
-}
-
static bool GetTrackIdBySsrc(const SessionDescription* session_description,
uint32_t ssrc,
std::string* track_id) {
@@ -523,7 +500,6 @@
quic_data_transport_.reset();
}
#endif
- SignalDestroyed();
LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
}
@@ -1248,49 +1224,6 @@
return desc.str();
}
-bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
- RTC_DCHECK(signaling_thread()->IsCurrent());
- if (!voice_channel_) {
- LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
- return false;
- }
- uint32_t send_ssrc = 0;
- // The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
- // exists.
- if (!local_description() ||
- !GetAudioSsrcByTrackId(local_description()->description(), track_id,
- &send_ssrc)) {
- LOG(LS_ERROR) << "CanInsertDtmf: Track does not exist: " << track_id;
- return false;
- }
- return voice_channel_->CanInsertDtmf();
-}
-
-bool WebRtcSession::InsertDtmf(const std::string& track_id,
- int code, int duration) {
- RTC_DCHECK(signaling_thread()->IsCurrent());
- if (!voice_channel_) {
- LOG(LS_ERROR) << "InsertDtmf: No audio channel exists.";
- return false;
- }
- uint32_t send_ssrc = 0;
- if (!(local_description() &&
- GetAudioSsrcByTrackId(local_description()->description(),
- track_id, &send_ssrc))) {
- LOG(LS_ERROR) << "InsertDtmf: Track does not exist: " << track_id;
- return false;
- }
- if (!voice_channel_->InsertDtmf(send_ssrc, code, duration)) {
- LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
- return false;
- }
- return true;
-}
-
-sigslot::signal0<>* WebRtcSession::GetOnDestroyedSignal() {
- return &SignalDestroyed;
-}
-
bool WebRtcSession::SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) {
diff --git a/pc/webrtcsession.h b/pc/webrtcsession.h
index c250833..f8e0830 100644
--- a/pc/webrtcsession.h
+++ b/pc/webrtcsession.h
@@ -27,7 +27,6 @@
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/transportcontroller.h"
#include "webrtc/pc/datachannel.h"
-#include "webrtc/pc/dtmfsender.h"
#include "webrtc/pc/mediacontroller.h"
#include "webrtc/pc/mediasession.h"
@@ -140,8 +139,6 @@
// packets are represented by TransportChannels. The application-level protocol
// is represented by SessionDecription objects.
class WebRtcSession :
-
- public DtmfProviderInterface,
public DataChannelProviderInterface,
public sigslot::has_slots<> {
public:
@@ -285,12 +282,6 @@
virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
- // Implements DtmfProviderInterface.
- bool CanInsertDtmf(const std::string& track_id) override;
- bool InsertDtmf(const std::string& track_id,
- int code, int duration) override;
- sigslot::signal0<>* GetOnDestroyedSignal() override;
-
// Implements DataChannelProviderInterface.
bool SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
@@ -361,8 +352,6 @@
sigslot::signal0<> SignalVideoChannelDestroyed;
sigslot::signal0<> SignalDataChannelCreated;
sigslot::signal0<> SignalDataChannelDestroyed;
- // Called when the whole session is destroyed.
- sigslot::signal0<> SignalDestroyed;
// Called when a valid data channel OPEN message is received.
// std::string represents the data channel label.
diff --git a/pc/webrtcsession_unittest.cc b/pc/webrtcsession_unittest.cc
index efd5d0c..716ba25 100644
--- a/pc/webrtcsession_unittest.cc
+++ b/pc/webrtcsession_unittest.cc
@@ -434,8 +434,6 @@
fake_sctp_transport_factory_)));
session_->SignalDataChannelOpenMessage.connect(
this, &WebRtcSessionTest::OnDataChannelOpenMessage);
- session_->GetOnDestroyedSignal()->connect(
- this, &WebRtcSessionTest::OnSessionDestroyed);
configuration_.rtcp_mux_policy = rtcp_mux_policy;
EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
@@ -454,8 +452,6 @@
last_data_channel_config_ = config;
}
- void OnSessionDestroyed() { session_destroyed_ = true; }
-
void Init() {
Init(nullptr, PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
}
@@ -502,17 +498,6 @@
PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
}
- void InitWithDtmfCodec() {
- // Add kTelephoneEventCodec for dtmf test.
- const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000,
- 0, 1);
- std::vector<cricket::AudioCodec> codecs;
- codecs.push_back(kTelephoneEventCodec);
- media_engine_->SetAudioCodecs(codecs);
- desc_factory_->set_audio_codecs(codecs, codecs);
- Init();
- }
-
void InitWithGcm() {
rtc::CryptoOptions crypto_options;
crypto_options.enable_gcm_crypto_suites = true;
@@ -1197,18 +1182,6 @@
EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
}
}
- // Tests that we can only send DTMF when the dtmf codec is supported.
- void TestCanInsertDtmf(bool can) {
- if (can) {
- InitWithDtmfCodec();
- } else {
- Init();
- }
- SendAudioVideoStream1();
- CreateAndSetRemoteOfferAndLocalAnswer();
- EXPECT_FALSE(session_->CanInsertDtmf(""));
- EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
- }
bool ContainsVideoCodecWithName(const SessionDescriptionInterface* desc,
const std::string& codec_name) {
@@ -1567,7 +1540,6 @@
// Last values received from data channel creation signal.
std::string last_data_channel_label_;
InternalDataChannelInit last_data_channel_config_;
- bool session_destroyed_ = false;
bool with_gcm_ = false;
};
@@ -3507,39 +3479,6 @@
CreateAndSetRemoteOfferAndLocalAnswer();
}
-TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
- TestCanInsertDtmf(false);
-}
-
-TEST_F(WebRtcSessionTest, CanInsertDtmf) {
- TestCanInsertDtmf(true);
-}
-
-TEST_F(WebRtcSessionTest, InsertDtmf) {
- // Setup
- Init();
- SendAudioVideoStream1();
- CreateAndSetRemoteOfferAndLocalAnswer();
- FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
- EXPECT_EQ(0U, channel->dtmf_info_queue().size());
-
- // Insert DTMF
- const int expected_duration = 90;
- session_->InsertDtmf(kAudioTrack1, 0, expected_duration);
- session_->InsertDtmf(kAudioTrack1, 1, expected_duration);
- session_->InsertDtmf(kAudioTrack1, 2, expected_duration);
-
- // Verify
- ASSERT_EQ(3U, channel->dtmf_info_queue().size());
- const uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
- EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[0], send_ssrc, 0,
- expected_duration));
- EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[1], send_ssrc, 1,
- expected_duration));
- EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[2], send_ssrc, 2,
- expected_duration));
-}
-
// This test verifies the |initial_offerer| flag when session initiates the
// call.
TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
@@ -4400,14 +4339,6 @@
TestPacketOptions();
}
-// Make sure the signal from "GetOnDestroyedSignal()" fires when the session
-// is destroyed.
-TEST_F(WebRtcSessionTest, TestOnDestroyedSignal) {
- Init();
- session_.reset();
- EXPECT_TRUE(session_destroyed_);
-}
-
// TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
// currently fails because upon disconnection and reconnection OnIceComplete is
// called more than once without returning to IceGatheringGathering.
