Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/1934513002
Cr-Commit-Position: refs/heads/master@{#12566}
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 5343800..13a3696 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -121,15 +121,13 @@
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec,
- public webrtc::VoEHardware,
- public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
+ public webrtc::VoEHardware, public webrtc::VoERTP_RTCP,
public webrtc::VoEVolumeControl {
public:
struct Channel {
Channel() {
memset(&send_codec, 0, sizeof(send_codec));
}
- bool external_transport = false;
bool playout = false;
float volume_scale = 1.0f;
bool vad = false;
@@ -146,8 +144,6 @@
int associate_send_channel = -1;
std::vector<webrtc::CodecInst> recv_codecs;
webrtc::CodecInst send_codec;
- webrtc::PacketTime last_rtp_packet_time;
- std::list<std::string> packets;
int neteq_capacity = -1;
bool neteq_fast_accelerate = false;
};
@@ -191,10 +187,6 @@
int GetNACKMaxPackets(int channel) {
return channels_[channel]->nack_max_packets;
}
- const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
- RTC_DCHECK(channels_.find(channel) != channels_.end());
- return channels_[channel]->last_rtp_packet_time;
- }
int GetSendCNPayloadType(int channel, bool wideband) {
return (wideband) ?
channels_[channel]->cn16_type :
@@ -455,40 +447,6 @@
WEBRTC_STUB(EnableBuiltInNS, (bool enable));
bool BuiltInNSIsAvailable() const override { return false; }
- // webrtc::VoENetwork
- WEBRTC_FUNC(RegisterExternalTransport, (int channel,
- webrtc::Transport& transport)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->external_transport = true;
- return 0;
- }
- WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->external_transport = false;
- return 0;
- }
- WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- size_t length)) {
- WEBRTC_CHECK_CHANNEL(channel);
- if (!channels_[channel]->external_transport) return -1;
- channels_[channel]->packets.push_back(
- std::string(static_cast<const char*>(data), length));
- return 0;
- }
- WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- size_t length,
- const webrtc::PacketTime& packet_time)) {
- WEBRTC_CHECK_CHANNEL(channel);
- if (ReceivedRTPPacket(channel, data, length) == -1) {
- return -1;
- }
- channels_[channel]->last_rtp_packet_time = packet_time;
- return 0;
- }
-
- WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
- size_t length));
-
// webrtc::VoERTP_RTCP
WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
WEBRTC_CHECK_CHANNEL(channel);
diff --git a/webrtc/media/engine/webrtcvoe.h b/webrtc/media/engine/webrtcvoe.h
index 2f7b997..c6d1caf 100644
--- a/webrtc/media/engine/webrtcvoe.h
+++ b/webrtc/media/engine/webrtcvoe.h
@@ -74,15 +74,13 @@
public:
VoEWrapper()
: engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
- base_(engine_), codec_(engine_),
- hw_(engine_), network_(engine_),
- rtp_(engine_), volume_(engine_) {
+ base_(engine_), codec_(engine_), hw_(engine_), rtp_(engine_),
+ volume_(engine_) {
}
VoEWrapper(webrtc::VoEAudioProcessing* processing,
webrtc::VoEBase* base,
webrtc::VoECodec* codec,
webrtc::VoEHardware* hw,
- webrtc::VoENetwork* network,
webrtc::VoERTP_RTCP* rtp,
webrtc::VoEVolumeControl* volume)
: engine_(NULL),
@@ -90,7 +88,6 @@
base_(base),
codec_(codec),
hw_(hw),
- network_(network),
rtp_(rtp),
volume_(volume) {
}
@@ -100,7 +97,6 @@
webrtc::VoEBase* base() const { return base_.get(); }
webrtc::VoECodec* codec() const { return codec_.get(); }
webrtc::VoEHardware* hw() const { return hw_.get(); }
- webrtc::VoENetwork* network() const { return network_.get(); }
webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
int error() { return base_->LastError(); }
@@ -111,7 +107,6 @@
scoped_voe_ptr<webrtc::VoEBase> base_;
scoped_voe_ptr<webrtc::VoECodec> codec_;
scoped_voe_ptr<webrtc::VoEHardware> hw_;
- scoped_voe_ptr<webrtc::VoENetwork> network_;
scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
};
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 64f53b7..ed67ce8 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2268,7 +2268,6 @@
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
packet->cdata(), packet->size(),
webrtc_packet_time);
-
if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
return;
}
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index a0d7417..c72775a 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -58,7 +58,6 @@
engine, // base
engine, // codec
engine, // hw
- engine, // network
engine, // rtp
engine) { // volume
}