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}
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@ -121,15 +121,13 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
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class FakeWebRtcVoiceEngine
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: public webrtc::VoEAudioProcessing,
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public webrtc::VoEBase, public webrtc::VoECodec,
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public webrtc::VoEHardware,
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public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
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public webrtc::VoEHardware, public webrtc::VoERTP_RTCP,
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public webrtc::VoEVolumeControl {
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public:
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struct Channel {
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Channel() {
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memset(&send_codec, 0, sizeof(send_codec));
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}
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bool external_transport = false;
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bool playout = false;
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float volume_scale = 1.0f;
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bool vad = false;
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@ -146,8 +144,6 @@ class FakeWebRtcVoiceEngine
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int associate_send_channel = -1;
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std::vector<webrtc::CodecInst> recv_codecs;
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webrtc::CodecInst send_codec;
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webrtc::PacketTime last_rtp_packet_time;
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std::list<std::string> packets;
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int neteq_capacity = -1;
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bool neteq_fast_accelerate = false;
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};
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@ -191,10 +187,6 @@ class FakeWebRtcVoiceEngine
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int GetNACKMaxPackets(int channel) {
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return channels_[channel]->nack_max_packets;
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}
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const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
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RTC_DCHECK(channels_.find(channel) != channels_.end());
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return channels_[channel]->last_rtp_packet_time;
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}
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int GetSendCNPayloadType(int channel, bool wideband) {
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return (wideband) ?
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channels_[channel]->cn16_type :
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@ -455,40 +447,6 @@ class FakeWebRtcVoiceEngine
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WEBRTC_STUB(EnableBuiltInNS, (bool enable));
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bool BuiltInNSIsAvailable() const override { return false; }
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// webrtc::VoENetwork
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WEBRTC_FUNC(RegisterExternalTransport, (int channel,
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webrtc::Transport& transport)) {
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WEBRTC_CHECK_CHANNEL(channel);
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channels_[channel]->external_transport = true;
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return 0;
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}
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WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
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WEBRTC_CHECK_CHANNEL(channel);
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channels_[channel]->external_transport = false;
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return 0;
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}
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WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
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size_t length)) {
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WEBRTC_CHECK_CHANNEL(channel);
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if (!channels_[channel]->external_transport) return -1;
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channels_[channel]->packets.push_back(
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std::string(static_cast<const char*>(data), length));
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return 0;
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}
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WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
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size_t length,
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const webrtc::PacketTime& packet_time)) {
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WEBRTC_CHECK_CHANNEL(channel);
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if (ReceivedRTPPacket(channel, data, length) == -1) {
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return -1;
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}
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channels_[channel]->last_rtp_packet_time = packet_time;
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return 0;
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}
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WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
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size_t length));
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// webrtc::VoERTP_RTCP
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WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
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WEBRTC_CHECK_CHANNEL(channel);
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@ -74,15 +74,13 @@ class VoEWrapper {
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public:
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VoEWrapper()
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: engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
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base_(engine_), codec_(engine_),
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hw_(engine_), network_(engine_),
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rtp_(engine_), volume_(engine_) {
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base_(engine_), codec_(engine_), hw_(engine_), rtp_(engine_),
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volume_(engine_) {
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}
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VoEWrapper(webrtc::VoEAudioProcessing* processing,
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webrtc::VoEBase* base,
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webrtc::VoECodec* codec,
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webrtc::VoEHardware* hw,
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webrtc::VoENetwork* network,
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webrtc::VoERTP_RTCP* rtp,
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webrtc::VoEVolumeControl* volume)
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: engine_(NULL),
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@ -90,7 +88,6 @@ class VoEWrapper {
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base_(base),
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codec_(codec),
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hw_(hw),
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network_(network),
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rtp_(rtp),
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volume_(volume) {
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}
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@ -100,7 +97,6 @@ class VoEWrapper {
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webrtc::VoEBase* base() const { return base_.get(); }
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webrtc::VoECodec* codec() const { return codec_.get(); }
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webrtc::VoEHardware* hw() const { return hw_.get(); }
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webrtc::VoENetwork* network() const { return network_.get(); }
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webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
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webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
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int error() { return base_->LastError(); }
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@ -111,7 +107,6 @@ class VoEWrapper {
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scoped_voe_ptr<webrtc::VoEBase> base_;
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scoped_voe_ptr<webrtc::VoECodec> codec_;
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scoped_voe_ptr<webrtc::VoEHardware> hw_;
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scoped_voe_ptr<webrtc::VoENetwork> network_;
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scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
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scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
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};
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@ -2268,7 +2268,6 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
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call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
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packet->cdata(), packet->size(),
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webrtc_packet_time);
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if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
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return;
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}
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@ -58,7 +58,6 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
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engine, // base
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engine, // codec
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engine, // hw
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engine, // network
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engine, // rtp
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engine) { // volume
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}
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