| Index: webrtc/media/engine/fakewebrtcvoiceengine.h
|
| diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| index 4aa6ea3d4380de25d3b127f750ddf3e1894a6b2d..8c9d14d168766b6906e95e8e792304b171211513 100644
|
| --- a/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| +++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| @@ -121,15 +121,13 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
|
| 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 @@ class FakeWebRtcVoiceEngine
|
| 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 @@ class FakeWebRtcVoiceEngine
|
| 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 @@ class FakeWebRtcVoiceEngine
|
| WEBRTC_STUB(EnableBuiltInNS, (bool enable));
|
| virtual bool BuiltInNSIsAvailable() const { 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);
|
|
|