| Index: webrtc/test/call_test.h
|
| diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
|
| index c9d7f32b579bf2f6e6c81eeead78845c72a85550..b9523f59388eee644207445167aa9357fb3913e7 100644
|
| --- a/webrtc/test/call_test.h
|
| +++ b/webrtc/test/call_test.h
|
| @@ -57,25 +57,9 @@ class CallTest : public ::testing::Test {
|
| static const uint32_t kReceiverLocalVideoSsrc;
|
| static const uint32_t kReceiverLocalAudioSsrc;
|
| static const int kNackRtpHistoryMs;
|
| + static const std::map<uint8_t, MediaType> payload_type_map_;
|
|
|
| protected:
|
| - // Needed for tests sending both audio and video on the same
|
| - // FakeNetworkPipe. We then need to set correct MediaType based on
|
| - // packet payload type, before passing the packet on to Call.
|
| - class PayloadDemuxer : public PacketReceiver {
|
| - public:
|
| - PayloadDemuxer() = default;
|
| -
|
| - void SetReceiver(PacketReceiver* receiver);
|
| - DeliveryStatus DeliverPacket(MediaType media_type,
|
| - const uint8_t* packet,
|
| - size_t length,
|
| - const PacketTime& packet_time) override;
|
| -
|
| - private:
|
| - PacketReceiver* receiver_ = nullptr;
|
| - };
|
| -
|
| // RunBaseTest overwrites the audio_state and the voice_engine of the send and
|
| // receive Call configs to simplify test code and avoid having old VoiceEngine
|
| // APIs in the tests.
|
| @@ -141,9 +125,6 @@ class CallTest : public ::testing::Test {
|
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
|
| test::FakeVideoRenderer fake_renderer_;
|
|
|
| - PayloadDemuxer receive_demuxer_;
|
| - PayloadDemuxer send_demuxer_;
|
| -
|
| private:
|
| // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
|
| // These methods are used to set up legacy voice engines and channels which is
|
| @@ -192,10 +173,6 @@ class BaseTest : public RtpRtcpObserver {
|
| virtual Call::Config GetReceiverCallConfig();
|
| virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
|
|
|
| - // Returns VIDEO for video-only tests, AUDIO for audio-only tests,
|
| - // and ANY for tests sending audio and video over the same
|
| - // transport.
|
| - virtual MediaType SelectMediaType();
|
| virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
|
| virtual test::PacketTransport* CreateReceiveTransport();
|
|
|
|
|