Delete obsolete interface class RtpData
Unused since cl https://webrtc-review.googlesource.com/c/103503
Bug: webrtc:8995
Change-Id: I62a3cab6f7c778fd0a126afb66073da511f0abc1
Reviewed-on: https://webrtc-review.googlesource.com/c/110700
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25613}diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 225c335..de2ae2a 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -104,7 +104,7 @@
State state_;
};
-class ChannelReceive : public RtpData, public MediaTransportAudioSinkInterface {
+class ChannelReceive : public MediaTransportAudioSinkInterface {
public:
// Used for receive streams.
ChannelReceive(ProcessThread* module_process_thread,
@@ -198,11 +198,9 @@
void OnData(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) override;
- // From RtpData in the RTP/RTCP module
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
- const WebRtcRTPHeader* rtpHeader) override;
-
+ const WebRtcRTPHeader* rtpHeader);
rtc::CriticalSection _callbackCritSect;
rtc::CriticalSection volume_settings_critsect_;
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 138ed0a..b0ff13d 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -223,15 +223,6 @@
bool media_has_been_sent;
};
-class RtpData {
- public:
- virtual ~RtpData() {}
-
- virtual int32_t OnReceivedPayloadData(const uint8_t* payload_data,
- size_t payload_size,
- const WebRtcRTPHeader* rtp_header) = 0;
-};
-
// Callback interface for packets recovered by FlexFEC or ULPFEC. In
// the FlexFEC case, the implementation should be able to demultiplex
// the recovered RTP packets based on SSRC.
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
index d24c1b0..061f827 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
@@ -12,9 +12,6 @@
namespace webrtc {
-MockRtpData::MockRtpData() = default;
-MockRtpData::~MockRtpData() = default;
-
MockRtpRtcp::MockRtpRtcp() = default;
MockRtpRtcp::~MockRtpRtcp() = default;
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 9f00654..3b9b943 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -27,17 +27,6 @@
namespace webrtc {
-class MockRtpData : public RtpData {
- public:
- MockRtpData();
- ~MockRtpData();
-
- MOCK_METHOD3(OnReceivedPayloadData,
- int32_t(const uint8_t* payload_data,
- size_t payload_size,
- const WebRtcRTPHeader* rtp_header));
-};
-
class MockRtpRtcp : public RtpRtcp {
public:
MockRtpRtcp();
diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 3e37cc6..a0e1528 100644
--- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -45,7 +45,7 @@
RtcpPacketTypeCounter counter_;
};
-class TestTransport : public Transport, public RtpData {
+class TestTransport : public Transport {
public:
TestTransport() {}
@@ -58,11 +58,6 @@
parser_.Parse(data, len);
return true;
}
- int OnReceivedPayloadData(const uint8_t* payload_data,
- size_t payload_size,
- const WebRtcRTPHeader* rtp_header) override {
- return 0;
- }
test::RtcpPacketParser parser_;
};
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 5160a64..5884bc0 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -49,7 +49,7 @@
int64_t rtt_ms_;
};
-class SendTransport : public Transport, public RtpData {
+class SendTransport : public Transport {
public:
SendTransport()
: receiver_(nullptr),
@@ -90,11 +90,6 @@
++rtcp_packets_sent_;
return true;
}
- int32_t OnReceivedPayloadData(const uint8_t* payload_data,
- size_t payload_size,
- const WebRtcRTPHeader* rtp_header) override {
- return 0;
- }
void SetKeepalivePayloadType(uint8_t payload_type) {
keepalive_payload_type_ = payload_type;
}
