1 files changed, 37 insertions, 3 deletions
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index 76a70aaa57..e8c7f6e0c9 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -96,9 +96,31 @@ bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
   return true;
 }
 
-void FakeAudioReceiveStream::Reconfigure(
-    const webrtc::AudioReceiveStream::Config& config) {
-  config_ = config;
+void FakeAudioReceiveStream::SetDepacketizerToDecoderFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  config_.frame_transformer = std::move(frame_transformer);
+}
+
+void FakeAudioReceiveStream::SetDecoderMap(
+    std::map<int, webrtc::SdpAudioFormat> decoder_map) {
+  config_.decoder_map = std::move(decoder_map);
+}
+
+void FakeAudioReceiveStream::SetUseTransportCcAndNackHistory(
+    bool use_transport_cc,
+    int history_ms) {
+  config_.rtp.transport_cc = use_transport_cc;
+  config_.rtp.nack.rtp_history_ms = history_ms;
+}
+
+void FakeAudioReceiveStream::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  config_.frame_decryptor = std::move(frame_decryptor);
+}
+
+void FakeAudioReceiveStream::SetRtpExtensions(
+    std::vector<webrtc::RtpExtension> extensions) {
+  config_.rtp.extensions = std::move(extensions);
 }
 
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats(
@@ -646,6 +668,18 @@ void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
 void FakeCall::OnAudioTransportOverheadChanged(
     int transport_overhead_per_packet) {}
 
+void FakeCall::OnLocalSsrcUpdated(webrtc::AudioReceiveStream& stream,
+                                  uint32_t local_ssrc) {
+  auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
+  fake_stream.SetLocalSsrc(local_ssrc);
+}
+
+void FakeCall::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream,
+                                 const std::string& sync_group) {
+  auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
+  fake_stream.SetSyncGroup(sync_group);
+}
+
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {