Split out RtpSource from libjingle_peerconnection_api
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 2bb8be1..0b764a1 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -20,15 +20,15 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
#include "api/media_transport_config.h"
#include "api/rtp_parameters.h"
-#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
+#include "api/transport/rtp/rtp_source.h"
#include "call/rtp_config.h"
namespace webrtc {
class AudioSinkInterface;
-class FrameDecryptorInterface;
class AudioReceiveStream {
public:
