1 files changed, 7 insertions, 48 deletions

diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index cc9020b5ed..61edc886cb 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -20,17 +20,15 @@
 
 #include "api/call/transport.h"
 #include "api/crypto/crypto_options.h"
-#include "api/crypto/frame_decryptor_interface.h"
-#include "api/frame_transformer_interface.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
-#include "api/transport/rtp/rtp_source.h"
 #include "api/video/recordable_encoded_frame.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_timing.h"
 #include "api/video_codecs/sdp_video_format.h"
+#include "call/receive_stream.h"
 #include "call/rtp_config.h"
 #include "common_video/frame_counts.h"
 #include "modules/rtp_rtcp/include/rtcp_statistics.h"
@@ -41,7 +39,7 @@ namespace webrtc {
 class RtpPacketSinkInterface;
 class VideoDecoderFactory;
 
-class VideoReceiveStream {
+class VideoReceiveStream : public MediaReceiveStream {
  public:
   // Class for handling moving in/out recording state.
   struct RecordingState {
@@ -53,11 +51,6 @@ class VideoReceiveStream {
     // Callback stored from the VideoReceiveStream. The VideoReceiveStream
     // client should not interpret the attribute.
     std::function<void(const RecordableEncodedFrame&)> callback;
-    // Memento of internal state in VideoReceiveStream, recording wether
-    // we're currently causing generation of a keyframe from the sender. Needed
-    // to avoid sending double keyframe requests. The VideoReceiveStream client
-    // should not interpret the attribute.
-    bool keyframe_needed = false;
     // Memento of when a keyframe request was last sent. The VideoReceiveStream
     // client should not interpret the attribute.
     absl::optional<int64_t> last_keyframe_request_ms;
@@ -157,7 +150,8 @@ class VideoReceiveStream {
    public:
     Config() = delete;
     Config(Config&&);
-    explicit Config(Transport* rtcp_send_transport);
+    Config(Transport* rtcp_send_transport,
+           VideoDecoderFactory* decoder_factory = nullptr);
     Config& operator=(Config&&);
     Config& operator=(const Config&) = delete;
     ~Config();
@@ -174,17 +168,14 @@ class VideoReceiveStream {
     VideoDecoderFactory* decoder_factory = nullptr;
 
     // Receive-stream specific RTP settings.
-    struct Rtp {
+    struct Rtp : public RtpConfig {
       Rtp();
       Rtp(const Rtp&);
       ~Rtp();
       std::string ToString() const;
 
-      // Synchronization source (stream identifier) to be received.
-      uint32_t remote_ssrc = 0;
-
-      // Sender SSRC used for sending RTCP (such as receiver reports).
-      uint32_t local_ssrc = 0;
+      // See NackConfig for description.
+      NackConfig nack;
 
       // See RtcpMode for description.
       RtcpMode rtcp_mode = RtcpMode::kCompound;
@@ -196,15 +187,9 @@ class VideoReceiveStream {
         bool receiver_reference_time_report = false;
       } rtcp_xr;
 
-      // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
-      bool transport_cc = false;
-
       // See LntfConfig for description.
       LntfConfig lntf;
 
-      // See NackConfig for description.
-      NackConfig nack;
-
       // Payload types for ULPFEC and RED, respectively.
       int ulpfec_payload_type = -1;
       int red_payload_type = -1;
@@ -228,9 +213,6 @@ class VideoReceiveStream {
       // meta data is expected to be present in generic frame descriptor
       // RTP header extension).
       std::set<int> raw_payload_types;
-
-      // RTP header extensions used for the received stream.
-      std::vector<RtpExtension> extensions;
     } rtp;
 
     // Transport for outgoing packets (RTCP).
@@ -256,10 +238,6 @@ class VideoReceiveStream {
     // used for streaming instead of a real-time call.
     int target_delay_ms = 0;
 
-    // TODO(nisse): Used with VideoDecoderFactory::LegacyCreateVideoDecoder.
-    // Delete when that method is retired.
-    std::string stream_id;
-
     // An optional custom frame decryptor that allows the entire frame to be
     // decrypted in whatever way the caller choses. This is not required by
     // default.
@@ -271,18 +249,9 @@ class VideoReceiveStream {
     rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
   };
 
-  // Starts stream activity.
-  // When a stream is active, it can receive, process and deliver packets.
-  virtual void Start() = 0;
-  // Stops stream activity.
-  // When a stream is stopped, it can't receive, process or deliver packets.
-  virtual void Stop() = 0;
-
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
 
-  virtual std::vector<RtpSource> GetSources() const = 0;
-
   // Sets a base minimum for the playout delay. Base minimum delay sets lower
   // bound on minimum delay value determining lower bound on playout delay.
   //
@@ -292,16 +261,6 @@ class VideoReceiveStream {
   // Returns current value of base minimum delay in milliseconds.
   virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
 
-  // Allows a FrameDecryptor to be attached to a VideoReceiveStream after
-  // creation without resetting the decoder state.
-  virtual void SetFrameDecryptor(
-      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) = 0;
-
-  // Allows a frame transformer to be attached to a VideoReceiveStream after
-  // creation without resetting the decoder state.
-  virtual void SetDepacketizerToDecoderFrameTransformer(
-      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
-
   // Sets and returns recording state. The old state is moved out
   // of the video receive stream and returned to the caller, and |state|
   // is moved in. If the state's callback is set, it will be called with