Merge "Merge upstream-master and enable ARM64" into emu-master-devemu-31-stable-releaseemu-31-release

1 files changed, 16 insertions, 32 deletions

diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 6f74492927..54729d0dc0 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -20,17 +20,14 @@
 #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/frame_transformer_interface.h"
 #include "api/rtp_parameters.h"
-#include "api/scoped_refptr.h"
-#include "api/transport/rtp/rtp_source.h"
+#include "call/receive_stream.h"
 #include "call/rtp_config.h"
 
 namespace webrtc {
 class AudioSinkInterface;
 
-class AudioReceiveStream {
+class AudioReceiveStream : public MediaReceiveStream {
  public:
   struct Stats {
     Stats();
@@ -106,29 +103,14 @@ class AudioReceiveStream {
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
-    struct Rtp {
+    struct Rtp : public RtpConfig {
       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;
-
-      // Enable feedback for send side bandwidth estimation.
-      // See
-      // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
-      // for details.
-      bool transport_cc = false;
-
       // See NackConfig for description.
       NackConfig nack;
-
-      // RTP header extensions used for the received stream.
-      std::vector<RtpExtension> extensions;
     } rtp;
 
     Transport* rtcp_send_transport = nullptr;
@@ -157,22 +139,26 @@ class AudioReceiveStream {
     // 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.
+    // TODO(tommi): Remove this member variable from the struct. It's not
+    // a part of the AudioReceiveStream state but rather a pass through
+    // variable.
     rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor;
 
     // An optional frame transformer used by insertable streams to transform
     // encoded frames.
+    // TODO(tommi): Remove this member variable from the struct. It's not
+    // a part of the AudioReceiveStream state but rather a pass through
+    // variable.
     rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
   };
 
-  // Reconfigure the stream according to the Configuration.
-  virtual void Reconfigure(const Config& config) = 0;
-
-  // 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;
+  // Methods that support reconfiguring the stream post initialization.
+  virtual void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) = 0;
+  virtual void SetUseTransportCcAndNackHistory(bool use_transport_cc,
+                                               int history_ms) = 0;
+  // Set/change the rtp header extensions. Must be called on the packet
+  // delivery thread.
+  virtual void SetRtpExtensions(std::vector<RtpExtension> extensions) = 0;
 
   // Returns true if the stream has been started.
   virtual bool IsRunning() const = 0;
@@ -202,8 +188,6 @@ class AudioReceiveStream {
   // Returns current value of base minimum delay in milliseconds.
   virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
 
-  virtual std::vector<RtpSource> GetSources() const = 0;
-
  protected:
   virtual ~AudioReceiveStream() {}
 };