Add PeerConnectionInterface::UpdateCallBitrate.

webrtc/call/call.h

Index: webrtc/call/call.h
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index f67b6907e5f7a1c3465b984dc23de768f6130631..ae2933a87adfdbf49ff8f32cd848bd7eb5790fb8 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -14,6 +14,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/rtcerror.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/socket.h"
@@ -68,13 +69,24 @@ class Call {
     static const int kDefaultStartBitrateBps;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
-    // used to cap total bitrate used.
+    // used to cap total bitrate used. This comes from the remote connection.
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
       int max_bitrate_bps = -1;
     } bitrate_config;
 
+    // The local client's bitrate preferences. The actual configuration used
+    // is a combination of this and |bitrate_config|. The combination is
+    // currently more complicated than a simple mask operation (see
+    // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
+    // start <= max holds for set parameters.
+    struct BitrateConfigMask {
+      rtc::Optional<int> min_bitrate_bps;
+      rtc::Optional<int> start_bitrate_bps;
+      rtc::Optional<int> max_bitrate_bps;
+    };
+
     // AudioState which is possibly shared between multiple calls.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     rtc::scoped_refptr<AudioState> audio_state;
@@ -141,14 +153,21 @@ class Call {
   // pacing delay, etc.
   virtual Stats GetStats() const = 0;
 
-  // TODO(pbos): Like BitrateConfig above this is currently per-stream instead
-  // of maximum for entire Call. This should be fixed along with the above.
-  // Specifying a start bitrate (>0) will currently reset the current bitrate
-  // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
+  // The greater min and smaller max set by this and SetBitrateConfigMask will
+  // be used. The latest non-negative start value from either call will be used.
+  // Specifying a start bitrate (>0) will reset the current bitrate estimate.

[holmer, 2017/05/31 07:46:34]

I think it would be good to specify what start bitrate == 0 does too.

[Zach Stein, 2017/06/01 03:37:49]

start_bitrate == 0 is disallowed by DCHECK, but it will update the value to
whatever start gets clamped to. Negative values other than -1 behave similarly.
I don't think adding to this comment provides much value since this is supposed
to be an internal header now - reading the code is just as clear and can't get
out of sync. Or maybe I have just been looking at this code for too long :)

[Taylor Brandstetter, 2017/06/01 16:24:43]

The value of a comment here is that it tells you what the method is *designed*
to do, whereas reading the code only tells you what it actually does. In my
opinion, it's ideal if every method in a header file has a sufficient level of
comments that its behavior is clear and tests can be written for it without
looking at the implementation.

+  // This is due to how the 'x-google-start-bitrate' flag is currently
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
 
+  // The greater min and smaller max set by this and SetBitrateConfig will be
+  // used. The latest non-negative start value form either call will be used.
+  // Specifying a start bitrate will reset the current bitrate estimate.
+  // Assumes 0 <= min <= start <= max holds for set parameters.
+  virtual void SetBitrateConfigMask(
+      const Config::BitrateConfigMask& bitrate_mask) = 0;
+
   // TODO(skvlad): When the unbundled case with multiple streams for the same
   // media type going over different networks is supported, track the state
   // for each stream separately. Right now it's global per media type.