Bug 1242199: Add lower-limit WebRTC bandwidth pref for testing r=pkerr

--HG--
extra : commitid : EHEwqw6bVUo

dom/media/tests/mochitest/mochitest.ini

@@ -139,7 +139,7 @@ skip-if = toolkit == 'gonk' || buildapp == 'mulet' || (android_version == '18' &
 [test_peerConnection_promiseSendOnly.html]
 skip-if = toolkit == 'gonk' || buildapp == 'mulet' || (android_version == '18' && debug) # b2g(Bug 960442, video support for WebRTC is disabled on b2g), android(Bug 1189784, timeouts on 4.3 emulator)
 [test_peerConnection_simulcastOffer.html]
-skip-if = toolkit == 'gonk' || buildapp == 'mulet' || android_version  || (os == 'linux' && debug) # b2g(Bug 960442, video support for WebRTC is disabled on b2g), no simulcast support on android, linux (bug 1242058)
+skip-if = toolkit == 'gonk' || buildapp == 'mulet' || android_version # b2g(Bug 960442, video support for WebRTC is disabled on b2g), no simulcast support on android
 #[test_peerConnection_relayOnly.html]
 #skip-if = toolkit == 'gonk' || buildapp == 'mulet' # b2g(Bug 960442, video support for WebRTC is disabled on b2g)
 [test_peerConnection_callbacks.html]

dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html

@@ -52,7 +52,9 @@
   ]);
 
   runNetworkTest(() =>
-    pushPrefs(['media.peerconnection.simulcast', true]).then(() => {
+    pushPrefs(['media.peerconnection.simulcast', true],
+              ['media.peerconnection.video.min_bitrate_estimate', 100*1000]).then(() => {
+      SimpleTest.requestCompleteLog();
       var stream;
       var helper = new CaptureStreamTestHelper2D(50,50);
       var canvas = helper.createAndAppendElement('canvas', 'source_canvas');

media/webrtc/signaling/src/media-conduit/VideoConduit.cpp

@@ -93,6 +93,7 @@ WebrtcVideoConduit::WebrtcVideoConduit():
   mMinBitrate(0),
   mStartBitrate(0),
   mMaxBitrate(0),
+  mMinBitrateEstimate(0),
   mCodecMode(webrtc::kRealtimeVideo)
 {}
 
@@ -295,6 +296,10 @@ WebrtcVideoConduit::InitMain()
       if (mStartBitrate > mMaxBitrate) {
         mStartBitrate = mMaxBitrate;
       }
+      (void) NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.min_bitrate_estimate", &temp)));
+      if (temp >= 0) {
+        mMinBitrateEstimate = temp;
+      }
       bool use_loadmanager = false;
       (void) NS_WARN_IF(NS_FAILED(branch->GetBoolPref("media.navigator.load_adapt", &use_loadmanager)));
       if (use_loadmanager) {
@@ -721,6 +726,15 @@ WebrtcVideoConduit::ConfigureSendMediaCodec(const VideoCodecConfig* codecConfig)
     return kMediaConduitUnknownError;
   }
 
+  if (mMinBitrateEstimate != 0) {
+    mPtrViENetwork->SetBitrateConfig(mChannel,
+                                     mMinBitrateEstimate,
+                                     std::max(video_codec.startBitrate,
+                                              mMinBitrateEstimate),
+                                     std::max(video_codec.maxBitrate,
+                                              mMinBitrateEstimate));
+  }
+
   if (!mVideoCodecStat) {
     mVideoCodecStat = new VideoCodecStatistics(mChannel, mPtrViECodec);
   }
@@ -1040,6 +1054,12 @@ WebrtcVideoConduit::SelectBandwidth(webrtc::VideoCodec& vie_codec,
   if (mStartBitrate && mStartBitrate > vie_codec.startBitrate) {
     vie_codec.startBitrate = mStartBitrate;
   }
+  vie_codec.startBitrate = std::max(vie_codec.startBitrate, vie_codec.minBitrate);
+
+  // Note: mMaxBitrate is the max transport bitrate - it applies to a
+  // single codec encoding, but should also apply to the sum of all
+  // simulcast layers in this encoding!
+  // So sum(layers.maxBitrate) <= mMaxBitrate
   if (mMaxBitrate && mMaxBitrate > vie_codec.maxBitrate) {
     vie_codec.maxBitrate = mMaxBitrate;
   }
@@ -1307,6 +1327,15 @@ WebrtcVideoConduit::ReconfigureSendCodec(unsigned short width,
                   __FUNCTION__, width, height, err);
       return NS_ERROR_FAILURE;
     }
+    if (mMinBitrateEstimate != 0) {
+      mPtrViENetwork->SetBitrateConfig(mChannel,
+                                       mMinBitrateEstimate,
+                                       std::max(vie_codec.startBitrate,
+                                                mMinBitrateEstimate),
+                                       std::max(vie_codec.maxBitrate,
+                                                mMinBitrateEstimate));
+    }
+
     CSFLogDebug(logTag, "%s: Encoder resolution changed to %ux%u @ %ufps, bitrate %u:%u",
                 __FUNCTION__, width, height, mSendingFramerate,
                 vie_codec.minBitrate, vie_codec.maxBitrate);

media/webrtc/signaling/src/media-conduit/VideoConduit.h

@@ -402,6 +402,7 @@ private:
   uint32_t mMinBitrate;
   uint32_t mStartBitrate;
   uint32_t mMaxBitrate;
+  uint32_t mMinBitrateEstimate;
 
   static const unsigned int sAlphaNum = 7;
   static const unsigned int sAlphaDen = 8;

modules/libpref/init/all.js

@@ -415,6 +415,7 @@ pref("media.getusermedia.browser.enabled", true);
 pref("media.peerconnection.video.min_bitrate", 0);
 pref("media.peerconnection.video.start_bitrate", 0);
 pref("media.peerconnection.video.max_bitrate", 0);
+pref("media.peerconnection.video.min_bitrate_estimate", 0);
 pref("media.navigator.audio.fake_frequency", 1000);
 pref("media.navigator.permission.disabled", false);
 pref("media.peerconnection.simulcast", true);