Replace DataSize and DataRate factories with newer versions

This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format

Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30545}
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 8600a67..e7c6540 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -82,7 +82,7 @@
 // TODO(dklee): This mirrors calculation in audio_send_stream.cc, which
 // should be made more precise in the future. This can be changed when that
 // logic is more accurate.
-const DataSize kOverheadPerPacket = DataSize::bytes(20 + 8 + 10 + 12);
+const DataSize kOverheadPerPacket = DataSize::Bytes(20 + 8 + 10 + 12);
 const TimeDelta kMinFrameLength = TimeDelta::Millis(20);
 const TimeDelta kMaxFrameLength = TimeDelta::Millis(120);
 const DataRate kMinOverheadRate = kOverheadPerPacket / kMaxFrameLength;
@@ -549,11 +549,12 @@
   ConfigHelper helper(false, true);
   auto send_stream = helper.CreateAudioSendStream();
   EXPECT_CALL(*helper.channel_send(),
-              OnBitrateAllocation(
-                  Field(&BitrateAllocationUpdate::target_bitrate,
-                        Eq(DataRate::bps(helper.config().max_bitrate_bps)))));
+              OnBitrateAllocation(Field(
+                  &BitrateAllocationUpdate::target_bitrate,
+                  Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps)))));
   BitrateAllocationUpdate update;
-  update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps + 5000);
+  update.target_bitrate =
+      DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
   update.packet_loss_ratio = 0;
   update.round_trip_time = TimeDelta::Millis(50);
   update.bwe_period = TimeDelta::Millis(6000);
@@ -565,12 +566,14 @@
   ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
   ConfigHelper helper(true, true);
   auto send_stream = helper.CreateAudioSendStream();
-  EXPECT_CALL(*helper.channel_send(),
-              OnBitrateAllocation(Field(
-                  &BitrateAllocationUpdate::target_bitrate,
-                  Eq(DataRate::bps(helper.config().max_bitrate_bps - 5000)))));
+  EXPECT_CALL(
+      *helper.channel_send(),
+      OnBitrateAllocation(Field(
+          &BitrateAllocationUpdate::target_bitrate,
+          Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000)))));
   BitrateAllocationUpdate update;
-  update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps - 5000);
+  update.target_bitrate =
+      DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000);
   helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
                             RTC_FROM_HERE);
 }
@@ -584,9 +587,9 @@
   EXPECT_CALL(
       *helper.channel_send(),
       OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
-                                Eq(DataRate::kbps(6)))));
+                                Eq(DataRate::KilobitsPerSec(6)))));
   BitrateAllocationUpdate update;
-  update.target_bitrate = DataRate::kbps(1);
+  update.target_bitrate = DataRate::KilobitsPerSec(1);
   helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
                             RTC_FROM_HERE);
 }
@@ -600,9 +603,9 @@
   EXPECT_CALL(
       *helper.channel_send(),
       OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
-                                Eq(DataRate::kbps(64)))));
+                                Eq(DataRate::KilobitsPerSec(64)))));
   BitrateAllocationUpdate update;
-  update.target_bitrate = DataRate::kbps(128);
+  update.target_bitrate = DataRate::KilobitsPerSec(128);
   helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
                             RTC_FROM_HERE);
 }
@@ -617,7 +620,7 @@
   EXPECT_CALL(*helper.channel_send(), CallEncoder(_)).Times(1);
   send_stream->OnOverheadChanged(kOverheadPerPacket.bytes<size_t>());
   const DataRate bitrate =
-      DataRate::bps(helper.config().max_bitrate_bps) + kMaxOverheadRate;
+      DataRate::BitsPerSec(helper.config().max_bitrate_bps) + kMaxOverheadRate;
   EXPECT_CALL(*helper.channel_send(),
               OnBitrateAllocation(Field(
                   &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
@@ -637,12 +640,12 @@
   auto send_stream = helper.CreateAudioSendStream();
   EXPECT_CALL(*helper.channel_send(), CallEncoder(_)).Times(1);
   send_stream->OnOverheadChanged(kOverheadPerPacket.bytes<size_t>());
-  const DataRate bitrate = DataRate::kbps(6) + kMinOverheadRate;
+  const DataRate bitrate = DataRate::KilobitsPerSec(6) + kMinOverheadRate;
   EXPECT_CALL(*helper.channel_send(),
               OnBitrateAllocation(Field(
                   &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
   BitrateAllocationUpdate update;
-  update.target_bitrate = DataRate::kbps(1);
+  update.target_bitrate = DataRate::KilobitsPerSec(1);
   helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
                             RTC_FROM_HERE);
 }
@@ -657,12 +660,12 @@
   auto send_stream = helper.CreateAudioSendStream();
   EXPECT_CALL(*helper.channel_send(), CallEncoder(_)).Times(1);
   send_stream->OnOverheadChanged(kOverheadPerPacket.bytes<size_t>());
-  const DataRate bitrate = DataRate::kbps(64) + kMaxOverheadRate;
+  const DataRate bitrate = DataRate::KilobitsPerSec(64) + kMaxOverheadRate;
   EXPECT_CALL(*helper.channel_send(),
               OnBitrateAllocation(Field(
                   &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
   BitrateAllocationUpdate update;
-  update.target_bitrate = DataRate::kbps(128);
+  update.target_bitrate = DataRate::KilobitsPerSec(128);
   helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
                             RTC_FROM_HERE);
 }
@@ -675,7 +678,8 @@
               OnBitrateAllocation(Field(&BitrateAllocationUpdate::bwe_period,
                                         Eq(TimeDelta::Millis(5000)))));
   BitrateAllocationUpdate update;
-  update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps + 5000);
+  update.target_bitrate =
+      DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
   update.packet_loss_ratio = 0;
   update.round_trip_time = TimeDelta::Millis(50);
   update.bwe_period = TimeDelta::Millis(5000);