Remove webrtc::MinPositive from api/.

Follow-up of https://webrtc-review.googlesource.com/c/src/+/153220,
where during code review it was suggested to move webrtc::MinPositive
out of the api/ directory.

Bug: None
Change-Id: I0c3b87a9ffd1cd205a85dddd9f44cfd95eb02206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153480
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29220}

api/transport/bitrate_settings.h

@@ -45,18 +45,6 @@ struct BitrateConstraints {
   static constexpr int kDefaultStartBitrateBps = 300000;
 };
 
-// Like std::min, but considers non-positive values to be unset.
-template <typename T>
-static T MinPositive(T a, T b) {
-  if (a <= 0) {
-    return b;
-  }
-  if (b <= 0) {
-    return a;
-  }
-  return std::min(a, b);
-}
-
 }  // namespace webrtc
 
 #endif  // API_TRANSPORT_BITRATE_SETTINGS_H_

call/rtp_bitrate_configurator.cc

@@ -14,6 +14,22 @@
 
 #include "rtc_base/checks.h"
 
+namespace {
+
+// Returns its smallest positive argument. If neither argument is positive,
+// returns an arbitrary nonpositive value.
+int MinPositive(int a, int b) {
+  if (a <= 0) {
+    return b;
+  }
+  if (b <= 0) {
+    return a;
+  }
+  return std::min(a, b);
+}
+
+}  // namespace
+
 namespace webrtc {
 RtpBitrateConfigurator::RtpBitrateConfigurator(
     const BitrateConstraints& bitrate_config)

media/engine/webrtc_video_engine.cc

@@ -337,6 +337,19 @@ int GetMinVideoBitrateBps(webrtc::VideoCodecType type) {
   }
   return kMinVideoBitrateBps;
 }
+
+// Returns its smallest positive argument. If neither argument is positive,
+// returns an arbitrary nonpositive value.
+int MinPositive(int a, int b) {
+  if (a <= 0) {
+    return b;
+  }
+  if (b <= 0) {
+    return a;
+  }
+  return std::min(a, b);
+}
+
 }  // namespace
 
 // This constant is really an on/off, lower-level configurable NACK history
@@ -2217,8 +2230,8 @@ WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
   if (rtp_parameters_.encodings[0].max_bitrate_bps &&
       rtp_parameters_.encodings.size() == 1) {
     stream_max_bitrate =
-        webrtc::MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps),
-                            parameters_.max_bitrate_bps);
+        MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps),
+                    parameters_.max_bitrate_bps);
   }
 
   // The codec max bitrate comes from the "x-google-max-bitrate" parameter

media/engine/webrtc_voice_engine.cc

@@ -141,6 +141,18 @@ absl::optional<std::string> GetAudioNetworkAdaptorConfig(
   return absl::nullopt;
 }
 
+// Returns its smallest positive argument. If neither argument is positive,
+// returns an arbitrary nonpositive value.
+int MinPositive(int a, int b) {
+  if (a <= 0) {
+    return b;
+  }
+  if (b <= 0) {
+    return a;
+  }
+  return std::min(a, b);
+}
+
 // |max_send_bitrate_bps| is the bitrate from "b=" in SDP.
 // |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters.
 absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
@@ -148,10 +160,9 @@ absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
                                        const webrtc::AudioCodecSpec& spec) {
   // If application-configured bitrate is set, take minimum of that and SDP
   // bitrate.
-  const int bps =
-      rtp_max_bitrate_bps
-          ? webrtc::MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
-          : max_send_bitrate_bps;
+  const int bps = rtp_max_bitrate_bps
+                      ? MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
+                      : max_send_bitrate_bps;
   if (bps <= 0) {
     return spec.info.default_bitrate_bps;
   }