Remove the deprecated EncodeInternal interface from AudioEncoder

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index a599e291d479b0ba3926e0d8d7e14e5dbaf5f75c..e57487c1d2f97c874b22cbd5d030d96d30af0e2f 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
+#include <algorithm>

[kwiberg-webrtc, 2016/04/07 08:23:16]

Why is this needed?

[ossu, 2016/04/07 08:52:53]

It's for std::min and std::max. It's another lint issue that I might've put in a
separate CL. Would you like me to put that separate in together with the other
small lint fix?

[kwiberg-webrtc, 2016/04/07 09:53:26]

That would be ideal. (I won't insist, though, since this one time isn't a big
deal. But there's no time like the present when it comes to establishing a new
good habit...)

+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
@@ -100,16 +102,6 @@ AudioEncoderOpus::~AudioEncoderOpus() {
   RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
 }
 
-size_t AudioEncoderOpus::MaxEncodedBytes() const {
-  // Calculate the number of bytes we expect the encoder to produce,
-  // then multiply by two to give a wide margin for error.
-  const size_t bytes_per_millisecond =
-      static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
-  const size_t approx_encoded_bytes =
-      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
-  return 2 * approx_encoded_bytes;
-}
-
 int AudioEncoderOpus::SampleRateHz() const {
   return kSampleRateHz;
 }
@@ -231,6 +223,16 @@ size_t AudioEncoderOpus::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
 }
 
+size_t AudioEncoderOpus::MaxEncodedBytes() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
 // If the given config is OK, recreate the Opus encoder instance with those
 // settings, save the config, and return true. Otherwise, do nothing and return
 // false.