Reland "Upconvert various types to int.", misc. codecs portion.

webrtc/modules/audio_coding/main/test/opus_test.cc

Index: webrtc/modules/audio_coding/main/test/opus_test.cc
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
index a407fc5d36f0222897f7792d2f125415a480bf5d..c61d25ad19acd026b11bca7dd528f87a3a19d4be 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -273,17 +273,11 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
       int16_t bitstream_len_byte;
       uint8_t bitstream[kMaxBytes];
       for (int i = 0; i < loop_encode; i++) {
-        if (channels == 1) {
-          bitstream_len_byte = WebRtcOpus_Encode(
-              opus_mono_encoder_, &audio[read_samples],
-              frame_length, kMaxBytes, bitstream);
-          ASSERT_GE(bitstream_len_byte, 0);
-        } else {
-          bitstream_len_byte = WebRtcOpus_Encode(
-              opus_stereo_encoder_, &audio[read_samples],
-              frame_length, kMaxBytes, bitstream);
-          ASSERT_GE(bitstream_len_byte, 0);
-        }
+        int bitstream_len_byte_int = WebRtcOpus_Encode(
+            (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
+            &audio[read_samples], frame_length, kMaxBytes, bitstream);
+        ASSERT_GE(bitstream_len_byte_int, 0);
+        bitstream_len_byte = static_cast<int16_t>(bitstream_len_byte_int);
 
         // Simulate packet loss by setting |packet_loss_| to "true" in
         // |percent_loss| percent of the loops.