Match existing type usage better.

This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones.  For example:

* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps.  For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika

Review URL: https://codereview.webrtc.org/1168753002

Cr-Commit-Position: refs/heads/master@{#9419}

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

@@ -115,9 +115,9 @@ 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.
   int frame_size_ms = num_10ms_frames_per_packet_ * 10;
-  int bytes_per_millisecond = bitrate_bps_ / (1000 * 8) + 1;
-  size_t approx_encoded_bytes =
-      static_cast<size_t>(frame_size_ms * bytes_per_millisecond);
+  size_t bytes_per_millisecond =
+      static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
+  size_t approx_encoded_bytes = frame_size_ms * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
 }
 
@@ -206,7 +206,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
   CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
   EncodedInfo info;
-  info.encoded_bytes = status;
+  info.encoded_bytes = static_cast<size_t>(status);
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.

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

@@ -196,7 +196,7 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
       EncodeABlock();
 
       // Check if payload has FEC.
-      int16_t fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
+      int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
 
       // If FEC is disabled or the target packet loss rate is set to 0, there
       // should be no FEC in the bit stream.