Misc. small cleanups.

* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}

10 files changed, 27 insertions, 26 deletions

diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index 210791cbd9..26c7838861 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -83,7 +83,8 @@ size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
 }
 
 int AudioEncoderPcm::GetTargetBitrate() const {
-  return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
+  return static_cast<int>(
+      8 * BytesPerSample() * SampleRateHz() * NumChannels());
 }
 
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
@@ -122,7 +123,7 @@ size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
   return WebRtcG711_EncodeA(audio, input_len, encoded);
 }
 
-int AudioEncoderPcmA::BytesPerSample() const {
+size_t AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
@@ -135,7 +136,7 @@ size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
-int AudioEncoderPcmU::BytesPerSample() const {
+size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index fd996dca75..6891cbdc3a 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -54,7 +54,7 @@ class AudioEncoderPcm : public AudioEncoder {
                             size_t input_len,
                             uint8_t* encoded) = 0;
 
-  virtual int BytesPerSample() const = 0;
+  virtual size_t BytesPerSample() const = 0;
 
  private:
   const int sample_rate_hz_;
@@ -83,7 +83,7 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
                     size_t input_len,
                     uint8_t* encoded) override;
 
-  int BytesPerSample() const override;
+  size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
@@ -105,7 +105,7 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
                     size_t input_len,
                     uint8_t* encoded) override;
 
-  int BytesPerSample() const override;
+  size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index 632a4fe825..32f36c5261 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -92,7 +92,7 @@ float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream,
   value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes,
                                out_data, &audio_type);
   clocks = clock() - clocks;
-  EXPECT_EQ(output_length_sample_, value);
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
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 3e7d3ec738..0806bb81d9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -137,15 +137,14 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
     uint8_t* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
-  RTC_DCHECK_EQ(static_cast<size_t>(SamplesPer10msFrame()), audio.size());
+  RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
   input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
-      (static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
-  RTC_CHECK_EQ(
-      input_buffer_.size(),
-      static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame());
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
       rtc::CheckedDivExact(input_buffer_.size(),
@@ -214,11 +213,11 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps));
 }
 
-int AudioEncoderOpus::Num10msFramesPerPacket() const {
-  return rtc::CheckedDivExact(config_.frame_size_ms, 10);
+size_t AudioEncoderOpus::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
 }
 
-int AudioEncoderOpus::SamplesPer10msFrame() const {
+size_t AudioEncoderOpus::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 36011fab74..f37e344d4d 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -85,8 +85,8 @@ class AudioEncoderOpus final : public AudioEncoder {
   bool dtx_enabled() const { return config_.dtx_enabled; }
 
  private:
-  int Num10msFramesPerPacket() const;
-  int SamplesPer10msFrame() const;
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
   bool RecreateEncoderInstance(const Config& config);
 
   Config config_;
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
index f95cc7145d..4d1aa42c89 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -77,7 +77,7 @@ float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
   value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
                             &audio_type);
   clocks = clock() - clocks;
-  EXPECT_EQ(output_length_sample_, value);
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
index 50d2041b83..f4d4022302 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -22,7 +22,7 @@ size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
   return WebRtcPcm16b_Encode(audio, input_len, encoded);
 }
 
-int AudioEncoderPcm16B::BytesPerSample() const {
+size_t AudioEncoderPcm16B::BytesPerSample() const {
   return 2;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index 3645a6f718..68ca2da77e 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -37,7 +37,7 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm {
                     size_t input_len,
                     uint8_t* encoded) override;
 
-  int BytesPerSample() const override;
+  size_t BytesPerSample() const override;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B);
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index 3395721f8b..07a15ff578 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -23,8 +23,10 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
     : block_duration_ms_(block_duration_ms),
       input_sampling_khz_(input_sampling_khz),
       output_sampling_khz_(output_sampling_khz),
-      input_length_sample_(block_duration_ms_ * input_sampling_khz_),
-      output_length_sample_(block_duration_ms_ * output_sampling_khz_),
+      input_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
+      output_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
       data_pointer_(0),
       loop_length_samples_(0),
       max_bytes_(0),
@@ -65,8 +67,7 @@ void AudioCodecSpeedTest::SetUp() {
   memcpy(&in_data_[loop_length_samples_], &in_data_[0],
          input_length_sample_ * channels_ * sizeof(int16_t));
 
-  max_bytes_ =
-      static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
+  max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
   out_data_.reset(new int16_t[output_length_sample_ * channels_]);
   bit_stream_.reset(new uint8_t[max_bytes_]);
 
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index 2736c2912e..b5aef75e95 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -55,10 +55,10 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   int output_sampling_khz_;
 
   // Number of samples-per-channel in a frame.
-  int input_length_sample_;
+  size_t input_length_sample_;
 
   // Expected output number of samples-per-channel in a frame.
-  int output_length_sample_;
+  size_t output_length_sample_;
 
   rtc::scoped_ptr<int16_t[]> in_data_;
   rtc::scoped_ptr<int16_t[]> out_data_;