Reland "Upconvert various types to int.", misc. codecs portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as well as webrtc/modules/audio_coding/main/test/opus_test.cc, are relanded. The original commit message is below: Upconvert various types to int. Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t. Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C." This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change. BUG=none TBR=kwiberg Review URL: https://codereview.webrtc.org/1179093003 Cr-Commit-Position: refs/heads/master@{#9424}
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@ -273,17 +273,11 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
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int16_t bitstream_len_byte;
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uint8_t bitstream[kMaxBytes];
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for (int i = 0; i < loop_encode; i++) {
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if (channels == 1) {
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bitstream_len_byte = WebRtcOpus_Encode(
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opus_mono_encoder_, &audio[read_samples],
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frame_length, kMaxBytes, bitstream);
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ASSERT_GE(bitstream_len_byte, 0);
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} else {
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bitstream_len_byte = WebRtcOpus_Encode(
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opus_stereo_encoder_, &audio[read_samples],
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frame_length, kMaxBytes, bitstream);
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ASSERT_GE(bitstream_len_byte, 0);
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}
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int bitstream_len_byte_int = WebRtcOpus_Encode(
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(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
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&audio[read_samples], frame_length, kMaxBytes, bitstream);
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ASSERT_GE(bitstream_len_byte_int, 0);
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bitstream_len_byte = static_cast<int16_t>(bitstream_len_byte_int);
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// Simulate packet loss by setting |packet_loss_| to "true" in
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// |percent_loss| percent of the loops.
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