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Upconvert various types to int.

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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
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54629004

Cr-Commit-Position: refs/heads/master@{#9405}
This commit is contained in:
Peter Kasting
2015-06-09 17:19:57 -07:00
parent 54b0ca553f
commit 83ad33a8ae
61 changed files with 314 additions and 314 deletions
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16
webrtc/modules/audio_coding/main/test/opus_test.cc
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@ -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_GT(bitstream_len_byte, -1);
} else {
bitstream_len_byte = WebRtcOpus_Encode(
opus_stereo_encoder_, &audio[read_samples],
frame_length, kMaxBytes, bitstream);
ASSERT_GT(bitstream_len_byte, -1);
}
int bitstream_len_byte_int = WebRtcOpus_Encode(
(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
&audio[read_samples], frame_length, kMaxBytes, bitstream);
ASSERT_GT(bitstream_len_byte_int, -1);
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.