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Match existing type usage better.

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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}
This commit is contained in:
Peter Kasting
2015-06-11 12:55:50 -07:00
parent cb180976dd
commit b7e5054414
57 changed files with 175 additions and 151 deletions
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15
webrtc/modules/audio_coding/neteq/merge.cc
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@ -108,10 +108,11 @@ int Merge::Process(int16_t* input, size_t input_length,
// Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
// and so on.
int increment = 4194 / fs_mult_;
*external_mute_factor = DspHelper::RampSignal(input_channel,
interpolation_length,
*external_mute_factor,
increment);
*external_mute_factor =
static_cast<int16_t>(DspHelper::RampSignal(input_channel,
interpolation_length,
*external_mute_factor,
increment));
DspHelper::UnmuteSignal(&input_channel[interpolation_length],
input_length_per_channel - interpolation_length,
external_mute_factor, increment,
@ -125,7 +126,8 @@ int Merge::Process(int16_t* input, size_t input_length,
}
// Do overlap and mix linearly.
int increment = 16384 / (interpolation_length + 1); // In Q14.
int16_t increment =
static_cast<int16_t>(16384 / (interpolation_length + 1)); // In Q14.
int16_t mute_factor = 16384 - increment;
memmove(temp_data, expanded_channel,
sizeof(int16_t) * best_correlation_index);
@ -246,7 +248,8 @@ int16_t Merge::SignalScaling(const int16_t* input, int input_length,
// energy_expanded / energy_input is in Q14.
energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
// Calculate sqrt(energy_expanded / energy_input) in Q14.
mute_factor = WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14);
mute_factor = static_cast<int16_t>(
WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14));
} else {
// Set to 1 (in Q14) when |expanded| has higher energy than |input|.
mute_factor = 16384;