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}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 6512515..3a3ad98 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -788,7 +788,8 @@
}
case kAudioRepetitionIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
- sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
+ sync_buffer_->IncreaseEndTimestamp(
+ static_cast<uint32_t>(output_size_samples_));
// Skipping break on purpose. Execution should move on into the
// next case.
FALLTHROUGH();
@@ -881,7 +882,7 @@
}
} else {
// Use dead reckoning to estimate the |playout_timestamp_|.
- playout_timestamp_ += output_size_samples_;
+ playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
}
if (decode_return_value) return decode_return_value;
@@ -940,9 +941,10 @@
}
// Check if it is time to play a DTMF event.
- if (dtmf_buffer_->GetEvent(end_timestamp +
- decision_logic_->generated_noise_samples(),
- dtmf_event)) {
+ if (dtmf_buffer_->GetEvent(
+ static_cast<uint32_t>(
+ end_timestamp + decision_logic_->generated_noise_samples()),
+ dtmf_event)) {
*play_dtmf = true;
}
@@ -1030,7 +1032,8 @@
if (decision_logic_->generated_noise_samples() > 0 &&
last_mode_ != kModeDtmf) {
// Make a jump in timestamp due to the recently played comfort noise.
- uint32_t timestamp_jump = decision_logic_->generated_noise_samples();
+ uint32_t timestamp_jump =
+ static_cast<uint32_t>(decision_logic_->generated_noise_samples());
sync_buffer_->IncreaseEndTimestamp(timestamp_jump);
timestamp_ += timestamp_jump;
}
@@ -1224,7 +1227,8 @@
if (*decoded_length < 0) {
// Error returned from the decoder.
*decoded_length = 0;
- sync_buffer_->IncreaseEndTimestamp(decoder_frame_length_);
+ sync_buffer_->IncreaseEndTimestamp(
+ static_cast<uint32_t>(decoder_frame_length_));
int error_code = 0;
if (decoder)
error_code = decoder->ErrorCode();
@@ -1719,7 +1723,8 @@
// algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
// }
- sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
+ sync_buffer_->IncreaseEndTimestamp(
+ static_cast<uint32_t>(output_size_samples_));
expand_->Reset();
last_mode_ = kModeDtmf;
@@ -1749,7 +1754,7 @@
stats_.AddZeros(length);
}
if (increase_timestamp) {
- sync_buffer_->IncreaseEndTimestamp(length);
+ sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
}
expand_->Reset();
}