Reformat the WebRTC code base
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
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@ -231,8 +231,8 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id,
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if (!audio_decoder) {
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ret_val = neteq_->RegisterPayloadType(neteq_decoder, name, payload_type);
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} else {
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ret_val = neteq_->RegisterExternalDecoder(
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audio_decoder, neteq_decoder, name, payload_type);
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ret_val = neteq_->RegisterExternalDecoder(audio_decoder, neteq_decoder,
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name, payload_type);
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}
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if (ret_val != NetEq::kOK) {
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RTC_LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
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@ -402,10 +402,9 @@ uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
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// the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
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// We masked 6 most significant bits of 32-bit so there is no overflow in
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// the conversion from milliseconds to timestamp.
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const uint32_t now_in_ms = static_cast<uint32_t>(
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clock_->TimeInMilliseconds() & 0x03ffffff);
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return static_cast<uint32_t>(
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(decoder_sampling_rate / 1000) * now_in_ms);
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const uint32_t now_in_ms =
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static_cast<uint32_t>(clock_->TimeInMilliseconds() & 0x03ffffff);
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return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
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}
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void AcmReceiver::GetDecodingCallStatistics(
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