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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@ -92,8 +92,7 @@ RtpPacketizerH264::RtpPacketizerH264(size_t max_payload_len,
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RTC_CHECK_GT(max_payload_len, last_packet_reduction_len);
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}
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RtpPacketizerH264::~RtpPacketizerH264() {
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}
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RtpPacketizerH264::~RtpPacketizerH264() {}
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RtpPacketizerH264::Fragment::~Fragment() = default;
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@ -582,8 +581,8 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
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uint32_t pps_id;
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uint32_t sps_id;
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if (PpsParser::ParsePpsIds(&payload_data[start_offset],
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end_offset - start_offset, &pps_id,
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&sps_id)) {
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end_offset - start_offset, &pps_id,
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&sps_id)) {
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nalu.pps_id = pps_id;
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nalu.sps_id = sps_id;
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} else {
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