Renames RtcEventLogParseNew to RtcEventLogParser
Bug: webrtc:10170 Change-Id: I9232c276229a64fa4d8321b6c996387fe130f68b Reviewed-on: https://webrtc-review.googlesource.com/c/116064 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26128}
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@ -24,10 +24,10 @@ namespace webrtc {
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namespace test {
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namespace {
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bool ShouldSkipStream(ParsedRtcEventLogNew::MediaType media_type,
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bool ShouldSkipStream(ParsedRtcEventLog::MediaType media_type,
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uint32_t ssrc,
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absl::optional<uint32_t> ssrc_filter) {
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if (media_type != ParsedRtcEventLogNew::MediaType::AUDIO)
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if (media_type != ParsedRtcEventLog::MediaType::AUDIO)
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return true;
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if (ssrc_filter.has_value() && ssrc != *ssrc_filter)
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return true;
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@ -65,7 +65,7 @@ RtcEventLogSource::RtcEventLogSource() : PacketSource() {}
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bool RtcEventLogSource::OpenFile(const std::string& file_name,
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absl::optional<uint32_t> ssrc_filter) {
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ParsedRtcEventLogNew parsed_log;
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ParsedRtcEventLog parsed_log;
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if (!parsed_log.ParseFile(file_name))
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return false;
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@ -104,7 +104,7 @@ bool RtcEventLogSource::OpenFile(const std::string& file_name,
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// This wouldn't be needed if we knew that there was at most one audio stream.
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webrtc::RtcEventProcessor event_processor;
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for (const auto& rtp_packets : parsed_log.incoming_rtp_packets_by_ssrc()) {
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ParsedRtcEventLogNew::MediaType media_type =
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ParsedRtcEventLog::MediaType media_type =
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parsed_log.GetMediaType(rtp_packets.ssrc, webrtc::kIncomingPacket);
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if (ShouldSkipStream(media_type, rtp_packets.ssrc, ssrc_filter)) {
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continue;
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@ -16,7 +16,7 @@
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#include <vector>
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#include "absl/types/optional.h"
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#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
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#include "logging/rtc_event_log/rtc_event_log_parser.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructormagic.h"
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