Remove TRACE_ASYNC without matching TRACE_BEGIN in rtc_sender_audio
This seems to confuse perfetto, and the data ends up on its own track
and the end event is just ignored. As it was invalid, I am assuming it
is not used, and can be simply removed.
#rtc_fixit
Bug: webrtc:15867
Change-Id: I77e59adcd35c51911474446a5f92505bf6b860f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342780
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41892}
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 7351f31..57fb4f4 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -31,26 +31,10 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
-#include "rtc_base/trace_event.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
-namespace {
-[[maybe_unused]] const char* FrameTypeToString(AudioFrameType frame_type) {
- switch (frame_type) {
- case AudioFrameType::kEmptyFrame:
- return "empty";
- case AudioFrameType::kAudioFrameSpeech:
- return "audio_speech";
- case AudioFrameType::kAudioFrameCN:
- return "audio_cn";
- }
- RTC_CHECK_NOTREACHED();
-}
-
-} // namespace
-
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
: clock_(clock),
rtp_sender_(rtp_sender),
@@ -145,8 +129,6 @@
bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) {
RTC_DCHECK_GE(frame.payload_id, 0);
RTC_DCHECK_LE(frame.payload_id, 127);
- TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", frame.rtp_timestamp, "Send",
- "type", FrameTypeToString(frame.type));
// From RFC 4733:
// A source has wide latitude as to how often it sends event updates. A
@@ -279,9 +261,6 @@
MutexLock lock(&send_audio_mutex_);
last_payload_type_ = frame.payload_id;
}
- TRACE_EVENT_ASYNC_END2("webrtc", "Audio", frame.rtp_timestamp, "timestamp",
- packet->Timestamp(), "seqnum",
- packet->SequenceNumber());
packet->set_packet_type(RtpPacketMediaType::kAudio);
packet->set_allow_retransmission(true);
std::vector<std::unique_ptr<RtpPacketToSend>> packets(1);