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Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/

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BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2663063008
Cr-Commit-Position: refs/heads/master@{#16457}
This commit is contained in:
solenberg
2017-02-06 13:39:38 -08:00
committed by Commit bot
parent f748ca47b0
commit 6b34124a6d
3 changed files with 2 additions and 17 deletions
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15
webrtc/audio/audio_receive_stream.cc
View File

@ -69,14 +69,12 @@ AudioReceiveStream::AudioReceiveStream(
webrtc::RtcEventLog* event_log)
: remote_bitrate_estimator_(remote_bitrate_estimator),
config_(config),
audio_state_(audio_state),
rtp_header_parser_(RtpHeaderParser::Create()) {
audio_state_(audio_state) {
LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(packet_router);
RTC_DCHECK(remote_bitrate_estimator);
RTC_DCHECK(rtp_header_parser_);
module_process_thread_checker_.DetachFromThread();
@ -107,14 +105,8 @@ AudioReceiveStream::AudioReceiveStream(
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, extension.id);
RTC_DCHECK(registered);
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber, extension.id);
RTC_DCHECK(registered);
} else {
RTC_NOTREACHED() << "Unsupported RTP extension.";
}
@ -321,11 +313,6 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.CalledOnValidThread());
RTPHeader header;
if (!rtp_header_parser_->Parse(packet, length, &header)) {
return false;
}
return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
}

2
webrtc/audio/audio_receive_stream.h
View File

@ -19,7 +19,6 @@
#include "webrtc/base/thread_checker.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/syncable.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
namespace webrtc {
class PacketRouter;
@ -81,7 +80,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
RemoteBitrateEstimator* const remote_bitrate_estimator_;
const webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
bool playing_ ACCESS_ON(worker_thread_checker_) = false;

2
webrtc/call/call.cc
View File

@ -208,7 +208,7 @@ class Call : public webrtc::Call,
// extensions per SSRC instead, which leads to some storage overhead.
RtpHeaderExtensionMap extensions;
// Set if the RTCP feedback message needed for send side BWE was negotiated.
bool transport_cc;
bool transport_cc = false;
};
std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
GUARDED_BY(receive_crit_);