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Unified Diff: webrtc/modules/audio_coding/neteq/neteq_impl.cc

Issue 2290153002: NetEq: Change member variables for current RTP types to rtc::Optionals (Closed)
Patch Set: Created 4 years, 4 months ago
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Index: webrtc/modules/audio_coding/neteq/neteq_impl.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 78e31121848d166826239086835678c051280cd7..e023c0377765e5e61399e1e90027f7346551a318 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -95,8 +95,6 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
new_codec_(false),
timestamp_(0),
reset_decoder_(false),
- current_rtp_payload_type_(0xFF), // Invalid RTP payload type.
- current_cng_rtp_payload_type_(0xFF), // Invalid RTP payload type.
ssrc_(0),
first_packet_(true),
error_code_(0),
@@ -537,10 +535,10 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
<< static_cast<int>(rtp_header.header.payloadType);
return kSyncPacketNotAccepted;
}
- if (first_packet_ ||
- rtp_header.header.payloadType != current_rtp_payload_type_ ||
+ if (first_packet_ || !current_rtp_payload_type_ ||
+ rtp_header.header.payloadType != *current_rtp_payload_type_ ||
rtp_header.header.ssrc != ssrc_) {
- // Even if |current_rtp_payload_type_| is 0xFF, sync-packet isn't
+ // Even if |current_rtp_payload_type_| is empty, sync-packet isn't
// accepted.
LOG_F(LS_ERROR)
<< "Changing codec, SSRC or first packet with sync-packet.";
@@ -743,9 +741,9 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
new_codec_ = true;
}
- RTC_DCHECK(current_rtp_payload_type_ == 0xFF ||
- decoder_database_->GetDecoderInfo(current_rtp_payload_type_))
- << "Payload type " << static_cast<int>(current_rtp_payload_type_)
+ RTC_DCHECK(!current_rtp_payload_type_ ||
+ decoder_database_->GetDecoderInfo(*current_rtp_payload_type_))
+ << "Payload type " << static_cast<int>(*current_rtp_payload_type_)
<< " is unknown where it shouldn't be";
kwiberg-webrtc 2016/08/30 10:29:32 If !current_rtp_payload_type_, you evaluate *curre
hlundin-webrtc 2016/08/30 10:53:16 Oh, that's not good. Changed. Done.
if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {

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