Index: webrtc/call/rtc_event_log_unittest.cc |
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc |
index 70032039303129242ea52cfb4509adc361993c80..b86a6885674d6bdeedcf422d1528c74b08dfffa1 100644 |
--- a/webrtc/call/rtc_event_log_unittest.cc |
+++ b/webrtc/call/rtc_event_log_unittest.cc |
@@ -350,7 +350,7 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector, |
return header_size; |
} |
-rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(Random* prng) { |
+rtc::Buffer GenerateRtcpPacket(Random* prng) { |
rtcp::ReportBlock report_block; |
report_block.To(prng->Rand<uint32_t>()); // Remote SSRC. |
report_block.WithFractionLost(prng->Rand(50)); |
@@ -427,7 +427,7 @@ void LogSessionAndReadBack(size_t rtp_count, |
ASSERT_LE(playout_count, rtp_count); |
ASSERT_LE(bwe_loss_count, rtp_count); |
std::vector<rtc::Buffer> rtp_packets; |
- std::vector<rtc::scoped_ptr<rtcp::RawPacket> > rtcp_packets; |
+ std::vector<rtc::Buffer> rtcp_packets; |
std::vector<size_t> rtp_header_sizes; |
std::vector<uint32_t> playout_ssrcs; |
std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates; |
@@ -488,8 +488,8 @@ void LogSessionAndReadBack(size_t rtp_count, |
log_dumper->LogRtcpPacket( |
(rtcp_index % 2 == 0) ? kIncomingPacket : kOutgoingPacket, |
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, |
- rtcp_packets[rtcp_index - 1]->Buffer(), |
- rtcp_packets[rtcp_index - 1]->Length()); |
+ rtcp_packets[rtcp_index - 1].data(), |
+ rtcp_packets[rtcp_index - 1].size()); |
rtcp_index++; |
} |
if (i * playout_count >= playout_index * rtp_count) { |
@@ -536,8 +536,8 @@ void LogSessionAndReadBack(size_t rtp_count, |
VerifyRtcpEvent(parsed_stream.stream(event_index), |
rtcp_index % 2 == 0, // Every second packet is incoming. |
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, |
- rtcp_packets[rtcp_index - 1]->Buffer(), |
- rtcp_packets[rtcp_index - 1]->Length()); |
+ rtcp_packets[rtcp_index - 1].data(), |
+ rtcp_packets[rtcp_index - 1].size()); |
event_index++; |
rtcp_index++; |
} |
@@ -604,8 +604,8 @@ void DropOldEvents(uint32_t extensions_bitvector, |
unsigned int random_seed) { |
rtc::Buffer old_rtp_packet; |
rtc::Buffer recent_rtp_packet; |
- rtc::scoped_ptr<rtcp::RawPacket> old_rtcp_packet; |
- rtc::scoped_ptr<rtcp::RawPacket> recent_rtcp_packet; |
+ rtc::Buffer old_rtcp_packet; |
+ rtc::Buffer recent_rtcp_packet; |
VideoReceiveStream::Config receiver_config(nullptr); |
VideoSendStream::Config sender_config(nullptr); |
@@ -647,8 +647,8 @@ void DropOldEvents(uint32_t extensions_bitvector, |
log_dumper->LogRtpHeader(kOutgoingPacket, MediaType::AUDIO, |
old_rtp_packet.data(), old_rtp_packet.size()); |
log_dumper->LogRtcpPacket(kIncomingPacket, MediaType::AUDIO, |
- old_rtcp_packet->Buffer(), |
- old_rtcp_packet->Length()); |
+ old_rtcp_packet.data(), |
+ old_rtcp_packet.size()); |
// Sleep 55 ms to let old events be removed from the queue. |
rtc::Thread::SleepMs(55); |
log_dumper->StartLogging(temp_filename, 10000000); |
@@ -656,8 +656,8 @@ void DropOldEvents(uint32_t extensions_bitvector, |
recent_rtp_packet.data(), |
recent_rtp_packet.size()); |
log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO, |
- recent_rtcp_packet->Buffer(), |
- recent_rtcp_packet->Length()); |
+ recent_rtcp_packet.data(), |
+ recent_rtcp_packet.size()); |
} |
// Read the generated file from disk. |
@@ -675,7 +675,7 @@ void DropOldEvents(uint32_t extensions_bitvector, |
recent_rtp_packet.data(), recent_header_size, |
recent_rtp_packet.size()); |
VerifyRtcpEvent(parsed_stream.stream(4), false, MediaType::VIDEO, |
- recent_rtcp_packet->Buffer(), recent_rtcp_packet->Length()); |
+ recent_rtcp_packet.data(), recent_rtcp_packet.size()); |
// Clean up temporary file - can be pretty slow. |
remove(temp_filename.c_str()); |