| 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());
|
|
|