| Index: webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
|
| index df14081febe5c8fedfa8e53607ee5898470a6ba5..77787d87bdfe581557329be31877005836d5ed0a 100644
|
| --- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
|
| @@ -12,6 +12,7 @@
|
|
|
| #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
|
| #include "webrtc/base/checks.h"
|
| +#include "webrtc/common_types.h"
|
| #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
|
| #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
|
| #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
|
| @@ -24,7 +25,6 @@
|
| using webrtc::NetEq;
|
| using webrtc::test::AudioLoop;
|
| using webrtc::test::RtpGenerator;
|
| -using webrtc::WebRtcRTPHeader;
|
|
|
| namespace webrtc {
|
| namespace test {
|
| @@ -59,7 +59,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
|
| int32_t time_now_ms = 0;
|
|
|
| // Get first input packet.
|
| - WebRtcRTPHeader rtp_header;
|
| + RTPHeader rtp_header;
|
| RtpGenerator rtp_gen(kSampRateHz / 1000);
|
| // Start with positive drift first half of simulation.
|
| rtp_gen.set_drift_factor(drift_factor);
|
| @@ -83,12 +83,12 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
|
| // Drop every N packets, where N = FLAGS_lossrate.
|
| bool lost = false;
|
| if (lossrate > 0) {
|
| - lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
|
| + lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;
|
| }
|
| if (!lost) {
|
| // Insert packet.
|
| int error =
|
| - neteq->InsertPacket(rtp_header.header, input_payload,
|
| + neteq->InsertPacket(rtp_header, input_payload,
|
| packet_input_time_ms * kSampRateHz / 1000);
|
| if (error != NetEq::kOK)
|
| return -1;
|
|
|