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; |