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Unified Diff: webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc

Issue 2809153002: Change from WebRtcRTPHeader to RTPHeader in NetEq tests and tools (Closed)
Patch Set: Fixing neteq_rtp_fuzzer Created 3 years, 8 months ago
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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;

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