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 7d1f9f9798c6e19b772964b873d35e091a2e15ed..f1577df26ffcd40faba5a8a27fc21285e733cc02 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
@@ -14,6 +14,7 @@ |
#include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" |
+#include "webrtc/modules/include/module_common_types.h" |
#include "webrtc/system_wrappers/include/clock.h" |
#include "webrtc/test/testsupport/fileutils.h" |
#include "webrtc/typedefs.h" |
@@ -103,21 +104,15 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, |
} |
// Get output audio, but don't do anything with it. |
- static const int kMaxChannels = 1; |
- static const size_t kMaxSamplesPerMs = 48000 / 1000; |
- static const int kOutputBlockSizeMs = 10; |
- static const size_t kOutDataLen = |
- kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels; |
- int16_t out_data[kOutDataLen]; |
- size_t num_channels; |
- size_t samples_per_channel; |
- int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel, |
- &num_channels, NULL); |
+ AudioFrame out_frame; |
+ int error = neteq->GetAudio(&out_frame, NULL); |
if (error != NetEq::kOK) |
return -1; |
- assert(samples_per_channel == static_cast<size_t>(kSampRateHz * 10 / 1000)); |
+ assert(out_frame.samples_per_channel_ == |
+ static_cast<size_t>(kSampRateHz * 10 / 1000)); |
+ static const int kOutputBlockSizeMs = 10; |
time_now_ms += kOutputBlockSizeMs; |
if (time_now_ms >= runtime_ms / 2 && !drift_flipped) { |
// Apply negative drift second half of simulation. |