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 9fe4dffa91e145dae134e2fbb677efda8b91cf70..dbea1c62a2e1ee579db6c8393e463dad7ab0e1be 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
@@ -62,11 +62,12 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, |
bool drift_flipped = false; |
int32_t packet_input_time_ms = |
rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); |
- const int16_t* input_samples = audio_loop.GetNextBlock(); |
- if (!input_samples) exit(1); |
+ auto input_samples = audio_loop.GetNextBlock(); |
+ if (input_samples.empty()) |
+ exit(1); |
uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)]; |
- size_t payload_len = |
- WebRtcPcm16b_Encode(input_samples, kInputBlockSizeSamples, input_payload); |
+ size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(), |
+ input_samples.size(), input_payload); |
assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); |
// Main loop. |
@@ -93,10 +94,10 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, |
kInputBlockSizeSamples, |
&rtp_header); |
input_samples = audio_loop.GetNextBlock(); |
- if (!input_samples) return -1; |
- payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples), |
- kInputBlockSizeSamples, |
- input_payload); |
+ if (input_samples.empty()) |
+ return -1; |
+ payload_len = WebRtcPcm16b_Encode(input_samples.data(), |
+ input_samples.size(), input_payload); |
assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); |
} |