| 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));
|
| }
|
|
|
|
|