diff options
Diffstat (limited to 'webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc')
| -rw-r--r-- | webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc | 32 |
1 files changed, 17 insertions, 15 deletions
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 9fe4dffa91..7d1f9f9798 100644 --- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc +++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc @@ -10,7 +10,7 @@ #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h" -#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.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" #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" @@ -34,6 +34,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, const int kSampRateHz = 32000; const webrtc::NetEqDecoder kDecoderType = webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz; + const std::string kDecoderName = "pcm16-swb32"; const int kPayloadType = 95; // Initialize NetEq instance. @@ -41,7 +42,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, config.sample_rate_hz = kSampRateHz; NetEq* neteq = NetEq::Create(config); // Register decoder in |neteq|. - if (neteq->RegisterPayloadType(kDecoderType, kPayloadType) != 0) + if (neteq->RegisterPayloadType(kDecoderType, kDecoderName, kPayloadType) != 0) return -1; // Set up AudioLoop object. @@ -62,12 +63,13 @@ 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); - assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); + size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(), + input_samples.size(), input_payload); + RTC_CHECK_EQ(sizeof(input_payload), payload_len); // Main loop. webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock(); @@ -81,9 +83,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, } if (!lost) { // Insert packet. - int error = neteq->InsertPacket( - rtp_header, input_payload, payload_len, - packet_input_time_ms * kSampRateHz / 1000); + int error = + neteq->InsertPacket(rtp_header, input_payload, + packet_input_time_ms * kSampRateHz / 1000); if (error != NetEq::kOK) return -1; } @@ -93,10 +95,10 @@ 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)); } @@ -107,7 +109,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, static const size_t kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels; int16_t out_data[kOutDataLen]; - int num_channels; + size_t num_channels; size_t samples_per_channel; int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel, &num_channels, NULL); |