| /* |
| * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h" |
| |
| #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.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" |
| #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| #include "webrtc/typedefs.h" |
| |
| using webrtc::NetEq; |
| using webrtc::test::AudioLoop; |
| using webrtc::test::RtpGenerator; |
| |
| namespace webrtc { |
| namespace test { |
| |
| int64_t NetEqPerformanceTest::Run(int runtime_ms, |
| int lossrate, |
| double drift_factor) { |
| const std::string kInputFileName = |
| webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
| const int kSampRateHz = 32000; |
| const webrtc::NetEqDecoder kDecoderType = |
| webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz; |
| const std::string kDecoderName = "pcm16-swb32"; |
| const int kPayloadType = 95; |
| |
| // Initialize NetEq instance. |
| NetEq::Config config; |
| config.sample_rate_hz = kSampRateHz; |
| NetEq* neteq = NetEq::Create(config, CreateBuiltinAudioDecoderFactory()); |
| // Register decoder in |neteq|. |
| if (neteq->RegisterPayloadType(kDecoderType, kDecoderName, kPayloadType) != 0) |
| return -1; |
| |
| // Set up AudioLoop object. |
| AudioLoop audio_loop; |
| const size_t kMaxLoopLengthSamples = kSampRateHz * 10; // 10 second loop. |
| const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000; // 60 ms. |
| if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, |
| kInputBlockSizeSamples)) |
| return -1; |
| |
| int32_t time_now_ms = 0; |
| |
| // Get first input packet. |
| RTPHeader rtp_header; |
| RtpGenerator rtp_gen(kSampRateHz / 1000); |
| // Start with positive drift first half of simulation. |
| rtp_gen.set_drift_factor(drift_factor); |
| bool drift_flipped = false; |
| int32_t packet_input_time_ms = |
| rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); |
| 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.data(), |
| input_samples.size(), input_payload); |
| RTC_CHECK_EQ(sizeof(input_payload), payload_len); |
| |
| // Main loop. |
| webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock(); |
| int64_t start_time_ms = clock->TimeInMilliseconds(); |
| AudioFrame out_frame; |
| while (time_now_ms < runtime_ms) { |
| while (packet_input_time_ms <= time_now_ms) { |
| // Drop every N packets, where N = FLAG_lossrate. |
| bool lost = false; |
| if (lossrate > 0) { |
| lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0; |
| } |
| if (!lost) { |
| // Insert packet. |
| int error = |
| neteq->InsertPacket(rtp_header, input_payload, |
| packet_input_time_ms * kSampRateHz / 1000); |
| if (error != NetEq::kOK) |
| return -1; |
| } |
| |
| // Get next packet. |
| packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType, |
| kInputBlockSizeSamples, |
| &rtp_header); |
| input_samples = audio_loop.GetNextBlock(); |
| 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)); |
| } |
| |
| // Get output audio, but don't do anything with it. |
| bool muted; |
| int error = neteq->GetAudio(&out_frame, &muted); |
| RTC_CHECK(!muted); |
| if (error != NetEq::kOK) |
| return -1; |
| |
| 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. |
| rtp_gen.set_drift_factor(-drift_factor); |
| drift_flipped = true; |
| } |
| } |
| int64_t end_time_ms = clock->TimeInMilliseconds(); |
| delete neteq; |
| return end_time_ms - start_time_ms; |
| } |
| |
| } // namespace test |
| } // namespace webrtc |