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d00ce747c7
Bug: webrtc:12338 Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34621}
128 lines
4.6 KiB
C++
128 lines
4.6 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
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#include "api/audio/audio_frame.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/neteq/neteq.h"
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#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
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#include "modules/audio_coding/neteq/default_neteq_factory.h"
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#include "modules/audio_coding/neteq/tools/audio_loop.h"
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#include "modules/audio_coding/neteq/tools/rtp_generator.h"
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/clock.h"
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#include "test/testsupport/file_utils.h"
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using webrtc::NetEq;
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using webrtc::test::AudioLoop;
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using webrtc::test::RtpGenerator;
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namespace webrtc {
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namespace test {
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int64_t NetEqPerformanceTest::Run(int runtime_ms,
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int lossrate,
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double drift_factor) {
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const std::string kInputFileName =
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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const int kSampRateHz = 32000;
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const std::string kDecoderName = "pcm16-swb32";
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const int kPayloadType = 95;
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// Initialize NetEq instance.
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NetEq::Config config;
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config.sample_rate_hz = kSampRateHz;
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webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
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auto audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
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auto neteq =
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DefaultNetEqFactory().CreateNetEq(config, audio_decoder_factory, clock);
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// Register decoder in `neteq`.
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if (!neteq->RegisterPayloadType(kPayloadType,
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SdpAudioFormat("l16", kSampRateHz, 1)))
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return -1;
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// Set up AudioLoop object.
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AudioLoop audio_loop;
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const size_t kMaxLoopLengthSamples = kSampRateHz * 10; // 10 second loop.
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const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000; // 60 ms.
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if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
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kInputBlockSizeSamples))
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return -1;
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int32_t time_now_ms = 0;
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// Get first input packet.
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RTPHeader rtp_header;
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RtpGenerator rtp_gen(kSampRateHz / 1000);
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// Start with positive drift first half of simulation.
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rtp_gen.set_drift_factor(drift_factor);
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bool drift_flipped = false;
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int32_t packet_input_time_ms =
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rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
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auto input_samples = audio_loop.GetNextBlock();
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if (input_samples.empty())
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exit(1);
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uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
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size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
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input_samples.size(), input_payload);
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RTC_CHECK_EQ(sizeof(input_payload), payload_len);
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// Main loop.
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int64_t start_time_ms = clock->TimeInMilliseconds();
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AudioFrame out_frame;
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while (time_now_ms < runtime_ms) {
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while (packet_input_time_ms <= time_now_ms) {
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// Drop every N packets, where N = FLAG_lossrate.
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bool lost = false;
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if (lossrate > 0) {
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lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;
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}
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if (!lost) {
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// Insert packet.
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int error = neteq->InsertPacket(rtp_header, input_payload);
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if (error != NetEq::kOK)
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return -1;
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}
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// Get next packet.
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packet_input_time_ms = rtp_gen.GetRtpHeader(
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kPayloadType, kInputBlockSizeSamples, &rtp_header);
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input_samples = audio_loop.GetNextBlock();
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if (input_samples.empty())
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return -1;
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payload_len = WebRtcPcm16b_Encode(input_samples.data(),
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input_samples.size(), input_payload);
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RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));
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}
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// Get output audio, but don't do anything with it.
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bool muted;
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int error = neteq->GetAudio(&out_frame, &muted);
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RTC_CHECK(!muted);
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if (error != NetEq::kOK)
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return -1;
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RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);
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static const int kOutputBlockSizeMs = 10;
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time_now_ms += kOutputBlockSizeMs;
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if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
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// Apply negative drift second half of simulation.
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rtp_gen.set_drift_factor(-drift_factor);
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drift_flipped = true;
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}
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}
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int64_t end_time_ms = clock->TimeInMilliseconds();
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return end_time_ms - start_time_ms;
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}
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} // namespace test
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} // namespace webrtc
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