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/*
* 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