diff options
Diffstat (limited to 'webrtc/modules/audio_coding/main/test/target_delay_unittest.cc')
-rw-r--r-- | webrtc/modules/audio_coding/main/test/target_delay_unittest.cc | 223 |
1 files changed, 0 insertions, 223 deletions
diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc deleted file mode 100644 index 20b10a376e..0000000000 --- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc +++ /dev/null @@ -1,223 +0,0 @@ -/* - * Copyright (c) 2013 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 "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" -#include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/system_wrappers/include/sleep.h" -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/test/testsupport/gtest_disable.h" - -namespace webrtc { - -class TargetDelayTest : public ::testing::Test { - protected: - TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {} - - ~TargetDelayTest() {} - - void SetUp() { - EXPECT_TRUE(acm_.get() != NULL); - - CodecInst codec; - ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1)); - ASSERT_EQ(0, acm_->InitializeReceiver()); - ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec)); - - rtp_info_.header.payloadType = codec.pltype; - rtp_info_.header.timestamp = 0; - rtp_info_.header.ssrc = 0x12345678; - rtp_info_.header.markerBit = false; - rtp_info_.header.sequenceNumber = 0; - rtp_info_.type.Audio.channel = 1; - rtp_info_.type.Audio.isCNG = false; - rtp_info_.frameType = kAudioFrameSpeech; - - int16_t audio[kFrameSizeSamples]; - const int kRange = 0x7FF; // 2047, easy for masking. - for (size_t n = 0; n < kFrameSizeSamples; ++n) - audio[n] = (rand() & kRange) - kRange / 2; - WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_); - } - - void OutOfRangeInput() { - EXPECT_EQ(-1, SetMinimumDelay(-1)); - EXPECT_EQ(-1, SetMinimumDelay(10001)); - } - - void NoTargetDelayBufferSizeChanges() { - for (int n = 0; n < 30; ++n) // Run enough iterations. - Run(true); - int clean_optimal_delay = GetCurrentOptimalDelayMs(); - Run(false); // Run with jitter. - int jittery_optimal_delay = GetCurrentOptimalDelayMs(); - EXPECT_GT(jittery_optimal_delay, clean_optimal_delay); - int required_delay = RequiredDelay(); - EXPECT_GT(required_delay, 0); - EXPECT_NEAR(required_delay, jittery_optimal_delay, 1); - } - - void WithTargetDelayBufferNotChanging() { - // A target delay that is one packet larger than jitter. - const int kTargetDelayMs = (kInterarrivalJitterPacket + 1) * - kNum10msPerFrame * 10; - ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs)); - for (int n = 0; n < 30; ++n) // Run enough iterations to fill the buffer. - Run(true); - int clean_optimal_delay = GetCurrentOptimalDelayMs(); - EXPECT_EQ(kTargetDelayMs, clean_optimal_delay); - Run(false); // Run with jitter. - int jittery_optimal_delay = GetCurrentOptimalDelayMs(); - EXPECT_EQ(jittery_optimal_delay, clean_optimal_delay); - } - - void RequiredDelayAtCorrectRange() { - for (int n = 0; n < 30; ++n) // Run clean and store delay. - Run(true); - int clean_optimal_delay = GetCurrentOptimalDelayMs(); - - // A relatively large delay. - const int kTargetDelayMs = (kInterarrivalJitterPacket + 10) * - kNum10msPerFrame * 10; - ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs)); - for (int n = 0; n < 300; ++n) // Run enough iterations to fill the buffer. - Run(true); - Run(false); // Run with jitter. - - int jittery_optimal_delay = GetCurrentOptimalDelayMs(); - EXPECT_EQ(kTargetDelayMs, jittery_optimal_delay); - - int required_delay = RequiredDelay(); - - // Checking |required_delay| is in correct range. - EXPECT_GT(required_delay, 0); - EXPECT_GT(jittery_optimal_delay, required_delay); - EXPECT_GT(required_delay, clean_optimal_delay); - - // A tighter check for the value of |required_delay|. - // The jitter forces a delay of - // |kInterarrivalJitterPacket * kNum10msPerFrame * 10| milliseconds. So we - // expect |required_delay| be close to that. - EXPECT_NEAR(kInterarrivalJitterPacket * kNum10msPerFrame * 10, - required_delay, 1); - } - - void TargetDelayBufferMinMax() { - const int kTargetMinDelayMs = kNum10msPerFrame * 10; - ASSERT_EQ(0, SetMinimumDelay(kTargetMinDelayMs)); - for (int m = 0; m < 30; ++m) // Run enough iterations to fill the buffer. - Run(true); - int clean_optimal_delay = GetCurrentOptimalDelayMs(); - EXPECT_EQ(kTargetMinDelayMs, clean_optimal_delay); - - const int kTargetMaxDelayMs = 2 * (kNum10msPerFrame * 10); - ASSERT_EQ(0, SetMaximumDelay(kTargetMaxDelayMs)); - for (int n = 0; n < 30; ++n) // Run enough iterations to fill the buffer. - Run(false); - - int capped_optimal_delay = GetCurrentOptimalDelayMs(); - EXPECT_EQ(kTargetMaxDelayMs, capped_optimal_delay); - } - - private: - static const int kSampleRateHz = 16000; - static const int kNum10msPerFrame = 2; - static const size_t kFrameSizeSamples = 320; // 20 ms @ 16 kHz. - // payload-len = frame-samples * 2 bytes/sample. - static const int kPayloadLenBytes = 320 * 2; - // Inter-arrival time in number of packets in a jittery channel. One is no - // jitter. - static const int kInterarrivalJitterPacket = 2; - - void Push() { - rtp_info_.header.timestamp += kFrameSizeSamples; - rtp_info_.header.sequenceNumber++; - ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, - rtp_info_)); - } - - // Pull audio equivalent to the amount of audio in one RTP packet. - void Pull() { - AudioFrame frame; - for (int k = 0; k < kNum10msPerFrame; ++k) { // Pull one frame. - ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame)); - // Had to use ASSERT_TRUE, ASSERT_EQ generated error. - ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_); - ASSERT_EQ(1, frame.num_channels_); - ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_); - } - } - - void Run(bool clean) { - for (int n = 0; n < 10; ++n) { - for (int m = 0; m < 5; ++m) { - Push(); - Pull(); - } - - if (!clean) { - for (int m = 0; m < 10; ++m) { // Long enough to trigger delay change. - Push(); - for (int n = 0; n < kInterarrivalJitterPacket; ++n) - Pull(); - } - } - } - } - - int SetMinimumDelay(int delay_ms) { - return acm_->SetMinimumPlayoutDelay(delay_ms); - } - - int SetMaximumDelay(int delay_ms) { - return acm_->SetMaximumPlayoutDelay(delay_ms); - } - - int GetCurrentOptimalDelayMs() { - NetworkStatistics stats; - acm_->GetNetworkStatistics(&stats); - return stats.preferredBufferSize; - } - - int RequiredDelay() { - return acm_->LeastRequiredDelayMs(); - } - - rtc::scoped_ptr<AudioCodingModule> acm_; - WebRtcRTPHeader rtp_info_; - uint8_t payload_[kPayloadLenBytes]; -}; - -TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) { - OutOfRangeInput(); -} - -TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) { - NoTargetDelayBufferSizeChanges(); -} - -TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) { - WithTargetDelayBufferNotChanging(); -} - -TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) { - RequiredDelayAtCorrectRange(); -} - -TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) { - TargetDelayBufferMinMax(); -} - -} // namespace webrtc - |