Index: webrtc/modules/audio_coding/main/test/target_delay_unittest.cc |
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 afc0e102255dd64f2729563de6640c74f4e1635b..0000000000000000000000000000000000000000 |
--- 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/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/include/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 |
- |