audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/target_delay_unittest.cc

-/*
- *  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
-