Removed sync packet support from NetEq.

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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.
  */
 
@@ skipping 43 matching lines @@
   #ifdef WEBRTC_ARCH_64_BITS
     return checksum_win_64;
   #else
     return checksum_win_32;
   #endif  // WEBRTC_ARCH_64_BITS
 #else
   return checksum_general;
 #endif  // WEBRTC_WIN
 }
 
-bool IsAllZero(const int16_t* buf, size_t buf_length) {
-  bool all_zero = true;
-  for (size_t n = 0; n < buf_length && all_zero; ++n)
-    all_zero = buf[n] == 0;
-  return all_zero;
-}
-
-bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
-  bool all_non_zero = true;
-  for (size_t n = 0; n < buf_length && all_non_zero; ++n)
-    all_non_zero = buf[n] != 0;
-  return all_non_zero;
-}
-
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
 void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
              webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
   stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
   stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
   stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
   stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
   stats->set_packet_discard_rate(stats_raw.packet_discard_rate);
   stats->set_expand_rate(stats_raw.expand_rate);
   stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
@@ skipping 984 matching lines @@
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
 TEST_F(NetEqBgnTestFade, RunTest) {
   CheckBgn(8000);
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define MAYBE_SyncPacketInsert SyncPacketInsert
-#else
-#define MAYBE_SyncPacketInsert DISABLED_SyncPacketInsert
-#endif
-TEST_F(NetEqDecodingTest, MAYBE_SyncPacketInsert) {
-  WebRtcRTPHeader rtp_info;
-  uint32_t receive_timestamp = 0;
-  // For the readability use the following payloads instead of the defaults of
-  // this test.
-  uint8_t kPcm16WbPayloadType = 1;
-  uint8_t kCngNbPayloadType = 2;
-  uint8_t kCngWbPayloadType = 3;
-  uint8_t kCngSwb32PayloadType = 4;
-  uint8_t kCngSwb48PayloadType = 5;
-  uint8_t kAvtPayloadType = 6;
-  uint8_t kRedPayloadType = 7;
-  uint8_t kIsacPayloadType = 9;  // Payload type 8 is already registered.
-
-  // Register decoders.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb,
-                                           "pcm16-wb", kPcm16WbPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
-                                           "cng-nb", kCngNbPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
-                                           "cng-wb", kCngWbPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb32kHz,
-                                           "cng-swb32", kCngSwb32PayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb48kHz,
-                                           "cng-swb48", kCngSwb48PayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderAVT, "avt",
-                                           kAvtPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderRED, "red",
-                                           kRedPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac",
-                                           kIsacPayloadType));
-
-  PopulateRtpInfo(0, 0, &rtp_info);
-  rtp_info.header.payloadType = kPcm16WbPayloadType;
-
-  // The first packet injected cannot be sync-packet.
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  // Payload length of 10 ms PCM16 16 kHz.
-  const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
-  uint8_t payload[kPayloadBytes] = {0};
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-
-  // Next packet. Last packet contained 10 ms audio.
-  rtp_info.header.sequenceNumber++;
-  rtp_info.header.timestamp += kBlockSize16kHz;
-  receive_timestamp += kBlockSize16kHz;
-
-  // Unacceptable payload types CNG, AVT (DTMF), RED.
-  rtp_info.header.payloadType = kCngNbPayloadType;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  rtp_info.header.payloadType = kCngWbPayloadType;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  rtp_info.header.payloadType = kCngSwb32PayloadType;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  rtp_info.header.payloadType = kCngSwb48PayloadType;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  rtp_info.header.payloadType = kAvtPayloadType;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  rtp_info.header.payloadType = kRedPayloadType;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  // Change of codec cannot be initiated with a sync packet.
-  rtp_info.header.payloadType = kIsacPayloadType;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  // Change of SSRC is not allowed with a sync packet.
-  rtp_info.header.payloadType = kPcm16WbPayloadType;
-  ++rtp_info.header.ssrc;
-  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-
-  --rtp_info.header.ssrc;
-  EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-}
-
-// First insert several noise like packets, then sync-packets. Decoding all
-// packets should not produce error, statistics should not show any packet loss
-// and sync-packets should decode to zero.
-// TODO(turajs) we will have a better test if we have a referece NetEq, and
-// when Sync packets are inserted in "test" NetEq we insert all-zero payload
-// in reference NetEq and compare the output of those two.
-TEST_F(NetEqDecodingTest, SyncPacketDecode) {
-  WebRtcRTPHeader rtp_info;
-  PopulateRtpInfo(0, 0, &rtp_info);
-  const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
-  uint8_t payload[kPayloadBytes];
-  AudioFrame output;
-  int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
-  for (size_t n = 0; n < kPayloadBytes; ++n) {
-    payload[n] = (rand() & 0xF0) + 1;  // Non-zero random sequence.
-  }
-  // Insert some packets which decode to noise. We are not interested in
-  // actual decoded values.
-  uint32_t receive_timestamp = 0;
-  bool muted;
-  for (int n = 0; n < 100; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
-    ASSERT_EQ(1u, output.num_channels_);
-
-    rtp_info.header.sequenceNumber++;
-    rtp_info.header.timestamp += kBlockSize16kHz;
-    receive_timestamp += kBlockSize16kHz;
-  }
-  const int kNumSyncPackets = 10;
-
-  // Make sure sufficient number of sync packets are inserted that we can
-  // conduct a test.
-  ASSERT_GT(kNumSyncPackets, algorithmic_frame_delay);
-  // Insert sync-packets, the decoded sequence should be all-zero.
-  for (int n = 0; n < kNumSyncPackets; ++n) {
-    ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-    ASSERT_FALSE(muted);
-    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
-    ASSERT_EQ(1u, output.num_channels_);
-    if (n > algorithmic_frame_delay) {
-      EXPECT_TRUE(IsAllZero(
-          output.data_, output.samples_per_channel_ * output.num_channels_));
-    }
-    rtp_info.header.sequenceNumber++;
-    rtp_info.header.timestamp += kBlockSize16kHz;
-    receive_timestamp += kBlockSize16kHz;
-  }
-
-  // We insert regular packets, if sync packet are not correctly buffered then
-  // network statistics would show some packet loss.
-  for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-    ASSERT_FALSE(muted);
-    if (n >= algorithmic_frame_delay + 1) {
-      // Expect that this frame contain samples from regular RTP.
-      EXPECT_TRUE(IsAllNonZero(
-          output.data_, output.samples_per_channel_ * output.num_channels_));
-    }
-    rtp_info.header.sequenceNumber++;
-    rtp_info.header.timestamp += kBlockSize16kHz;
-    receive_timestamp += kBlockSize16kHz;
-  }
-  NetEqNetworkStatistics network_stats;
-  ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
-  // Expecting a "clean" network.
-  EXPECT_EQ(0, network_stats.packet_loss_rate);
-  EXPECT_EQ(0, network_stats.expand_rate);
-  EXPECT_EQ(0, network_stats.accelerate_rate);
-  EXPECT_LE(network_stats.preemptive_rate, 150);
-}
-
-// Test if the size of the packet buffer reported correctly when containing
-// sync packets. Also, test if network packets override sync packets. That is to
-// prefer decoding a network packet to a sync packet, if both have same sequence
-// number and timestamp.
-TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
-  WebRtcRTPHeader rtp_info;
-  PopulateRtpInfo(0, 0, &rtp_info);
-  const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
-  uint8_t payload[kPayloadBytes];
-  AudioFrame output;
-  for (size_t n = 0; n < kPayloadBytes; ++n) {
-    payload[n] = (rand() & 0xF0) + 1;  // Non-zero random sequence.
-  }
-  // Insert some packets which decode to noise. We are not interested in
-  // actual decoded values.
-  uint32_t receive_timestamp = 0;
-  int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
-  bool muted;
-  for (int n = 0; n < algorithmic_frame_delay; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
-    ASSERT_EQ(1u, output.num_channels_);
-    rtp_info.header.sequenceNumber++;
-    rtp_info.header.timestamp += kBlockSize16kHz;
-    receive_timestamp += kBlockSize16kHz;
-  }
-  const int kNumSyncPackets = 10;
-
-  WebRtcRTPHeader first_sync_packet_rtp_info;
-  memcpy(&first_sync_packet_rtp_info, &rtp_info, sizeof(rtp_info));
-
-  // Insert sync-packets, but no decoding.
-  for (int n = 0; n < kNumSyncPackets; ++n) {
-    ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-    rtp_info.header.sequenceNumber++;
-    rtp_info.header.timestamp += kBlockSize16kHz;
-    receive_timestamp += kBlockSize16kHz;
-  }
-  NetEqNetworkStatistics network_stats;
-  ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
-  EXPECT_EQ(kNumSyncPackets * 10 + algorithmic_delay_ms_,
-            network_stats.current_buffer_size_ms);
-
-  // Rewind |rtp_info| to that of the first sync packet.
-  memcpy(&rtp_info, &first_sync_packet_rtp_info, sizeof(rtp_info));
-
-  // Insert.
-  for (int n = 0; n < kNumSyncPackets; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    rtp_info.header.sequenceNumber++;
-    rtp_info.header.timestamp += kBlockSize16kHz;
-    receive_timestamp += kBlockSize16kHz;
-  }
-
-  // Decode.
-  for (int n = 0; n < kNumSyncPackets; ++n) {
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-    ASSERT_FALSE(muted);
-    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
-    ASSERT_EQ(1u, output.num_channels_);
-    EXPECT_TRUE(IsAllNonZero(
-        output.data_, output.samples_per_channel_ * output.num_channels_));
-  }
-}
-
 void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
                                  uint32_t start_timestamp,
                                  const std::set<uint16_t>& drop_seq_numbers,
                                  bool expect_seq_no_wrap,
                                  bool expect_timestamp_wrap) {
   uint16_t seq_no = start_seq_no;
   uint32_t timestamp = start_timestamp;
   const int kBlocksPerFrame = 3;  // Number of 10 ms blocks per frame.
   const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
   const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
@@ skipping 464 matching lines @@
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc