Remove the type parameter to NetEq::GetAudio

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 408 matching lines @@
                        rtc::ArrayView<const uint8_t>(
                            packet_->payload(), packet_->payload_length_bytes()),
                        static_cast<uint32_t>(packet_->time_ms() *
                                              (output_sample_rate_ / 1000))));
     }
     // Get next packet.
     packet_.reset(rtp_source_->NextPacket());
   }
 
   // Get audio from NetEq.
-  NetEqOutputType type;
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
   ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
               (out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
               (out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
               (out_frame_.samples_per_channel_ == kBlockSize48kHz));
   output_sample_rate_ = out_frame_.sample_rate_hz_;
   EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
 
   // Increase time.
   sim_clock_ += kTimeStepMs;
 }
@@ skipping 163 matching lines @@
     WebRtcRTPHeader rtp_info;
     rtp_info.header.sequenceNumber = i;
     rtp_info.header.timestamp = i * kSamples;
     rtp_info.header.ssrc = 0x1234;  // Just an arbitrary SSRC.
     rtp_info.header.payloadType = 94;  // PCM16b WB codec.
     rtp_info.header.markerBit = 0;
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   }
   // Pull out all data.
   for (size_t i = 0; i < num_frames; ++i) {
-    NetEqOutputType type;
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics stats;
   EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
   // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
   // spacing (per definition), we expect the delay to increase with 10 ms for
   // each packet. Thus, we are calculating the statistics for a series from 10
   // to 300, in steps of 10 ms.
   EXPECT_EQ(155, stats.mean_waiting_time_ms);
@@ skipping 20 matching lines @@
     int num_packets = (frame_index % 10 == 0 ? 2 : 1);
     for (int n = 0; n < num_packets; ++n) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
-    NetEqOutputType type;
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
   EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
 }
 
 TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
   const int kNumFrames = 5000;  // Needed for convergence.
   int frame_index = 0;
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   for (int i = 0; i < kNumFrames; ++i) {
     // Insert one packet each time, except every 10th time where we don't insert
     // any packet. This will create a positive clock-drift of approx. 11%.
     int num_packets = (i % 10 == 9 ? 0 : 1);
     for (int n = 0; n < num_packets; ++n) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
-    NetEqOutputType type;
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
   EXPECT_EQ(110946, network_stats.clockdrift_ppm);
 }
 
 void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
                                               double network_freeze_ms,
                                               bool pull_audio_during_freeze,
                                               int delay_tolerance_ms,
                                               int max_time_to_speech_ms) {
   uint16_t seq_no = 0;
   uint32_t timestamp = 0;
   const int kFrameSizeMs = 30;
   const size_t kSamples = kFrameSizeMs * 16;
   const size_t kPayloadBytes = kSamples * 2;
   double next_input_time_ms = 0.0;
   double t_ms;
-  NetEqOutputType type;
 
   // Insert speech for 5 seconds.
   const int kSpeechDurationMs = 5000;
   for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one 30 ms speech frame.
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
     }
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
-  EXPECT_EQ(kOutputNormal, type);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
   int32_t delay_before = timestamp - PlayoutTimestamp();
 
   // Insert CNG for 1 minute (= 60000 ms).
   const int kCngPeriodMs = 100;
   const int kCngPeriodSamples = kCngPeriodMs * 16;  // Period in 16 kHz samples.
   const int kCngDurationMs = 60000;
   for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one CNG frame each 100 ms.
       uint8_t payload[kPayloadBytes];
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
       ASSERT_EQ(0, neteq_->InsertPacket(
                        rtp_info,
                        rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
     }
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
-  EXPECT_EQ(kOutputCNG, type);
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
 
   if (network_freeze_ms > 0) {
     // First keep pulling audio for |network_freeze_ms| without inserting
     // any data, then insert CNG data corresponding to |network_freeze_ms|
     // without pulling any output audio.
     const double loop_end_time = t_ms + network_freeze_ms;
     for (; t_ms < loop_end_time; t_ms += 10) {
       // Pull out data once.
-      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
       ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-      EXPECT_EQ(kOutputCNG, type);
+      EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
     }
     bool pull_once = pull_audio_during_freeze;
     // If |pull_once| is true, GetAudio will be called once half-way through
     // the network recovery period.
     double pull_time_ms = (t_ms + next_input_time_ms) / 2;
     while (next_input_time_ms <= t_ms) {
       if (pull_once && next_input_time_ms >= pull_time_ms) {
         pull_once = false;
         // Pull out data once.
-        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
         ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-        EXPECT_EQ(kOutputCNG, type);
+        EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
         t_ms += 10;
       }
       // Insert one CNG frame each 100 ms.
       uint8_t payload[kPayloadBytes];
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
       ASSERT_EQ(0, neteq_->InsertPacket(
                        rtp_info,
                        rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += kCngPeriodMs * drift_factor;
     }
   }
 
   // Insert speech again until output type is speech.
   double speech_restart_time_ms = t_ms;
-  while (type != kOutputNormal) {
+  while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one 30 ms speech frame.
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += kFrameSizeMs * drift_factor;
     }
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
     // Increase clock.
     t_ms += 10;
   }
 
   // Check that the speech starts again within reasonable time.
   double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
   EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
   int32_t delay_after = timestamp - PlayoutTimestamp();
   // Compare delay before and after, and make sure it differs less than 20 ms.
@@ skipping 100 matching lines @@
 #define MAYBE_DecoderError DISABLED_DecoderError
 #endif
 
 TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 103;  // iSAC, but the payload is invalid.
   EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
-  NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
     out_frame_.data_[i] = 1;
   }
-  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &type));
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_));
   // Verify that there is a decoder error to check.
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
 
   enum NetEqDecoderError {
     ISAC_LENGTH_MISMATCH = 6730,
     ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH = 6640
   };
 #if defined(WEBRTC_CODEC_ISAC)
   EXPECT_EQ(ISAC_LENGTH_MISMATCH, neteq_->LastDecoderError());
 #elif defined(WEBRTC_CODEC_ISACFX)
@@ skipping 11 matching lines @@
   for (size_t i = kExpectedOutputLength; i < AudioFrame::kMaxDataSizeSamples;
        ++i) {
     std::ostringstream ss;
     ss << "i = " << i;
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(1, out_frame_.data_[i]);
   }
 }
 
 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
-  NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
     out_frame_.data_[i] = 1;
   }
-  EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+  EXPECT_EQ(0, neteq_->GetAudio(&out_frame_));
   // Verify that the first block of samples is set to 0.
   static const int kExpectedOutputLength =
       kInitSampleRateHz / 100;  // 10 ms at initial sample rate.
   for (int i = 0; i < kExpectedOutputLength; ++i) {
     std::ostringstream ss;
     ss << "i = " << i;
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(0, out_frame_.data_[i]);
   }
   // Verify that the sample rate did not change from the initial configuration.
@@ skipping 14 matching lines @@
     } else if (sampling_rate_hz == 16000) {
       expected_samples_per_channel = kBlockSize16kHz;
       payload_type = 94;  // PCM 16, 16 kHZ.
     } else if (sampling_rate_hz == 32000) {
       expected_samples_per_channel = kBlockSize32kHz;
       payload_type = 95;  // PCM 16, 32 kHz.
     } else {
       ASSERT_TRUE(false);  // Unsupported test case.
     }
 
-    NetEqOutputType type;
     AudioFrame output;
     test::AudioLoop input;
     // We are using the same 32 kHz input file for all tests, regardless of
     // |sampling_rate_hz|. The output may sound weird, but the test is still
     // valid.
     ASSERT_TRUE(input.Init(
         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
         10 * sampling_rate_hz,  // Max 10 seconds loop length.
         expected_samples_per_channel));
 
     // Payload of 10 ms of PCM16 32 kHz.
     uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
     WebRtcRTPHeader rtp_info;
     PopulateRtpInfo(0, 0, &rtp_info);
     rtp_info.header.payloadType = payload_type;
 
     uint32_t receive_timestamp = 0;
     for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
       auto block = input.GetNextBlock();
       ASSERT_EQ(expected_samples_per_channel, block.size());
       size_t enc_len_bytes =
           WebRtcPcm16b_Encode(block.data(), block.size(), payload);
       ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
 
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
                                                       payload, enc_len_bytes),
                                         receive_timestamp));
       output.Reset();
-      ASSERT_EQ(0, neteq_->GetAudio(&output, &type));
+      ASSERT_EQ(0, neteq_->GetAudio(&output));
       ASSERT_EQ(1u, output.num_channels_);
       ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
-      ASSERT_EQ(kOutputNormal, type);
+      ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
 
       // Next packet.
       rtp_info.header.timestamp += expected_samples_per_channel;
       rtp_info.header.sequenceNumber++;
       receive_timestamp += expected_samples_per_channel;
     }
 
     output.Reset();
 
     // Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull
     // one frame without checking speech-type. This is the first frame pulled
     // without inserting any packet, and might not be labeled as PLC.
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&output));
     ASSERT_EQ(1u, output.num_channels_);
     ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
 
     // To be able to test the fading of background noise we need at lease to
     // pull 611 frames.
     const int kFadingThreshold = 611;
 
     // Test several CNG-to-PLC packet for the expected behavior. The number 20
     // is arbitrary, but sufficiently large to test enough number of frames.
     const int kNumPlcToCngTestFrames = 20;
     bool plc_to_cng = false;
     for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
       output.Reset();
       memset(output.data_, 1, sizeof(output.data_));  // Set to non-zero.
-      ASSERT_EQ(0, neteq_->GetAudio(&output, &type));
+      ASSERT_EQ(0, neteq_->GetAudio(&output));
       ASSERT_EQ(1u, output.num_channels_);
       ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
-      if (type == kOutputPLCtoCNG) {
+      if (output.speech_type_ == AudioFrame::kPLCCNG) {
         plc_to_cng = true;
         double sum_squared = 0;
         for (size_t k = 0;
              k < output.num_channels_ * output.samples_per_channel_; ++k)
           sum_squared += output.data_[k] * output.data_[k];
         TestCondition(sum_squared, n > kFadingThreshold);
       } else {
-        EXPECT_EQ(kOutputPLC, type);
+        EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
       }
     }
     EXPECT_TRUE(plc_to_cng);  // Just to be sure that PLC-to-CNG has occurred.
   }
 };
 
 class NetEqBgnTestOn : public NetEqBgnTest {
  protected:
   NetEqBgnTestOn() : NetEqBgnTest() {
     config_.background_noise_mode = NetEq::kBgnOn;
@@ skipping 141 matching lines @@
   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.
-  NetEqOutputType output_type;
   uint32_t receive_timestamp = 0;
   for (int n = 0; n < 100; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
+    ASSERT_EQ(0, neteq_->GetAudio(&output));
     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, &output_type));
+    ASSERT_EQ(0, neteq_->GetAudio(&output));
     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, &output_type));
+    ASSERT_EQ(0, neteq_->GetAudio(&output));
     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;
@@ skipping 13 matching lines @@
   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.
-  NetEqOutputType output_type;
   uint32_t receive_timestamp = 0;
   int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
   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, &output_type));
+    ASSERT_EQ(0, neteq_->GetAudio(&output));
     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));
@@ skipping 16 matching lines @@
   // 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, &output_type));
+    ASSERT_EQ(0, neteq_->GetAudio(&output));
     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,
@@ skipping 46 matching lines @@
       ++seq_no;
       timestamp += kSamples;
       receive_timestamp += kSamples;
       next_input_time_ms += static_cast<double>(kFrameSizeMs);
 
       seq_no_wrapped |= seq_no < last_seq_no;
       timestamp_wrapped |= timestamp < last_timestamp;
     }
     // Pull out data once.
     AudioFrame output;
-    NetEqOutputType output_type;
-    ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
+    ASSERT_EQ(0, neteq_->GetAudio(&output));
     ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
     ASSERT_EQ(1u, output.num_channels_);
 
     // Expect delay (in samples) to be less than 2 packets.
     EXPECT_LE(timestamp - PlayoutTimestamp(),
               static_cast<uint32_t>(kSamples * 2));
   }
   // Make sure we have actually tested wrap-around.
   ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
   ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
@@ skipping 31 matching lines @@
   uint32_t timestamp = 0;
   const int kFrameSizeMs = 10;
   const int kSampleRateKhz = 16;
   const int kSamples = kFrameSizeMs * kSampleRateKhz;
   const size_t kPayloadBytes = kSamples * 2;
 
   const int algorithmic_delay_samples = std::max(
       algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
   // Insert three speech packets. Three are needed to get the frame length
   // correct.
-  NetEqOutputType type;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
     // Pull audio once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
-  EXPECT_EQ(kOutputNormal, type);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 
   // Insert same CNG packet twice.
   const int kCngPeriodMs = 100;
   const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
   size_t payload_len;
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   // This is the first time this CNG packet is inserted.
   ASSERT_EQ(
       0, neteq_->InsertPacket(
              rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio once and make sure CNG is played.
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-  EXPECT_EQ(kOutputCNG, type);
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
   EXPECT_EQ(timestamp - algorithmic_delay_samples, PlayoutTimestamp());
 
   // Insert the same CNG packet again. Note that at this point it is old, since
   // we have already decoded the first copy of it.
   ASSERT_EQ(
       0, neteq_->InsertPacket(
              rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
   // we have already pulled out CNG once.
   for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-    EXPECT_EQ(kOutputCNG, type);
+    EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
     EXPECT_EQ(timestamp - algorithmic_delay_samples,
               PlayoutTimestamp());
   }
 
   // Insert speech again.
   ++seq_no;
   timestamp += kCngPeriodSamples;
   PopulateRtpInfo(seq_no, timestamp, &rtp_info);
   ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
 
   // Pull audio once and verify that the output is speech again.
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-  EXPECT_EQ(kOutputNormal, type);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
   EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
             PlayoutTimestamp());
 }
 
 uint32_t NetEqDecodingTest::PlayoutTimestamp() {
   uint32_t playout_timestamp = 0;
   EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&playout_timestamp));
   return playout_timestamp;
 }
 
@@ skipping 15 matching lines @@
 
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   ASSERT_EQ(
       NetEq::kOK,
       neteq_->InsertPacket(
           rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
   ++seq_no;
   timestamp += kCngPeriodSamples;
 
   // Pull audio once and make sure CNG is played.
-  NetEqOutputType type;
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-  EXPECT_EQ(kOutputCNG, type);
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
 
   // Insert some speech packets.
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
     // Pull audio once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
-  EXPECT_EQ(kOutputNormal, type);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }
 
 }  // namespace webrtc