NetEq decoder database: Don't keep track of sample rate for builtin decoders

webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc

 /*
  *  Copyright (c) 2015 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 95 matching lines @@
   logic expand_rate;
   logic speech_expand_rate;
   logic preemptive_rate;
   logic accelerate_rate;
   logic secondary_decoded_rate;
   logic clockdrift_ppm;
   logic added_zero_samples;
   NetEqNetworkStatistics stats_ref;
 };
 
-  NetEqNetworkStatsTest(NetEqDecoder codec,
-                        MockAudioDecoder* decoder)
-      : NetEqExternalDecoderTest(codec, decoder),
-        external_decoder_(decoder),
-        samples_per_ms_(CodecSampleRateHz(codec) / 1000),
-        frame_size_samples_(kFrameSizeMs * samples_per_ms_),
-        rtp_generator_(new test::RtpGenerator(samples_per_ms_)),
-        last_lost_time_(0),
-        packet_loss_interval_(0xffffffff) {
-    Init();
+NetEqNetworkStatsTest(NetEqDecoder codec,
+                      int sample_rate_hz,
+                      MockAudioDecoder* decoder)
+    : NetEqExternalDecoderTest(codec, sample_rate_hz, decoder),
+      external_decoder_(decoder),
+      samples_per_ms_(sample_rate_hz / 1000),
+      frame_size_samples_(kFrameSizeMs * samples_per_ms_),
+      rtp_generator_(new test::RtpGenerator(samples_per_ms_)),
+      last_lost_time_(0),
+      packet_loss_interval_(0xffffffff) {
+  Init();
   }
 
   bool Lost(uint32_t send_time) {
     if (send_time - last_lost_time_ >= packet_loss_interval_) {
       last_lost_time_ = send_time;
       return true;
     }
     return false;
   }
 
@@ skipping 135 matching lines @@
   std::unique_ptr<test::RtpGenerator> rtp_generator_;
   WebRtcRTPHeader rtp_header_;
   uint32_t last_lost_time_;
   uint32_t packet_loss_interval_;
   uint8_t payload_[kPayloadSizeByte];
   AudioFrame output_frame_;
 };
 
 TEST(NetEqNetworkStatsTest, DecodeFec) {
   MockAudioDecoder decoder(1);
-  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
+  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, 48000, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, StereoDecodeFec) {
   MockAudioDecoder decoder(2);
-  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
+  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, 48000, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
   MockAudioDecoder decoder(1);
-  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
+  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, 48000, &decoder);
   test.NoiseExpansionTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 }  // namespace test
 }  // namespace webrtc