Propagate muted parameter to VoE::Channel

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

 /*
  *  Copyright (c) 2014 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 187 matching lines @@
 
   virtual void InsertPacket() {
     const uint8_t kPayload[kPayloadSizeBytes] = {0};
     ASSERT_EQ(0,
               acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
     rtp_utility_->Forward(&rtp_header_);
   }
 
   virtual void PullAudio() {
     AudioFrame audio_frame;
-    ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame));
+    bool muted;
+    ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
   }
 
   virtual void InsertAudio() {
     ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
     input_frame_.timestamp_ += kNumSamples10ms;
   }
 
   virtual void VerifyEncoding() {
     int last_length = packet_cb_.last_payload_len_bytes();
     EXPECT_TRUE(last_length == 2 * codec_.pacsize || last_length == 0)
@@ skipping 70 matching lines @@
   EXPECT_EQ(0, stats.calls_to_silence_generator);
   EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
   EXPECT_EQ(0, stats.decoded_cng);
   EXPECT_EQ(kNumPlc, stats.decoded_plc);
   EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
 }
 
 TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   AudioFrame audio_frame;
   const int kSampleRateHz = 32000;
-  EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
+  bool muted;
+  EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
+  ASSERT_FALSE(muted);
   EXPECT_EQ(id_, audio_frame.id_);
   EXPECT_EQ(0u, audio_frame.timestamp_);
   EXPECT_GT(audio_frame.num_channels_, 0u);
   EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
             audio_frame.samples_per_channel_);
   EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
 }
 
 TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
   AudioFrame audio_frame;
-  EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame));
+  bool muted;
+  EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame, &muted));
 }
 
 // Checks that the transport callback is invoked once for each speech packet.
 // Also checks that the frame type is kAudioFrameSpeech.
 TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
   const int k10MsBlocksPerPacket = 3;
   codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
   RegisterCodec();
   const int kLoops = 10;
   for (int i = 0; i < kLoops; ++i) {
@@ skipping 478 matching lines @@
     }
     // Now we're not holding the crit sect when calling ACM.
 
     // Insert into ACM.
     EXPECT_EQ(0, acm_->IncomingPacket(encoded.data(), info.encoded_bytes,
                                       rtp_header_));
 
     // Pull audio.
     for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) {
       AudioFrame audio_frame;
+      bool muted;
       EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */,
-                                         &audio_frame));
+                                         &audio_frame, &muted));
+      if (muted) {
+        ADD_FAILURE();
+        return false;
+      }
       fake_clock_->AdvanceTimeMilliseconds(10);
     }
     rtp_utility_->Forward(&rtp_header_);
     return true;
   }
 
   static bool CbCodecRegistrationThread(void* context) {
     return reinterpret_cast<AcmReRegisterIsacMtTestOldApi*>(context)
         ->CbCodecRegistrationImpl();
   }
@@ skipping 958 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc