Merging upstream version 127.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

2 files changed, 1 insertions, 64 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
index a5bf88e547..4deabdf7ff 100644
--- a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -50,11 +50,7 @@ std::unique_ptr<NetEq> CreateNetEq(
 
 AcmReceiver::Config::Config(
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
-    : clock(*Clock::GetRealTimeClockRaw()), decoder_factory(decoder_factory) {
-  // Post-decode VAD is disabled by default in NetEq, however, Audio
-  // Conference Mixer relies on VAD decisions and fails without them.
-  neteq_config.enable_post_decode_vad = true;
-}
+    : clock(*Clock::GetRealTimeClockRaw()), decoder_factory(decoder_factory) {}
 
 AcmReceiver::Config::Config(const Config&) = default;
 AcmReceiver::Config::~Config() = default;
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
index cda6688157..8b35f4a621 100644
--- a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -190,9 +190,6 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
     const size_t output_channels = info.num_channels;
     const size_t samples_per_ms = rtc::checked_cast<size_t>(
         rtc::CheckedDivExact(output_sample_rate_hz, 1000));
-    const AudioFrame::VADActivity expected_vad_activity =
-        output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
-                                      : AudioFrame::kVadPassive;
 
     // Expect the first output timestamp to be 5*fs/8000 samples before the
     // first inserted timestamp (because of NetEq's look-ahead). (This value is
@@ -217,7 +214,6 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
         EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
         EXPECT_EQ(output_channels, frame.num_channels_);
         EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
-        EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
         EXPECT_FALSE(muted);
       }
     }
@@ -243,61 +239,6 @@ TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
 }
 
 #if defined(WEBRTC_ANDROID)
-#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
-#else
-#define MAYBE_PostdecodingVad PostdecodingVad
-#endif
-TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
-  EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
-  constexpr int payload_type = 34;
-  const SdpAudioFormat codec = {"L16", 16000, 1};
-  const AudioCodecInfo info = SetEncoder(payload_type, codec);
-  receiver_->SetCodecs({{payload_type, codec}});
-  constexpr int kNumPackets = 5;
-  AudioFrame frame;
-  for (int n = 0; n < kNumPackets; ++n) {
-    const int num_10ms_frames = InsertOnePacketOfSilence(info);
-    for (int k = 0; k < num_10ms_frames; ++k) {
-      bool muted;
-      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
-    }
-  }
-  EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
-}
-
-class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
- protected:
-  AcmReceiverTestPostDecodeVadPassiveOldApi() {
-    config_.neteq_config.enable_post_decode_vad = false;
-  }
-};
-
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
-#else
-#define MAYBE_PostdecodingVad PostdecodingVad
-#endif
-TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
-  EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
-  constexpr int payload_type = 34;
-  const SdpAudioFormat codec = {"L16", 16000, 1};
-  const AudioCodecInfo info = SetEncoder(payload_type, codec);
-  auto const value = encoder_factory_->QueryAudioEncoder(codec);
-  ASSERT_TRUE(value.has_value());
-  receiver_->SetCodecs({{payload_type, codec}});
-  const int kNumPackets = 5;
-  AudioFrame frame;
-  for (int n = 0; n < kNumPackets; ++n) {
-    const int num_10ms_frames = InsertOnePacketOfSilence(info);
-    for (int k = 0; k < num_10ms_frames; ++k) {
-      bool muted;
-      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
-    }
-  }
-  EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
-}
-
-#if defined(WEBRTC_ANDROID)
 #define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
 #else
 #define MAYBE_LastAudioCodec LastAudioCodec