commit 84f75699c65384a961cdf9ebffefd6bc83e0d8e2
author Henrik Lundin <henrik.lundin@webrtc.org> Wed Feb 01 12:07:10 2023 +0000
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Feb 01 16:09:26 2023 +0000
tree ea5761e014beca19fe25b7e13842bcd2baead6a4
parent c5455e7b5391ec4b567f7dd0bdb88bdc02fdf07c

Break apart AudioCodingModule and AcmReceiver

This change makes AudioCodingModule a pure sender and AcmReceiver a pure
receiver.

The Config struct is in practice no longer used by AudioCodingModule,
so a new definition is included in AcmReceiver. The old definition
remains in AudioCodingModule while downstream clients are being
updated.

Bug: webrtc:14867
Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39244}

modules/audio_coding/acm2/acm_receiver_unittest.cc

diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 6dd44b6..a5095f0 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -44,11 +44,10 @@
   ~AcmReceiverTestOldApi() {}
 
   void SetUp() override {
-    acm_.reset(AudioCodingModule::Create(config_));
+    acm_ = AudioCodingModule::Create();
     receiver_.reset(new AcmReceiver(config_));
     ASSERT_TRUE(receiver_.get() != NULL);
     ASSERT_TRUE(acm_.get() != NULL);
-    acm_->InitializeReceiver();
     acm_->RegisterTransportCallback(this);
 
     rtp_header_.sequenceNumber = 0;
@@ -135,7 +134,7 @@
       CreateBuiltinAudioEncoderFactory();
   const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_ =
       CreateBuiltinAudioDecoderFactory();
-  AudioCodingModule::Config config_;
+  acm2::AcmReceiver::Config config_;
   std::unique_ptr<AcmReceiver> receiver_;
   std::unique_ptr<AudioCodingModule> acm_;
   RTPHeader rtp_header_;
@@ -383,6 +382,24 @@
   EXPECT_EQ(0, stats.decoded_muted_output);
 }
 
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyOutputFrame DISABLED_VerifyOutputFrame
+#else
+#define MAYBE_VerifyOutputFrame VerifyOutputFrame
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_VerifyOutputFrame) {
+  AudioFrame audio_frame;
+  const int kSampleRateHz = 32000;
+  bool muted;
+  EXPECT_EQ(0, receiver_->GetAudio(kSampleRateHz, &audio_frame, &muted));
+  ASSERT_FALSE(muted);
+  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_);
+}
+
 // Insert some packets and pull audio. Check statistics are valid. Then,
 // simulate packet loss and check if PLC and PLC-to-CNG statistics are
 // correctly updated.