Ensure the AudioCodingModule is reset when sending is stopped.

Bug: webrtc:42226041
Change-Id: Ife3548bda3042a7447b7c50f48f023a2bc0bc443
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362103
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43017}

audio/channel_send.cc

@@ -562,7 +562,7 @@ void ChannelSend::StopSend() {
   rtc::Event flush;
   encoder_queue_->PostTask([this, &flush]() {
     RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
-    CallEncoder([](AudioEncoder* encoder) { encoder->Reset(); });
+    audio_coding_->Reset();
     flush.Set();
   });
   flush.Wait(rtc::Event::kForever);

audio/channel_send_unittest.cc

@@ -140,6 +140,7 @@ TEST_F(ChannelSendTest, StopSendShouldResetEncoder) {
   ProcessNextFrame();
   // StopSend should clear the previous audio frame stored in the encoder.
   channel_->StopSend();
+
   channel_->StartSend();
   // The following frame should not trigger a new packet since the encoder
   // needs 20 ms audio.

modules/audio_coding/acm2/audio_coding_module.cc

@@ -43,6 +43,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   explicit AudioCodingModuleImpl();
   ~AudioCodingModuleImpl() override;
 
+  void Reset() override;
+
   /////////////////////////////////////////
   //   Sender
   //
@@ -280,7 +282,7 @@ int32_t AudioCodingModuleImpl::Encode(
           absolute_capture_timestamp_ms_.value_or(-1));
     }
   }
-  absolute_capture_timestamp_ms_ = std::nullopt;
+  absolute_capture_timestamp_ms_.reset();
   previous_pltype_ = encoded_info.payload_type;
   return static_cast<int32_t>(encode_buffer_.size());
 }
@@ -289,6 +291,14 @@ int32_t AudioCodingModuleImpl::Encode(
 //   Sender
 //
 
+void AudioCodingModuleImpl::Reset() {
+  MutexLock lock(&acm_mutex_);
+  absolute_capture_timestamp_ms_.reset();
+  if (HaveValidEncoder("Reset")) {
+    encoder_stack_->Reset();
+  }
+}
+
 void AudioCodingModuleImpl::ModifyEncoder(
     rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
   MutexLock lock(&acm_mutex_);

modules/audio_coding/acm2/audio_coding_module_unittest.cc

@@ -522,6 +522,17 @@ class AudioPacketizationCallbackMock : public AudioPacketizationCallback {
               (override));
 };
 
+TEST(AudioCodingModule, DoesResetEncoder) {
+  std::unique_ptr<AudioCodingModule> acm = AudioCodingModule::Create();
+  auto encoder = std::make_unique<MockAudioEncoder>();
+  MockAudioEncoder* encoder_mock = encoder.get();
+
+  acm->SetEncoder(std::move(encoder));
+
+  EXPECT_CALL(*encoder_mock, Reset()).Times(1);
+  acm->Reset();
+}
+
 class AcmAbsoluteCaptureTimestamp : public ::testing::Test {
  public:
   AcmAbsoluteCaptureTimestamp() : audio_frame_(kSampleRateHz, kNumChannels) {}
@@ -587,6 +598,40 @@ TEST_F(AcmAbsoluteCaptureTimestamp, HaveBeginningOfFrameCaptureTime) {
   }
 }
 
+TEST_F(AcmAbsoluteCaptureTimestamp, DoesResetWhenAudioCodingModuleDo) {
+  constexpr int64_t first_absolute_capture_timestamp_ms = 123456789;
+
+  int64_t absolute_capture_timestamp_ms = first_absolute_capture_timestamp_ms;
+  EXPECT_CALL(transport_,
+              SendData(_, _, _, _, _, first_absolute_capture_timestamp_ms))
+      .Times(1);
+  EXPECT_CALL(
+      transport_,
+      SendData(_, _, _, _, _, first_absolute_capture_timestamp_ms + kPTimeMs))
+      .Times(1);
+  for (int k = 0; k < 5; ++k) {
+    acm_->Add10MsData(
+        GetAudioWithAbsoluteCaptureTimestamp(absolute_capture_timestamp_ms));
+    absolute_capture_timestamp_ms += 10;
+  }
+
+  acm_->Reset();
+  constexpr int64_t after_reset_absolute_capture_timestamp_ms = 523456789;
+  EXPECT_CALL(transport_, SendData(_, _, _, _, _,
+                                   after_reset_absolute_capture_timestamp_ms))
+      .Times(1);
+  EXPECT_CALL(transport_,
+              SendData(_, _, _, _, _,
+                       after_reset_absolute_capture_timestamp_ms + kPTimeMs))
+      .Times(1);
+  absolute_capture_timestamp_ms = after_reset_absolute_capture_timestamp_ms;
+  for (int k = 0; k < 5; ++k) {
+    acm_->Add10MsData(
+        GetAudioWithAbsoluteCaptureTimestamp(absolute_capture_timestamp_ms));
+    absolute_capture_timestamp_ms += 10;
+  }
+}
+
 // Disabling all of these tests on iOS until file support has been added.
 // See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
 #if !defined(WEBRTC_IOS)

modules/audio_coding/include/audio_coding_module.h

@@ -78,6 +78,10 @@ class AudioCodingModule {
     });
   }
 
+  // Reset encoder and audio coding module. This throws away any audio passed
+  // and starts fresh.
+  virtual void Reset() = 0;
+
   // int32_t RegisterTransportCallback()
   // Register a transport callback which will be called to deliver
   // the encoded buffers whenever Process() is called and a