Enforcing return type handling on VoIP API.
- This CL also affects some return type handling in Android Voip demo
app due to changes in return type handling.
Bug: webrtc:12193
Change-Id: Id76faf7c871476ed1f2d08fb587211ae234ae8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196625
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32821}
diff --git a/api/voip/BUILD.gn b/api/voip/BUILD.gn
index c099bfb..4db59fd 100644
--- a/api/voip/BUILD.gn
+++ b/api/voip/BUILD.gn
@@ -21,6 +21,7 @@
]
deps = [
"..:array_view",
+ "../../rtc_base/system:unused",
"../audio_codecs:audio_codecs_api",
"../neteq:neteq_api",
]
diff --git a/api/voip/DEPS b/api/voip/DEPS
index 3845dff..837b9a6 100644
--- a/api/voip/DEPS
+++ b/api/voip/DEPS
@@ -3,6 +3,10 @@
"+third_party/absl/types/optional.h",
],
+ "voip_base.h": [
+ "+rtc_base/system/unused.h",
+ ],
+
"voip_engine_factory.h": [
"+modules/audio_device/include/audio_device.h",
"+modules/audio_processing/include/audio_processing.h",
diff --git a/api/voip/voip_base.h b/api/voip/voip_base.h
index c5f54aa..6a411f8 100644
--- a/api/voip/voip_base.h
+++ b/api/voip/voip_base.h
@@ -12,6 +12,7 @@
#define API_VOIP_VOIP_BASE_H_
#include "absl/types/optional.h"
+#include "rtc_base/system/unused.h"
namespace webrtc {
@@ -35,7 +36,7 @@
enum class ChannelId : int {};
-enum class VoipResult {
+enum class RTC_WARN_UNUSED_RESULT VoipResult {
// kOk indicates the function was successfully invoked with no error.
kOk,
// kInvalidArgument indicates the caller specified an invalid argument, such
diff --git a/audio/voip/test/voip_core_unittest.cc b/audio/voip/test/voip_core_unittest.cc
index f7a82f9..d290bd6 100644
--- a/audio/voip/test/voip_core_unittest.cc
+++ b/audio/voip/test/voip_core_unittest.cc
@@ -70,14 +70,18 @@
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat);
- voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}});
+ EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat),
+ VoipResult::kOk);
+ EXPECT_EQ(
+ voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}),
+ VoipResult::kOk);
EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk);
EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kOk);
- voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
- kPcmuSampleRateHz);
+ EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
+ kPcmuSampleRateHz),
+ VoipResult::kOk);
EXPECT_EQ(
voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
@@ -125,7 +129,8 @@
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat);
+ EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat),
+ VoipResult::kOk);
EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk);
// Send Dtmf event without registering beforehand, thus payload
@@ -145,8 +150,10 @@
TEST_F(VoipCoreTest, SendDtmfEventWithoutStartSend) {
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
- kPcmuSampleRateHz);
+ EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
+ kPcmuSampleRateHz),
+ VoipResult::kOk);
+
// Send Dtmf event without calling StartSend beforehand, thus
// Dtmf events cannot be sent and kFailedPrecondition is expected.
EXPECT_EQ(
diff --git a/examples/androidvoip/jni/android_voip_client.cc b/examples/androidvoip/jni/android_voip_client.cc
index d0763cd..95d3ed4 100644
--- a/examples/androidvoip/jni/android_voip_client.cc
+++ b/examples/androidvoip/jni/android_voip_client.cc
@@ -120,7 +120,7 @@
namespace webrtc_examples {
-bool AndroidVoipClient::Init(
+void AndroidVoipClient::Init(
JNIEnv* env,
const webrtc::JavaParamRef<jobject>& application_context) {
webrtc::VoipEngineConfig config;
@@ -132,20 +132,16 @@
config.audio_processing = webrtc::AudioProcessingBuilder().Create();
voip_thread_->Start();
+
// Due to consistent thread requirement on
// modules/audio_device/android/audio_device_template.h,
// code is invoked in the context of voip_thread_.
- return voip_thread_->Invoke<bool>(RTC_FROM_HERE, [this, &config] {
+ voip_thread_->Invoke<void>(RTC_FROM_HERE, [this, &config] {
RTC_DCHECK_RUN_ON(voip_thread_.get());
supported_codecs_ = config.encoder_factory->GetSupportedEncoders();
env_ = webrtc::AttachCurrentThreadIfNeeded();
voip_engine_ = webrtc::CreateVoipEngine(std::move(config));
- if (!voip_engine_) {
- RTC_LOG(LS_ERROR) << "VoipEngine creation failed";
- return false;
- }
- return true;
});
}
@@ -175,9 +171,7 @@
// Using `new` to access a non-public constructor.
auto voip_client =
absl::WrapUnique(new AndroidVoipClient(env, j_voip_client));
- if (!voip_client->Init(env, application_context)) {
- return nullptr;
- }
+ voip_client->Init(env, application_context);
return voip_client.release();
}
@@ -220,8 +214,9 @@
}
for (const webrtc::AudioCodecSpec& codec : supported_codecs_) {
if (codec.format.name == encoder) {
- voip_engine_->Codec().SetSendCodec(
+ webrtc::VoipResult result = voip_engine_->Codec().SetSendCodec(
*channel_, GetPayloadType(codec.format.name), codec.format);
+ RTC_CHECK(result == webrtc::VoipResult::kOk);
return;
}
}
@@ -251,7 +246,9 @@
}
}
- voip_engine_->Codec().SetReceiveCodecs(*channel_, decoder_specs);
+ webrtc::VoipResult result =
+ voip_engine_->Codec().SetReceiveCodecs(*channel_, decoder_specs);
+ RTC_CHECK(result == webrtc::VoipResult::kOk);
}
void AndroidVoipClient::SetDecoders(
@@ -305,13 +302,8 @@
void AndroidVoipClient::StartSession(JNIEnv* env) {
RUN_ON_VOIP_THREAD(StartSession, env);
+ // CreateChannel guarantees to return valid channel id.
channel_ = voip_engine_->Base().CreateChannel(this, absl::nullopt);
- if (!channel_) {
- RTC_LOG(LS_ERROR) << "Channel creation failed";
- Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_,
- /*isSuccessful=*/false);
- return;
- }
rtp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(),
rtp_local_address_));
@@ -357,7 +349,9 @@
rtp_socket_->Close();
rtcp_socket_->Close();
- voip_engine_->Base().ReleaseChannel(*channel_);
+ webrtc::VoipResult result = voip_engine_->Base().ReleaseChannel(*channel_);
+ RTC_CHECK(result == webrtc::VoipResult::kOk);
+
channel_ = absl::nullopt;
Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_,
/*isSuccessful=*/true);
@@ -470,9 +464,10 @@
RTC_LOG(LS_ERROR) << "Channel has not been created";
return;
}
- voip_engine_->Network().ReceivedRTPPacket(
+ webrtc::VoipResult result = voip_engine_->Network().ReceivedRTPPacket(
*channel_,
rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size()));
+ RTC_CHECK(result == webrtc::VoipResult::kOk);
}
void AndroidVoipClient::OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket,
@@ -495,9 +490,10 @@
RTC_LOG(LS_ERROR) << "Channel has not been created";
return;
}
- voip_engine_->Network().ReceivedRTCPPacket(
+ webrtc::VoipResult result = voip_engine_->Network().ReceivedRTCPPacket(
*channel_,
rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size()));
+ RTC_CHECK(result == webrtc::VoipResult::kOk);
}
void AndroidVoipClient::OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket,
diff --git a/examples/androidvoip/jni/android_voip_client.h b/examples/androidvoip/jni/android_voip_client.h
index 4dd0b0a..bfca7e8 100644
--- a/examples/androidvoip/jni/android_voip_client.h
+++ b/examples/androidvoip/jni/android_voip_client.h
@@ -141,7 +141,7 @@
: voip_thread_(rtc::Thread::CreateWithSocketServer()),
j_voip_client_(env, j_voip_client) {}
- bool Init(JNIEnv* env,
+ void Init(JNIEnv* env,
const webrtc::JavaParamRef<jobject>& application_context);
// Overloaded methods having native C++ variables as arguments.
