Defining API result types on VoIP API
Bug: webrtc:12193
Change-Id: I6f5ffd82cc838e6982257781f225f9d8159e6b82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193720
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32656}
diff --git a/api/voip/voip_base.h b/api/voip/voip_base.h
index ef83b51..c5f54aa 100644
--- a/api/voip/voip_base.h
+++ b/api/voip/voip_base.h
@@ -35,6 +35,21 @@
enum class ChannelId : int {};
+enum class VoipResult {
+ // kOk indicates the function was successfully invoked with no error.
+ kOk,
+ // kInvalidArgument indicates the caller specified an invalid argument, such
+ // as an invalid ChannelId.
+ kInvalidArgument,
+ // kFailedPrecondition indicates that the operation was failed due to not
+ // satisfying prerequisite such as not setting codec type before sending.
+ kFailedPrecondition,
+ // kInternal is used to indicate various internal failures that are not the
+ // caller's fault. Further detail is commented on each function that uses this
+ // return value.
+ kInternal,
+};
+
class VoipBase {
public:
// Creates a channel.
@@ -46,40 +61,48 @@
// and injection for incoming RTP from remote endpoint is handled via
// VoipNetwork interface. |local_ssrc| is optional and when local_ssrc is not
// set, some random value will be used by voip engine.
- // Returns value is optional as to indicate the failure to create channel.
- virtual absl::optional<ChannelId> CreateChannel(
- Transport* transport,
- absl::optional<uint32_t> local_ssrc) = 0;
+ // Returns a ChannelId created for caller to handle subsequent Channel
+ // operations.
+ virtual ChannelId CreateChannel(Transport* transport,
+ absl::optional<uint32_t> local_ssrc) = 0;
// Releases |channel_id| that no longer has any use.
- virtual void ReleaseChannel(ChannelId channel_id) = 0;
+ // Returns following VoipResult;
+ // kOk - |channel_id| is released.
+ // kInvalidArgument - |channel_id| is invalid.
+ // kInternal - Fails to stop audio output device.
+ virtual VoipResult ReleaseChannel(ChannelId channel_id) = 0;
- // Starts sending on |channel_id|. This will start microphone if not started
- // yet. Returns false if initialization has failed on selected microphone
- // device. API is subject to expand to reflect error condition to application
- // later.
- virtual bool StartSend(ChannelId channel_id) = 0;
+ // Starts sending on |channel_id|. This starts microphone if not started yet.
+ // Returns following VoipResult;
+ // kOk - Channel successfully started to send.
+ // kInvalidArgument - |channel_id| is invalid.
+ // kFailedPrecondition - Missing prerequisite on VoipCodec::SetSendCodec.
+ // kInternal - initialization has failed on selected microphone.
+ virtual VoipResult StartSend(ChannelId channel_id) = 0;
// Stops sending on |channel_id|. If this is the last active channel, it will
// stop microphone input from underlying audio platform layer.
- // Returns false if termination logic has failed on selected microphone
- // device. API is subject to expand to reflect error condition to application
- // later.
- virtual bool StopSend(ChannelId channel_id) = 0;
+ // Returns following VoipResult;
+ // kOk - Channel successfully stopped to send.
+ // kInvalidArgument - |channel_id| is invalid.
+ // kInternal - Failed to stop the active microphone device.
+ virtual VoipResult StopSend(ChannelId channel_id) = 0;
// Starts playing on speaker device for |channel_id|.
// This will start underlying platform speaker device if not started.
- // Returns false if initialization has failed
- // on selected speaker device. API is subject to expand to reflect error
- // condition to application later.
- virtual bool StartPlayout(ChannelId channel_id) = 0;
+ // Returns following VoipResult;
+ // kOk - Channel successfully started to play out.
+ // kInvalidArgument - |channel_id| is invalid.
+ // kFailedPrecondition - Missing prerequisite on VoipCodec::SetReceiveCodecs.
+ // kInternal - Failed to initializate the selected speaker device.
+ virtual VoipResult StartPlayout(ChannelId channel_id) = 0;
// Stops playing on speaker device for |channel_id|.
- // If this is the last active channel playing, then it will stop speaker
- // from the platform layer.
- // Returns false if termination logic has failed on selected speaker device.
- // API is subject to expand to reflect error condition to application later.
- virtual bool StopPlayout(ChannelId channel_id) = 0;
+ // Returns following VoipResult;
+ // kOk - Channel successfully stopped t play out.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult StopPlayout(ChannelId channel_id) = 0;
protected:
virtual ~VoipBase() = default;
diff --git a/api/voip/voip_codec.h b/api/voip/voip_codec.h
index eb42c44..fec3827 100644
--- a/api/voip/voip_codec.h
+++ b/api/voip/voip_codec.h
@@ -29,15 +29,21 @@
class VoipCodec {
public:
// Set encoder type here along with its payload type to use.
- virtual void SetSendCodec(ChannelId channel_id,
- int payload_type,
- const SdpAudioFormat& encoder_spec) = 0;
+ // Returns following VoipResult;
+ // kOk - sending codec is set as provided.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult SetSendCodec(ChannelId channel_id,
+ int payload_type,
+ const SdpAudioFormat& encoder_spec) = 0;
// Set decoder payload type here. In typical offer and answer model,
// this should be called after payload type has been agreed in media
// session. Note that payload type can differ with same codec in each
// direction.
- virtual void SetReceiveCodecs(
+ // Returns following VoipResult;
+ // kOk - receiving codecs are set as provided.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult SetReceiveCodecs(
ChannelId channel_id,
const std::map<int, SdpAudioFormat>& decoder_specs) = 0;
diff --git a/api/voip/voip_dtmf.h b/api/voip/voip_dtmf.h
index 56817ba..a7367be 100644
--- a/api/voip/voip_dtmf.h
+++ b/api/voip/voip_dtmf.h
@@ -43,9 +43,12 @@
// Register the payload type and sample rate for DTMF (RFC 4733) payload.
// Must be called exactly once prior to calling SendDtmfEvent after payload
// type has been negotiated with remote.
- virtual void RegisterTelephoneEventType(ChannelId channel_id,
- int rtp_payload_type,
- int sample_rate_hz) = 0;
+ // Returns following VoipResult;
+ // kOk - telephone event type is registered as provided.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult RegisterTelephoneEventType(ChannelId channel_id,
+ int rtp_payload_type,
+ int sample_rate_hz) = 0;
// Send DTMF named event as specified by
// https://tools.ietf.org/html/rfc4733#section-3.2
@@ -53,10 +56,14 @@
// in place of real RTP packets instead.
// Must be called after RegisterTelephoneEventType and VoipBase::StartSend
// have been called.
- // Returns true if the requested DTMF event is successfully scheduled.
- virtual bool SendDtmfEvent(ChannelId channel_id,
- DtmfEvent dtmf_event,
- int duration_ms) = 0;
+ // Returns following VoipResult;
+ // kOk - requested DTMF event is successfully scheduled.
+ // kInvalidArgument - |channel_id| is invalid.
+ // kFailedPrecondition - Missing prerequisite on RegisterTelephoneEventType
+ // or sending state.
+ virtual VoipResult SendDtmfEvent(ChannelId channel_id,
+ DtmfEvent dtmf_event,
+ int duration_ms) = 0;
protected:
virtual ~VoipDtmf() = default;
diff --git a/api/voip/voip_engine.h b/api/voip/voip_engine.h
index 69c0a85..d223f6a 100644
--- a/api/voip/voip_engine.h
+++ b/api/voip/voip_engine.h
@@ -23,7 +23,7 @@
// VoipEngine is the main interface serving as the entry point for all VoIP
// APIs. A single instance of VoipEngine should suffice the most of the need for
// typical VoIP applications as it handles multiple media sessions including a
-// specialized session type like ad-hoc mesh conferencing. Below example code
+// specialized session type like ad-hoc conference. Below example code
// describes the typical sequence of API usage. Each API header contains more
// description on what the methods are used for.
//
@@ -38,36 +38,35 @@
// config.audio_processing = AudioProcessingBuilder().Create();
//
// auto voip_engine = CreateVoipEngine(std::move(config));
-// if (!voip_engine) return some_failure;
//
// auto& voip_base = voip_engine->Base();
// auto& voip_codec = voip_engine->Codec();
// auto& voip_network = voip_engine->Network();
//
-// absl::optional<ChannelId> channel =
-// voip_base.CreateChannel(&app_transport_);
-// if (!channel) return some_failure;
+// ChannelId channel = voip_base.CreateChannel(&app_transport_);
//
// // After SDP offer/answer, set payload type and codecs that have been
// // decided through SDP negotiation.
-// voip_codec.SetSendCodec(*channel, ...);
-// voip_codec.SetReceiveCodecs(*channel, ...);
+// // VoipResult handling omitted here.
+// voip_codec.SetSendCodec(channel, ...);
+// voip_codec.SetReceiveCodecs(channel, ...);
//
// // Start sending and playing RTP on voip channel.
-// voip_base.StartSend(*channel);
-// voip_base.StartPlayout(*channel);
+// // VoipResult handling omitted here.
+// voip_base.StartSend(channel);
+// voip_base.StartPlayout(channel);
//
// // Inject received RTP/RTCP through VoipNetwork interface.
-// voip_network.ReceivedRTPPacket(*channel, ...);
-// voip_network.ReceivedRTCPPacket(*channel, ...);
+// // VoipResult handling omitted here.
+// voip_network.ReceivedRTPPacket(channel, ...);
+// voip_network.ReceivedRTCPPacket(channel, ...);
//
// // Stop and release voip channel.
-// voip_base.StopSend(*channel);
-// voip_base.StopPlayout(*channel);
-// voip_base.ReleaseChannel(*channel);
+// // VoipResult handling omitted here.
+// voip_base.StopSend(channel);
+// voip_base.StopPlayout(channel);
+// voip_base.ReleaseChannel(channel);
//
-// Current VoipEngine defines three sub-API classes and is subject to expand in
-// near future.
class VoipEngine {
public:
virtual ~VoipEngine() = default;
diff --git a/api/voip/voip_network.h b/api/voip/voip_network.h
index c49c769..c820ca0 100644
--- a/api/voip/voip_network.h
+++ b/api/voip/voip_network.h
@@ -18,20 +18,22 @@
// VoipNetwork interface provides any network related interfaces such as
// processing received RTP/RTCP packet from remote endpoint. This interface
-// requires a ChannelId created via VoipBase interface. Note that using invalid
-// (previously released) ChannelId will silently fail these API calls as it
-// would have released underlying audio components. It's anticipated that caller
-// may be using different thread for network I/O where released channel id is
-// still used to input incoming RTP packets in which case we should silently
-// ignore. The interface is subjected to expand as needed in near future.
+// requires a ChannelId created via VoipBase interface.
class VoipNetwork {
public:
// The data received from the network including RTP header is passed here.
- virtual void ReceivedRTPPacket(ChannelId channel_id,
- rtc::ArrayView<const uint8_t> rtp_packet) = 0;
+ // Returns following VoipResult;
+ // kOk - received RTP packet is processed.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult ReceivedRTPPacket(
+ ChannelId channel_id,
+ rtc::ArrayView<const uint8_t> rtp_packet) = 0;
// The data received from the network including RTCP header is passed here.
- virtual void ReceivedRTCPPacket(
+ // Returns following VoipResult;
+ // kOk - received RTCP packet is processed.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult ReceivedRTCPPacket(
ChannelId channel_id,
rtc::ArrayView<const uint8_t> rtcp_packet) = 0;
diff --git a/api/voip/voip_statistics.h b/api/voip/voip_statistics.h
index cf01e95..08f4cb7 100644
--- a/api/voip/voip_statistics.h
+++ b/api/voip/voip_statistics.h
@@ -30,10 +30,12 @@
// the jitter buffer (NetEq) performance.
class VoipStatistics {
public:
- // Gets the audio ingress statistics. Returns absl::nullopt when channel_id is
- // invalid.
- virtual absl::optional<IngressStatistics> GetIngressStatistics(
- ChannelId channel_id) = 0;
+ // Gets the audio ingress statistics by |ingress_stats| reference.
+ // Returns following VoipResult;
+ // kOk - successfully set provided IngressStatistics reference.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult GetIngressStatistics(ChannelId channel_id,
+ IngressStatistics& ingress_stats) = 0;
protected:
virtual ~VoipStatistics() = default;
diff --git a/api/voip/voip_volume_control.h b/api/voip/voip_volume_control.h
index 54e4467..d91eabc 100644
--- a/api/voip/voip_volume_control.h
+++ b/api/voip/voip_volume_control.h
@@ -36,17 +36,24 @@
// Mute/unmutes the microphone input sample before encoding process. Note that
// mute doesn't affect audio input level and energy values as input sample is
// silenced after the measurement.
- virtual void SetInputMuted(ChannelId channel_id, bool enable) = 0;
+ // Returns following VoipResult;
+ // kOk - input source muted or unmuted as provided by |enable|.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult SetInputMuted(ChannelId channel_id, bool enable) = 0;
- // Gets the microphone volume info.
- // Returns absl::nullopt if |channel_id| is invalid.
- virtual absl::optional<VolumeInfo> GetInputVolumeInfo(
- ChannelId channel_id) = 0;
+ // Gets the microphone volume info via |volume_info| reference.
+ // Returns following VoipResult;
+ // kOk - successfully set provided input volume info.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult GetInputVolumeInfo(ChannelId channel_id,
+ VolumeInfo& volume_info) = 0;
- // Gets the speaker volume info.
- // Returns absl::nullopt if |channel_id| is invalid.
- virtual absl::optional<VolumeInfo> GetOutputVolumeInfo(
- ChannelId channel_id) = 0;
+ // Gets the speaker volume info via |volume_info| reference.
+ // Returns following VoipResult;
+ // kOk - successfully set provided output volume info.
+ // kInvalidArgument - |channel_id| is invalid.
+ virtual VoipResult GetOutputVolumeInfo(ChannelId channel_id,
+ VolumeInfo& volume_info) = 0;
protected:
virtual ~VoipVolumeControl() = default;
diff --git a/audio/voip/test/voip_core_unittest.cc b/audio/voip/test/voip_core_unittest.cc
index 9763d58..8ab67b7 100644
--- a/audio/voip/test/voip_core_unittest.cc
+++ b/audio/voip/test/voip_core_unittest.cc
@@ -69,19 +69,19 @@
EXPECT_CALL(*audio_device_, StartPlayout()).WillOnce(Return(0));
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- EXPECT_TRUE(channel);
- voip_core_->SetSendCodec(*channel, kPcmuPayload, kPcmuFormat);
- voip_core_->SetReceiveCodecs(*channel, {{kPcmuPayload, kPcmuFormat}});
+ voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat);
+ voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}});
- EXPECT_TRUE(voip_core_->StartSend(*channel));
- EXPECT_TRUE(voip_core_->StartPlayout(*channel));
+ EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk);
+ EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kOk);
- voip_core_->RegisterTelephoneEventType(*channel, kPcmuPayload,
+ voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
kPcmuSampleRateHz);
- EXPECT_TRUE(voip_core_->SendDtmfEvent(*channel, kDtmfEventCode,
- kDtmfEventDurationMs));
+ EXPECT_EQ(
+ voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+ VoipResult::kOk);
// Program mock as operational that is ready to be stopped.
EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(true));
@@ -89,30 +89,32 @@
EXPECT_CALL(*audio_device_, StopRecording()).WillOnce(Return(0));
EXPECT_CALL(*audio_device_, StopPlayout()).WillOnce(Return(0));
- EXPECT_TRUE(voip_core_->StopSend(*channel));
- EXPECT_TRUE(voip_core_->StopPlayout(*channel));
- voip_core_->ReleaseChannel(*channel);
+ EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk);
+ EXPECT_EQ(voip_core_->StopPlayout(channel), VoipResult::kOk);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
}
TEST_F(VoipCoreTest, ExpectFailToUseReleasedChannelId) {
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- EXPECT_TRUE(channel);
// Release right after creation.
- voip_core_->ReleaseChannel(*channel);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
// Now use released channel.
- // These should be no-op.
- voip_core_->SetSendCodec(*channel, kPcmuPayload, kPcmuFormat);
- voip_core_->SetReceiveCodecs(*channel, {{kPcmuPayload, kPcmuFormat}});
- voip_core_->RegisterTelephoneEventType(*channel, kPcmuPayload,
- kPcmuSampleRateHz);
-
- EXPECT_FALSE(voip_core_->StartSend(*channel));
- EXPECT_FALSE(voip_core_->StartPlayout(*channel));
- EXPECT_FALSE(voip_core_->SendDtmfEvent(*channel, kDtmfEventCode,
- kDtmfEventDurationMs));
+ EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat),
+ VoipResult::kInvalidArgument);
+ EXPECT_EQ(
+ voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}),
+ VoipResult::kInvalidArgument);
+ EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
+ kPcmuSampleRateHz),
+ VoipResult::kInvalidArgument);
+ EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kInvalidArgument);
+ EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kInvalidArgument);
+ EXPECT_EQ(
+ voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+ VoipResult::kInvalidArgument);
}
TEST_F(VoipCoreTest, SendDtmfEventWithoutRegistering) {
@@ -122,64 +124,65 @@
EXPECT_CALL(*audio_device_, StartRecording()).WillOnce(Return(0));
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- EXPECT_TRUE(channel);
- voip_core_->SetSendCodec(*channel, kPcmuPayload, kPcmuFormat);
+ voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat);
- EXPECT_TRUE(voip_core_->StartSend(*channel));
+ EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk);
// Send Dtmf event without registering beforehand, thus payload
- // type is not set and false is expected.
- EXPECT_FALSE(voip_core_->SendDtmfEvent(*channel, kDtmfEventCode,
- kDtmfEventDurationMs));
+ // type is not set and kFailedPrecondition is expected.
+ EXPECT_EQ(
+ voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+ VoipResult::kFailedPrecondition);
// Program mock as sending and is ready to be stopped.
EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(true));
EXPECT_CALL(*audio_device_, StopRecording()).WillOnce(Return(0));
- EXPECT_TRUE(voip_core_->StopSend(*channel));
- voip_core_->ReleaseChannel(*channel);
+ EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
}
TEST_F(VoipCoreTest, SendDtmfEventWithoutStartSend) {
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- EXPECT_TRUE(channel);
- voip_core_->RegisterTelephoneEventType(*channel, kPcmuPayload,
+ voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
kPcmuSampleRateHz);
// Send Dtmf event without calling StartSend beforehand, thus
- // Dtmf events cannot be sent and false is expected.
- EXPECT_FALSE(voip_core_->SendDtmfEvent(*channel, kDtmfEventCode,
- kDtmfEventDurationMs));
+ // Dtmf events cannot be sent and kFailedPrecondition is expected.
+ EXPECT_EQ(
+ voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+ VoipResult::kFailedPrecondition);
- voip_core_->ReleaseChannel(*channel);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
}
TEST_F(VoipCoreTest, StartSendAndPlayoutWithoutSettingCodec) {
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- EXPECT_TRUE(channel);
// Call StartSend and StartPlayout without setting send/receive
// codec. Code should see that codecs aren't set and return false.
- EXPECT_FALSE(voip_core_->StartSend(*channel));
- EXPECT_FALSE(voip_core_->StartPlayout(*channel));
+ EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kFailedPrecondition);
+ EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kFailedPrecondition);
- voip_core_->ReleaseChannel(*channel);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
}
TEST_F(VoipCoreTest, StopSendAndPlayoutWithoutStarting) {
auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
- EXPECT_TRUE(channel);
- 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);
// Call StopSend and StopPlayout without starting them in
// the first place. Should see that it is already in the
// stopped state and return true.
- EXPECT_TRUE(voip_core_->StopSend(*channel));
- EXPECT_TRUE(voip_core_->StopPlayout(*channel));
+ EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk);
+ EXPECT_EQ(voip_core_->StopPlayout(channel), VoipResult::kOk);
- voip_core_->ReleaseChannel(*channel);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
}
// This tests correctness on ProcessThread usage where we expect the first/last
@@ -190,25 +193,22 @@
auto channel_one = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
auto channel_two = voip_core_->CreateChannel(&transport_, 0xdeadbeef);
- EXPECT_TRUE(channel_one);
- EXPECT_TRUE(channel_two);
EXPECT_CALL(*process_thread_, Stop);
EXPECT_CALL(*process_thread_, DeRegisterModule).Times(2);
- voip_core_->ReleaseChannel(*channel_one);
- voip_core_->ReleaseChannel(*channel_two);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel_one), VoipResult::kOk);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel_two), VoipResult::kOk);
EXPECT_CALL(*process_thread_, Start);
EXPECT_CALL(*process_thread_, RegisterModule);
auto channel_three = voip_core_->CreateChannel(&transport_, absl::nullopt);
- EXPECT_TRUE(channel_three);
EXPECT_CALL(*process_thread_, Stop);
EXPECT_CALL(*process_thread_, DeRegisterModule);
- voip_core_->ReleaseChannel(*channel_three);
+ EXPECT_EQ(voip_core_->ReleaseChannel(channel_three), VoipResult::kOk);
}
} // namespace
diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
index ac29fbf..f65352c 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -127,10 +127,9 @@
return true;
}
-absl::optional<ChannelId> VoipCore::CreateChannel(
- Transport* transport,
- absl::optional<uint32_t> local_ssrc) {
- absl::optional<ChannelId> channel_id;
+ChannelId VoipCore::CreateChannel(Transport* transport,
+ absl::optional<uint32_t> local_ssrc) {
+ ChannelId channel_id;
// Set local ssrc to random if not set by caller.
if (!local_ssrc) {
@@ -153,7 +152,7 @@
start_process_thread = channels_.empty();
channel_id = static_cast<ChannelId>(next_channel_id_);
- channels_[*channel_id] = channel;
+ channels_[channel_id] = channel;
next_channel_id_++;
if (next_channel_id_ >= kMaxChannelId) {
next_channel_id_ = 0;
@@ -161,7 +160,7 @@
}
// Set ChannelId in audio channel for logging/debugging purpose.
- channel->SetId(*channel_id);
+ channel->SetId(channel_id);
if (start_process_thread) {
process_thread_->Start();
@@ -170,7 +169,7 @@
return channel_id;
}
-void VoipCore::ReleaseChannel(ChannelId channel_id) {
+VoipResult VoipCore::ReleaseChannel(ChannelId channel_id) {
// Destroy channel outside of the lock.
rtc::scoped_refptr<AudioChannel> channel;
@@ -188,8 +187,10 @@
no_channels_after_release = channels_.empty();
}
+ VoipResult status_code = VoipResult::kOk;
if (!channel) {
RTC_LOG(LS_WARNING) << "Channel " << channel_id << " not found";
+ status_code = VoipResult::kInvalidArgument;
}
if (no_channels_after_release) {
@@ -201,9 +202,12 @@
if (audio_device_module_->Playing()) {
if (audio_device_module_->StopPlayout() != 0) {
RTC_LOG(LS_WARNING) << "StopPlayout failed";
+ status_code = VoipResult::kInternal;
}
}
}
+
+ return status_code;
}
rtc::scoped_refptr<AudioChannel> VoipCore::GetChannel(ChannelId channel_id) {
@@ -281,174 +285,219 @@
return true;
}
-bool VoipCore::StartSend(ChannelId channel_id) {
- rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
-
- if (!channel || !channel->StartSend()) {
- return false;
- }
-
- return UpdateAudioTransportWithSenders();
-}
-
-bool VoipCore::StopSend(ChannelId channel_id) {
+VoipResult VoipCore::StartSend(ChannelId channel_id) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
if (!channel) {
- return false;
+ return VoipResult::kInvalidArgument;
+ }
+
+ if (!channel->StartSend()) {
+ return VoipResult::kFailedPrecondition;
+ }
+
+ return UpdateAudioTransportWithSenders() ? VoipResult::kOk
+ : VoipResult::kInternal;
+}
+
+VoipResult VoipCore::StopSend(ChannelId channel_id) {
+ rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
channel->StopSend();
- return UpdateAudioTransportWithSenders();
+ return UpdateAudioTransportWithSenders() ? VoipResult::kOk
+ : VoipResult::kInternal;
}
-bool VoipCore::StartPlayout(ChannelId channel_id) {
+VoipResult VoipCore::StartPlayout(ChannelId channel_id) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
if (!channel) {
- return false;
+ return VoipResult::kInvalidArgument;
}
if (channel->IsPlaying()) {
- return true;
+ return VoipResult::kOk;
}
if (!channel->StartPlay()) {
- return false;
+ return VoipResult::kFailedPrecondition;
}
// Initialize audio device module and default device if needed.
if (!InitializeIfNeeded()) {
- return false;
+ return VoipResult::kInternal;
}
if (!audio_device_module_->Playing()) {
if (audio_device_module_->InitPlayout() != 0) {
RTC_LOG(LS_ERROR) << "InitPlayout failed";
- return false;
+ return VoipResult::kInternal;
}
if (audio_device_module_->StartPlayout() != 0) {
RTC_LOG(LS_ERROR) << "StartPlayout failed";
- return false;
+ return VoipResult::kInternal;
}
}
- return true;
+
+ return VoipResult::kOk;
}
-bool VoipCore::StopPlayout(ChannelId channel_id) {
+VoipResult VoipCore::StopPlayout(ChannelId channel_id) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
if (!channel) {
- return false;
+ return VoipResult::kInvalidArgument;
}
channel->StopPlay();
- return true;
+ return VoipResult::kOk;
}
-void VoipCore::ReceivedRTPPacket(ChannelId channel_id,
- rtc::ArrayView<const uint8_t> rtp_packet) {
+VoipResult VoipCore::ReceivedRTPPacket(
+ ChannelId channel_id,
+ rtc::ArrayView<const uint8_t> rtp_packet) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- channel->ReceivedRTPPacket(rtp_packet);
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
+
+ channel->ReceivedRTPPacket(rtp_packet);
+
+ return VoipResult::kOk;
}
-void VoipCore::ReceivedRTCPPacket(ChannelId channel_id,
- rtc::ArrayView<const uint8_t> rtcp_packet) {
+VoipResult VoipCore::ReceivedRTCPPacket(
+ ChannelId channel_id,
+ rtc::ArrayView<const uint8_t> rtcp_packet) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- channel->ReceivedRTCPPacket(rtcp_packet);
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
+
+ channel->ReceivedRTCPPacket(rtcp_packet);
+
+ return VoipResult::kOk;
}
-void VoipCore::SetSendCodec(ChannelId channel_id,
- int payload_type,
- const SdpAudioFormat& encoder_format) {
+VoipResult VoipCore::SetSendCodec(ChannelId channel_id,
+ int payload_type,
+ const SdpAudioFormat& encoder_format) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- auto encoder = encoder_factory_->MakeAudioEncoder(
- payload_type, encoder_format, absl::nullopt);
- channel->SetEncoder(payload_type, encoder_format, std::move(encoder));
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
+
+ auto encoder = encoder_factory_->MakeAudioEncoder(
+ payload_type, encoder_format, absl::nullopt);
+ channel->SetEncoder(payload_type, encoder_format, std::move(encoder));
+
+ return VoipResult::kOk;
}
-void VoipCore::SetReceiveCodecs(
+VoipResult VoipCore::SetReceiveCodecs(
ChannelId channel_id,
const std::map<int, SdpAudioFormat>& decoder_specs) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- channel->SetReceiveCodecs(decoder_specs);
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
+
+ channel->SetReceiveCodecs(decoder_specs);
+
+ return VoipResult::kOk;
}
-void VoipCore::RegisterTelephoneEventType(ChannelId channel_id,
- int rtp_payload_type,
- int sample_rate_hz) {
+VoipResult VoipCore::RegisterTelephoneEventType(ChannelId channel_id,
+ int rtp_payload_type,
+ int sample_rate_hz) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- channel->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz);
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
+
+ channel->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz);
+
+ return VoipResult::kOk;
}
-bool VoipCore::SendDtmfEvent(ChannelId channel_id,
- DtmfEvent dtmf_event,
- int duration_ms) {
+VoipResult VoipCore::SendDtmfEvent(ChannelId channel_id,
+ DtmfEvent dtmf_event,
+ int duration_ms) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- return channel->SendTelephoneEvent(static_cast<int>(dtmf_event),
- duration_ms);
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
- return false;
+
+ return (channel->SendTelephoneEvent(static_cast<int>(dtmf_event), duration_ms)
+ ? VoipResult::kOk
+ : VoipResult::kFailedPrecondition);
}
-absl::optional<IngressStatistics> VoipCore::GetIngressStatistics(
- ChannelId channel_id) {
+VoipResult VoipCore::GetIngressStatistics(ChannelId channel_id,
+ IngressStatistics& ingress_stats) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- return channel->GetIngressStatistics();
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
- return absl::nullopt;
+
+ ingress_stats = channel->GetIngressStatistics();
+
+ return VoipResult::kOk;
}
-void VoipCore::SetInputMuted(ChannelId channel_id, bool enable) {
+VoipResult VoipCore::SetInputMuted(ChannelId channel_id, bool enable) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- channel->SetMute(enable);
+
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
+
+ channel->SetMute(enable);
+
+ return VoipResult::kOk;
}
-absl::optional<VolumeInfo> VoipCore::GetInputVolumeInfo(ChannelId channel_id) {
+VoipResult VoipCore::GetInputVolumeInfo(ChannelId channel_id,
+ VolumeInfo& input_volume) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- VolumeInfo input_volume;
- input_volume.audio_level = channel->GetInputAudioLevel();
- input_volume.total_energy = channel->GetInputTotalEnergy();
- input_volume.total_duration = channel->GetInputTotalDuration();
- return input_volume;
+
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
- return absl::nullopt;
+
+ input_volume.audio_level = channel->GetInputAudioLevel();
+ input_volume.total_energy = channel->GetInputTotalEnergy();
+ input_volume.total_duration = channel->GetInputTotalDuration();
+
+ return VoipResult::kOk;
}
-absl::optional<VolumeInfo> VoipCore::GetOutputVolumeInfo(ChannelId channel_id) {
+VoipResult VoipCore::GetOutputVolumeInfo(ChannelId channel_id,
+ VolumeInfo& output_volume) {
rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
- if (channel) {
- VolumeInfo output_volume;
- output_volume.audio_level = channel->GetOutputAudioLevel();
- output_volume.total_energy = channel->GetOutputTotalEnergy();
- output_volume.total_duration = channel->GetOutputTotalDuration();
- return output_volume;
+
+ if (!channel) {
+ return VoipResult::kInvalidArgument;
}
- return absl::nullopt;
+
+ output_volume.audio_level = channel->GetOutputAudioLevel();
+ output_volume.total_energy = channel->GetOutputTotalEnergy();
+ output_volume.total_duration = channel->GetOutputTotalDuration();
+
+ return VoipResult::kOk;
}
} // namespace webrtc
diff --git a/audio/voip/voip_core.h b/audio/voip/voip_core.h
index 5ebf438..194f8fb 100644
--- a/audio/voip/voip_core.h
+++ b/audio/voip/voip_core.h
@@ -74,45 +74,48 @@
VoipVolumeControl& VolumeControl() override { return *this; }
// Implements VoipBase interfaces.
- absl::optional<ChannelId> CreateChannel(
- Transport* transport,
- absl::optional<uint32_t> local_ssrc) override;
- void ReleaseChannel(ChannelId channel_id) override;
- bool StartSend(ChannelId channel_id) override;
- bool StopSend(ChannelId channel_id) override;
- bool StartPlayout(ChannelId channel_id) override;
- bool StopPlayout(ChannelId channel_id) override;
+ ChannelId CreateChannel(Transport* transport,
+ absl::optional<uint32_t> local_ssrc) override;
+ VoipResult ReleaseChannel(ChannelId channel_id) override;
+ VoipResult StartSend(ChannelId channel_id) override;
+ VoipResult StopSend(ChannelId channel_id) override;
+ VoipResult StartPlayout(ChannelId channel_id) override;
+ VoipResult StopPlayout(ChannelId channel_id) override;
// Implements VoipNetwork interfaces.
- void ReceivedRTPPacket(ChannelId channel_id,
- rtc::ArrayView<const uint8_t> rtp_packet) override;
- void ReceivedRTCPPacket(ChannelId channel_id,
- rtc::ArrayView<const uint8_t> rtcp_packet) override;
+ VoipResult ReceivedRTPPacket(
+ ChannelId channel_id,
+ rtc::ArrayView<const uint8_t> rtp_packet) override;
+ VoipResult ReceivedRTCPPacket(
+ ChannelId channel_id,
+ rtc::ArrayView<const uint8_t> rtcp_packet) override;
// Implements VoipCodec interfaces.
- void SetSendCodec(ChannelId channel_id,
- int payload_type,
- const SdpAudioFormat& encoder_format) override;
- void SetReceiveCodecs(
+ VoipResult SetSendCodec(ChannelId channel_id,
+ int payload_type,
+ const SdpAudioFormat& encoder_format) override;
+ VoipResult SetReceiveCodecs(
ChannelId channel_id,
const std::map<int, SdpAudioFormat>& decoder_specs) override;
// Implements VoipDtmf interfaces.
- void RegisterTelephoneEventType(ChannelId channel_id,
- int rtp_payload_type,
- int sample_rate_hz) override;
- bool SendDtmfEvent(ChannelId channel_id,
- DtmfEvent dtmf_event,
- int duration_ms) override;
+ VoipResult RegisterTelephoneEventType(ChannelId channel_id,
+ int rtp_payload_type,
+ int sample_rate_hz) override;
+ VoipResult SendDtmfEvent(ChannelId channel_id,
+ DtmfEvent dtmf_event,
+ int duration_ms) override;
// Implements VoipStatistics interfaces.
- absl::optional<IngressStatistics> GetIngressStatistics(
- ChannelId channel_id) override;
+ VoipResult GetIngressStatistics(ChannelId channel_id,
+ IngressStatistics& ingress_stats) override;
// Implements VoipVolumeControl interfaces.
- void SetInputMuted(ChannelId channel_id, bool enable) override;
- absl::optional<VolumeInfo> GetInputVolumeInfo(ChannelId channel_id) override;
- absl::optional<VolumeInfo> GetOutputVolumeInfo(ChannelId channel_id) override;
+ VoipResult SetInputMuted(ChannelId channel_id, bool enable) override;
+ VoipResult GetInputVolumeInfo(ChannelId channel_id,
+ VolumeInfo& volume_info) override;
+ VoipResult GetOutputVolumeInfo(ChannelId channel_id,
+ VolumeInfo& volume_info) override;
private:
// Initialize ADM and default audio device if needed.
diff --git a/examples/androidvoip/jni/android_voip_client.cc b/examples/androidvoip/jni/android_voip_client.cc
index 2ad95bc..d0763cd 100644
--- a/examples/androidvoip/jni/android_voip_client.cc
+++ b/examples/androidvoip/jni/android_voip_client.cc
@@ -347,8 +347,8 @@
/*isSuccessful=*/false);
return;
}
- if (!voip_engine_->Base().StopSend(*channel_) ||
- !voip_engine_->Base().StopPlayout(*channel_)) {
+ if (voip_engine_->Base().StopSend(*channel_) != webrtc::VoipResult::kOk ||
+ voip_engine_->Base().StopPlayout(*channel_) != webrtc::VoipResult::kOk) {
Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_,
/*isSuccessful=*/false);
return;
@@ -372,8 +372,9 @@
/*isSuccessful=*/false);
return;
}
- Java_VoipClient_onStartSendCompleted(
- env_, j_voip_client_, voip_engine_->Base().StartSend(*channel_));
+ bool sending_started =
+ (voip_engine_->Base().StartSend(*channel_) == webrtc::VoipResult::kOk);
+ Java_VoipClient_onStartSendCompleted(env_, j_voip_client_, sending_started);
}
void AndroidVoipClient::StopSend(JNIEnv* env) {
@@ -385,8 +386,9 @@
/*isSuccessful=*/false);
return;
}
- Java_VoipClient_onStopSendCompleted(env_, j_voip_client_,
- voip_engine_->Base().StopSend(*channel_));
+ bool sending_stopped =
+ (voip_engine_->Base().StopSend(*channel_) == webrtc::VoipResult::kOk);
+ Java_VoipClient_onStopSendCompleted(env_, j_voip_client_, sending_stopped);
}
void AndroidVoipClient::StartPlayout(JNIEnv* env) {
@@ -398,8 +400,10 @@
/*isSuccessful=*/false);
return;
}
- Java_VoipClient_onStartPlayoutCompleted(
- env_, j_voip_client_, voip_engine_->Base().StartPlayout(*channel_));
+ bool playout_started =
+ (voip_engine_->Base().StartPlayout(*channel_) == webrtc::VoipResult::kOk);
+ Java_VoipClient_onStartPlayoutCompleted(env_, j_voip_client_,
+ playout_started);
}
void AndroidVoipClient::StopPlayout(JNIEnv* env) {
@@ -411,8 +415,9 @@
/*isSuccessful=*/false);
return;
}
- Java_VoipClient_onStopPlayoutCompleted(
- env_, j_voip_client_, voip_engine_->Base().StopPlayout(*channel_));
+ bool playout_stopped =
+ (voip_engine_->Base().StopPlayout(*channel_) == webrtc::VoipResult::kOk);
+ Java_VoipClient_onStopPlayoutCompleted(env_, j_voip_client_, playout_stopped);
}
void AndroidVoipClient::Delete(JNIEnv* env) {
