Enable clang::find_bad_constructs for audio_coding (part 1/2).
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7
Reviewed-on: https://webrtc-review.googlesource.com/89743
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24053}diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
index f0d5b6f..c576670 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -27,6 +27,8 @@
CreatePacket();
}
+EncodeNetEqInput::~EncodeNetEqInput() = default;
+
absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
RTC_DCHECK(packet_data_);
return static_cast<int64_t>(packet_data_->time_ms);
@@ -50,6 +52,10 @@
next_output_event_ms_ += kOutputPeriodMs;
}
+bool EncodeNetEqInput::ended() const {
+ return next_output_event_ms_ <= input_duration_ms_;
+}
+
absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
RTC_DCHECK(packet_data_);
return packet_data_->header;
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.h b/modules/audio_coding/neteq/tools/encode_neteq_input.h
index a75262b..caa9ac7 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.h
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -35,6 +35,7 @@
EncodeNetEqInput(std::unique_ptr<Generator> generator,
std::unique_ptr<AudioEncoder> encoder,
int64_t input_duration_ms);
+ ~EncodeNetEqInput() override;
absl::optional<int64_t> NextPacketTime() const override;
@@ -44,9 +45,7 @@
void AdvanceOutputEvent() override;
- bool ended() const override {
- return next_output_event_ms_ <= input_duration_ms_;
- }
+ bool ended() const override;
absl::optional<RTPHeader> NextHeader() const override;
diff --git a/modules/audio_coding/neteq/tools/neteq_input.cc b/modules/audio_coding/neteq/tools/neteq_input.cc
index fed3ecc..bb2a01e 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_input.cc
@@ -15,6 +15,9 @@
namespace webrtc {
namespace test {
+NetEqInput::PacketData::PacketData() = default;
+NetEqInput::PacketData::~PacketData() = default;
+
std::string NetEqInput::PacketData::ToString() const {
std::stringstream ss;
ss << "{"
@@ -34,6 +37,8 @@
start_time_ms_(input_->NextEventTime()),
duration_ms_(duration_ms) {}
+TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
+
absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
return ended_ ? absl::nullopt : input_->NextPacketTime();
}
diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h
index 1691585..b9cba09 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_input.h
@@ -28,6 +28,8 @@
class NetEqInput {
public:
struct PacketData {
+ PacketData();
+ ~PacketData();
std::string ToString() const;
RTPHeader header;
@@ -84,6 +86,7 @@
class TimeLimitedNetEqInput : public NetEqInput {
public:
TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
+ ~TimeLimitedNetEqInput() override;
absl::optional<int64_t> NextPacketTime() const override;
absl::optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index e2bed56..3448f4e 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -46,6 +46,8 @@
RegisterExternalDecoders(ext_codecs);
}
+NetEqTest::~NetEqTest() = default;
+
int64_t NetEqTest::Run() {
const int64_t start_time_ms = *input_->NextEventTime();
int64_t time_now_ms = start_time_ms;
diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h
index 9c05fc4..ada6bab 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_test.h
@@ -82,7 +82,7 @@
std::unique_ptr<AudioSink> output,
Callbacks callbacks);
- ~NetEqTest() = default;
+ ~NetEqTest();
// Runs the test. Returns the duration of the produced audio in ms.
int64_t Run();
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
index b44bc64..013333b 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -39,7 +39,7 @@
static bool ValidRtpDump(const std::string& file_name);
static bool ValidPcap(const std::string& file_name);
- virtual ~RtpFileSource();
+ ~RtpFileSource() override;
// Registers an RTP header extension and binds it to |id|.
virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);