diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc index 00c1f04f77..463ff7b2ed 100644 --- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc @@ -1648,7 +1648,7 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) { EXPECT_CALL(mock_encoder, GetTargetBitrate()) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate)); - EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder, EncodeImpl(_, _, _)) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&encoder, static_cast< diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc index bc68f155bb..8cc59d6151 100644 --- a/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc +++ b/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc @@ -122,14 +122,14 @@ TEST(RentACodecTest, ExternalEncoder) { ::testing::InSequence s; info.encoded_timestamp = 0; EXPECT_CALL(external_encoder, - EncodeInternal(0, rtc::ArrayView(audio), + EncodeImpl(0, rtc::ArrayView(audio), &encoded)) .WillOnce(Return(info)); EXPECT_CALL(external_encoder, Mark("A")); EXPECT_CALL(external_encoder, Mark("B")); info.encoded_timestamp = 2; EXPECT_CALL(external_encoder, - EncodeInternal(2, rtc::ArrayView(audio), + EncodeImpl(2, rtc::ArrayView(audio), &encoded)) .WillOnce(Return(info)); EXPECT_CALL(external_encoder, Die()); diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc index 0520918f9f..6f793e2531 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc @@ -32,7 +32,7 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode( static_cast(NumChannels() * SampleRateHz() / 100)); const size_t old_size = encoded->size(); - EncodedInfo info = EncodeInternal(rtp_timestamp, audio, encoded); + EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded); RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes); return info; } @@ -59,7 +59,7 @@ AudioEncoder::EncodedInfo AudioEncoder::DEPRECATED_Encode( return info; } -AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoder::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) @@ -80,7 +80,7 @@ AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal( uint8_t* encoded) { rtc::Buffer temp_buffer; - EncodedInfo info = EncodeInternal(rtp_timestamp, audio, &temp_buffer); + EncodedInfo info = EncodeImpl(rtp_timestamp, audio, &temp_buffer); RTC_DCHECK_LE(temp_buffer.size(), max_encoded_bytes); std::memcpy(encoded, temp_buffer.data(), info.encoded_bytes); return info; diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h index 8da7ebd728..3fdee259ce 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h @@ -89,7 +89,7 @@ class AudioEncoder { // NumChannels() samples). Multi-channel audio must be sample-interleaved. // The encoder appends zero or more bytes of output to |encoded| and returns // additional encoding information. Encode() checks some preconditions, calls - // EncodeInternal() which does the actual work, and then checks some + // EncodeImpl() which does the actual work, and then checks some // postconditions. EncodedInfo Encode(uint32_t rtp_timestamp, rtc::ArrayView audio, @@ -110,12 +110,12 @@ class AudioEncoder { size_t max_encoded_bytes, uint8_t* encoded); - // Deprecated interface of EncodeInternal (also bug 5591). May incur a copy. + // Deprecated interface EncodeInternal (see bug 5591). May incur a copy. // Subclasses implement this to perform the actual encoding. Called by // Encode(). By default, this is implemented as a call to the newer - // EncodeInternal() that accepts an rtc::Buffer instead of a raw pointer. - // That version is protected, so see below. At least one of the two - // interfaces of EncodeInternal _must_ be implemented by a subclass. + // EncodeImpl() that accepts an rtc::Buffer instead of a raw pointer. + // That version is protected, so see below. At least one of EncodeInternal + // or EncodeImpl _must_ be implemented by a subclass. virtual EncodedInfo EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView audio, @@ -163,12 +163,12 @@ class AudioEncoder { protected: // Subclasses implement this to perform the actual encoding. Called by // Encode(). For compatibility reasons, this is implemented by default as a - // call to the older version of EncodeInternal(). At least one of the two - // interfaces of EncodeInternal _must_ be implemented by a subclass. - // Preferably this one. - virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded); + // call to the older interface EncodeInternal(). At least one of + // EncodeInternal or EncodeImpl _must_ be implemented by a + // subclass. Preferably this one. + virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded); }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc b/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc index 8252c6a250..71ffcde323 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc +++ b/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc @@ -37,7 +37,7 @@ TEST(AudioEncoderTest, EncodeInternalRedirectsOk) { EXPECT_CALL(old_impl, EncodeInternal(_, _, _, _)).WillOnce( Invoke(MockAudioEncoderDeprecated::CopyEncoding(payload))); - EXPECT_CALL(new_impl, EncodeInternal(_, _, _)).WillOnce( + EXPECT_CALL(new_impl, EncodeImpl(_, _, _)).WillOnce( Invoke(MockAudioEncoder::CopyEncoding(payload))); int16_t audio[80]; diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc index ab699cae79..5daf7be304 100644 --- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc +++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc @@ -97,7 +97,7 @@ int AudioEncoderCng::GetTargetBitrate() const { return speech_encoder_->GetTargetBitrate(); } -AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h index 3c24260774..b581e32082 100644 --- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h +++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h @@ -30,8 +30,6 @@ class Vad; class AudioEncoderCng final : public AudioEncoder { public: - using AudioEncoder::EncodeInternal; - struct Config { bool IsOk() const; @@ -59,9 +57,9 @@ class AudioEncoderCng final : public AudioEncoder { size_t Num10MsFramesInNextPacket() const override; size_t Max10MsFramesInAPacket() const override; int GetTargetBitrate() const override; - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded) override; void Reset() override; bool SetFec(bool enable) override; bool SetDtx(bool enable) override; diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc index 459ccb58c0..a4416c955a 100644 --- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc +++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc @@ -88,11 +88,11 @@ class AudioEncoderCngTest : public ::testing::Test { InSequence s; AudioEncoder::EncodedInfo info; for (size_t j = 0; j < num_calls - 1; ++j) { - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Return(info)); } info.encoded_bytes = kMockReturnEncodedBytes; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Invoke( MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes))); } @@ -108,7 +108,7 @@ class AudioEncoderCngTest : public ::testing::Test { .WillRepeatedly(Return(active_speech ? Vad::kActive : Vad::kPassive)); // Don't expect any calls to the encoder yet. - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)).Times(0); + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)).Times(0); for (size_t i = 0; i < blocks_per_frame - 1; ++i) { Encode(); EXPECT_EQ(0u, encoded_info_.encoded_bytes); @@ -259,7 +259,7 @@ TEST_F(AudioEncoderCngTest, EncodePassive) { EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)) .WillRepeatedly(Return(Vad::kPassive)); // Expect no calls at all to the speech encoder mock. - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)).Times(0); + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)).Times(0); uint32_t expected_timestamp = timestamp_; for (size_t i = 0; i < 100; ++i) { Encode(); @@ -341,7 +341,7 @@ TEST_F(AudioEncoderCngTest, VadInputSize60Ms) { // Verifies that the correct payload type is set when CNG is encoded. TEST_F(AudioEncoderCngTest, VerifyCngPayloadType) { CreateCng(); - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)).Times(0); + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)).Times(0); EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1U)); EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)) .WillOnce(Return(Vad::kPassive)); @@ -373,7 +373,7 @@ TEST_F(AudioEncoderCngTest, VerifySidFrameAfterSpeech) { encoded_info_.payload_type = 0; EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)) .WillOnce(Return(Vad::kActive)); - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)).WillOnce( + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)).WillOnce( Invoke(MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes))); Encode(); EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes); diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc index d850675244..a24b1526fd 100644 --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc @@ -77,7 +77,7 @@ int AudioEncoderPcm::GetTargetBitrate() const { 8 * BytesPerSample() * SampleRateHz() * NumChannels()); } -AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h index 0b8baab209..6b3cebfb33 100644 --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h @@ -20,8 +20,6 @@ namespace webrtc { class AudioEncoderPcm : public AudioEncoder { public: - using AudioEncoder::EncodeInternal; - struct Config { public: bool IsOk() const; @@ -48,9 +46,9 @@ class AudioEncoderPcm : public AudioEncoder { protected: AudioEncoderPcm(const Config& config, int sample_rate_hz); - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded) override; virtual size_t EncodeCall(const int16_t* audio, size_t input_len, diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc index 57d95947c3..9256518445 100644 --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc @@ -97,7 +97,7 @@ void AudioEncoderG722::Reset() { RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder)); } -AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoderG722::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h index 0d7b919aa2..dec87b2b7a 100644 --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h @@ -23,8 +23,6 @@ struct CodecInst; class AudioEncoderG722 final : public AudioEncoder { public: - using AudioEncoder::EncodeInternal; - struct Config { bool IsOk() const; @@ -47,9 +45,9 @@ class AudioEncoderG722 final : public AudioEncoder { void Reset() override; protected: - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded) override; private: // The encoder state for one channel. diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc index 419fdfb54a..c7d7411c45 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc @@ -89,7 +89,7 @@ int AudioEncoderIlbc::GetTargetBitrate() const { } } -AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h index e0ba289d76..27329bbc4e 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h @@ -21,8 +21,6 @@ struct CodecInst; class AudioEncoderIlbc final : public AudioEncoder { public: - using AudioEncoder::EncodeInternal; - struct Config { bool IsOk() const; @@ -42,9 +40,9 @@ class AudioEncoderIlbc final : public AudioEncoder { size_t Num10MsFramesInNextPacket() const override; size_t Max10MsFramesInAPacket() const override; int GetTargetBitrate() const override; - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded) override; void Reset() override; private: diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h index 2e15fb5620..e840f3b1b1 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h @@ -23,8 +23,6 @@ struct CodecInst; template class AudioEncoderIsacT final : public AudioEncoder { public: - using AudioEncoder::EncodeInternal; - // Allowed combinations of sample rate, frame size, and bit rate are // - 16000 Hz, 30 ms, 10000-32000 bps // - 16000 Hz, 60 ms, 10000-32000 bps @@ -62,9 +60,9 @@ class AudioEncoderIsacT final : public AudioEncoder { size_t Num10MsFramesInNextPacket() const override; size_t Max10MsFramesInAPacket() const override; int GetTargetBitrate() const override; - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded) override; void Reset() override; private: diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h index 3832216d57..3ad0f27d4b 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h @@ -114,7 +114,7 @@ int AudioEncoderIsacT::GetTargetBitrate() const { } template -AudioEncoder::EncodedInfo AudioEncoderIsacT::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoderIsacT::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h index 09cd078953..58a1e756f9 100644 --- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h @@ -43,10 +43,8 @@ class MockAudioEncoderBase : public AudioEncoder { class MockAudioEncoder final : public MockAudioEncoderBase { public: - using AudioEncoder::EncodeInternal; - // Note, we explicitly chose not to create a mock for the Encode method. - MOCK_METHOD3(EncodeInternal, + MOCK_METHOD3(EncodeImpl, EncodedInfo(uint32_t timestamp, rtc::ArrayView audio, rtc::Buffer* encoded)); @@ -96,8 +94,6 @@ class MockAudioEncoder final : public MockAudioEncoderBase { class MockAudioEncoderDeprecated final : public MockAudioEncoderBase { public: - using AudioEncoder::EncodeInternal; - // Note, we explicitly chose not to create a mock for the Encode method. MOCK_METHOD4(EncodeInternal, EncodedInfo(uint32_t timestamp, diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc index 1f68e154ce..a599e291d4 100644 --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -182,7 +182,7 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) { RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps)); } -AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h index 343acd7bd0..3f11af1f9e 100644 --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -23,8 +23,6 @@ struct CodecInst; class AudioEncoderOpus final : public AudioEncoder { public: - using AudioEncoder::EncodeInternal; - enum ApplicationMode { kVoip = 0, kAudio = 1, @@ -82,9 +80,9 @@ class AudioEncoderOpus final : public AudioEncoder { bool dtx_enabled() const { return config_.dtx_enabled; } protected: - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded) override; private: size_t Num10msFramesPerPacket() const; diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h index d298b992ba..34a780b49d 100644 --- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h +++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h @@ -19,8 +19,6 @@ struct CodecInst; class AudioEncoderPcm16B final : public AudioEncoderPcm { public: - using AudioEncoder::EncodeInternal; - struct Config : public AudioEncoderPcm::Config { public: Config() : AudioEncoderPcm::Config(107), sample_rate_hz(8000) {} diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc index 1dde49d1ba..b5a124e43d 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc @@ -52,7 +52,7 @@ int AudioEncoderCopyRed::GetTargetBitrate() const { return speech_encoder_->GetTargetBitrate(); } -AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeInternal( +AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h index 48cec6549d..2aa8edcfcd 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h @@ -24,8 +24,6 @@ namespace webrtc { // into one packet. class AudioEncoderCopyRed final : public AudioEncoder { public: - using AudioEncoder::EncodeInternal; - struct Config { public: int payload_type; @@ -53,9 +51,9 @@ class AudioEncoderCopyRed final : public AudioEncoder { void SetTargetBitrate(int target_bps) override; protected: - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView audio, + rtc::Buffer* encoded) override; private: AudioEncoder* speech_encoder_; diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc index 74b3f30087..fefcbe22f8 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc @@ -115,12 +115,12 @@ TEST_F(AudioEncoderCopyRedTest, CheckProjectedPacketLossRatePropagation) { // encoder. TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) { // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction - // check ensures that exactly one call to EncodeInternal happens in each + // check ensures that exactly one call to EncodeImpl happens in each // Encode call. InSequence s; MockFunction check; for (int i = 1; i <= 6; ++i) { - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillRepeatedly(Return(AudioEncoder::EncodedInfo())); EXPECT_CALL(check, Call(i)); Encode(); @@ -134,7 +134,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckNoOutput) { static const size_t kEncodedSize = 17; { InSequence s; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize))) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(0))) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize))); @@ -165,7 +165,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes) { static const int kNumPackets = 10; InSequence s; for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) { - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size))); } @@ -191,7 +191,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckTimestamps) { info.encoded_bytes = 17; info.encoded_timestamp = timestamp_; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info))); // First call is a special case, since it does not include a secondary @@ -202,7 +202,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckTimestamps) { uint32_t secondary_timestamp = primary_timestamp; primary_timestamp = timestamp_; info.encoded_timestamp = timestamp_; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info))); Encode(); @@ -221,7 +221,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloads) { for (uint8_t i = 0; i < kPayloadLenBytes; ++i) { payload[i] = i; } - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillRepeatedly(Invoke(MockAudioEncoder::CopyEncoding(payload))); // First call is a special case, since it does not include a secondary @@ -257,7 +257,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadType) { AudioEncoder::EncodedInfo info; info.encoded_bytes = 17; info.payload_type = primary_payload_type; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info))); // First call is a special case, since it does not include a secondary @@ -269,7 +269,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadType) { const int secondary_payload_type = red_payload_type_ + 2; info.payload_type = secondary_payload_type; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _)) + EXPECT_CALL(mock_encoder_, EncodeImpl(_, _, _)) .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info))); Encode();