diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc index ae1bce179..ae82509ef 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc @@ -26,15 +26,15 @@ bool AudioEncoder::Encode(uint32_t rtp_timestamp, uint8_t* encoded, EncodedInfo* info) { CHECK_EQ(num_samples_per_channel, - static_cast(sample_rate_hz() / 100)); + static_cast(SampleRateHz() / 100)); bool ret = EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded, info); CHECK_LE(info->encoded_bytes, max_encoded_bytes); return ret; } -int AudioEncoder::rtp_timestamp_rate_hz() const { - return sample_rate_hz(); +int AudioEncoder::RtpTimestampRateHz() const { + return SampleRateHz(); } } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h index 91fff138a..c02c3efd2 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h @@ -70,12 +70,12 @@ class AudioEncoder { // Return the input sample rate in Hz and the number of input channels. // These are constants set at instantiation time. - virtual int sample_rate_hz() const = 0; - virtual int num_channels() const = 0; + virtual int SampleRateHz() const = 0; + virtual int NumChannels() const = 0; // Returns the rate with which the RTP timestamps are updated. By default, // this is the same as sample_rate_hz(). - virtual int rtp_timestamp_rate_hz() const; + virtual int RtpTimestampRateHz() const; // Returns the number of 10 ms frames the encoder will put in the next // packet. This value may only change when Encode() outputs a packet; i.e., 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 c85f60489..bb53e8c9f 100644 --- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc +++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc @@ -29,7 +29,7 @@ bool AudioEncoderCng::Config::IsOk() const { return false; if (!speech_encoder) return false; - if (num_channels != speech_encoder->num_channels()) + if (num_channels != speech_encoder->NumChannels()) return false; if (sid_frame_interval_ms < speech_encoder->Max10MsFramesInAPacket() * 10) return false; @@ -55,7 +55,7 @@ AudioEncoderCng::AudioEncoderCng(const Config& config) CNG_enc_inst* cng_inst; CHECK_EQ(WebRtcCng_CreateEnc(&cng_inst), 0) << "WebRtcCng_CreateEnc failed."; cng_inst_.reset(cng_inst); // Transfer ownership to scoped_ptr. - CHECK_EQ(WebRtcCng_InitEnc(cng_inst_.get(), sample_rate_hz(), + CHECK_EQ(WebRtcCng_InitEnc(cng_inst_.get(), SampleRateHz(), config.sid_frame_interval_ms, config.num_cng_coefficients), 0) @@ -65,15 +65,15 @@ AudioEncoderCng::AudioEncoderCng(const Config& config) AudioEncoderCng::~AudioEncoderCng() { } -int AudioEncoderCng::sample_rate_hz() const { - return speech_encoder_->sample_rate_hz(); +int AudioEncoderCng::SampleRateHz() const { + return speech_encoder_->SampleRateHz(); } -int AudioEncoderCng::rtp_timestamp_rate_hz() const { - return speech_encoder_->rtp_timestamp_rate_hz(); +int AudioEncoderCng::RtpTimestampRateHz() const { + return speech_encoder_->RtpTimestampRateHz(); } -int AudioEncoderCng::num_channels() const { +int AudioEncoderCng::NumChannels() const { return 1; } @@ -105,7 +105,7 @@ bool AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, return false; } info->encoded_bytes = 0; - const int num_samples = sample_rate_hz() / 100 * num_channels(); + const int num_samples = SampleRateHz() / 100 * NumChannels(); if (speech_buffer_.empty()) { CHECK_EQ(frames_in_buffer_, 0); first_timestamp_in_buffer_ = rtp_timestamp; @@ -119,7 +119,7 @@ bool AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, } CHECK_LE(frames_in_buffer_, 6) << "Frame size cannot be larger than 60 ms when using VAD/CNG."; - const size_t samples_per_10ms_frame = 10 * sample_rate_hz() / 1000; + const size_t samples_per_10ms_frame = 10 * SampleRateHz() / 1000; CHECK_EQ(speech_buffer_.size(), static_cast(frames_in_buffer_) * samples_per_10ms_frame); @@ -139,12 +139,12 @@ bool AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, // block. Vad::Activity activity = vad_->VoiceActivity( &speech_buffer_[0], samples_per_10ms_frame * blocks_in_first_vad_call, - sample_rate_hz()); + SampleRateHz()); if (activity == Vad::kPassive && blocks_in_second_vad_call > 0) { // Only check the second block if the first was passive. activity = vad_->VoiceActivity( &speech_buffer_[samples_per_10ms_frame * blocks_in_first_vad_call], - samples_per_10ms_frame * blocks_in_second_vad_call, sample_rate_hz()); + samples_per_10ms_frame * blocks_in_second_vad_call, SampleRateHz()); } DCHECK_NE(activity, Vad::kError); @@ -177,7 +177,7 @@ bool AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, bool AudioEncoderCng::EncodePassive(uint8_t* encoded, size_t* encoded_bytes) { bool force_sid = last_frame_active_; bool output_produced = false; - const size_t samples_per_10ms_frame = 10 * sample_rate_hz() / 1000; + const size_t samples_per_10ms_frame = 10 * SampleRateHz() / 1000; for (int i = 0; i < frames_in_buffer_; ++i) { int16_t encoded_bytes_tmp = 0; if (WebRtcCng_Encode(cng_inst_.get(), @@ -198,7 +198,7 @@ bool AudioEncoderCng::EncodePassive(uint8_t* encoded, size_t* encoded_bytes) { bool AudioEncoderCng::EncodeActive(size_t max_encoded_bytes, uint8_t* encoded, EncodedInfo* info) { - const size_t samples_per_10ms_frame = 10 * sample_rate_hz() / 1000; + const size_t samples_per_10ms_frame = 10 * SampleRateHz() / 1000; for (int i = 0; i < frames_in_buffer_; ++i) { if (!speech_encoder_->Encode(first_timestamp_in_buffer_, &speech_buffer_[i * samples_per_10ms_frame], 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 4338c56e5..2b2c047b6 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 @@ -39,7 +39,7 @@ class AudioEncoderCngTest : public ::testing::Test { memset(encoded_, 0, kMaxEncodedBytes); memset(audio_, 0, kMaxNumSamples * 2); config_.speech_encoder = &mock_encoder_; - EXPECT_CALL(mock_encoder_, num_channels()).WillRepeatedly(Return(1)); + EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1)); // Let the AudioEncoderCng object use a MockVad instead of its internally // created Vad object. config_.vad = mock_vad_; @@ -60,7 +60,7 @@ class AudioEncoderCngTest : public ::testing::Test { // is called, thus we cannot use the values until now. num_audio_samples_10ms_ = 10 * sample_rate_hz_ / 1000; ASSERT_LE(num_audio_samples_10ms_, kMaxNumSamples); - EXPECT_CALL(mock_encoder_, sample_rate_hz()) + EXPECT_CALL(mock_encoder_, SampleRateHz()) .WillRepeatedly(Return(sample_rate_hz_)); // Max10MsFramesInAPacket() is just used to verify that the SID frame period // is not too small. The return value does not matter that much, as long as @@ -443,7 +443,7 @@ TEST_F(AudioEncoderCngDeathTest, NullSpeechEncoder) { } TEST_F(AudioEncoderCngDeathTest, Stereo) { - EXPECT_CALL(mock_encoder_, num_channels()).WillRepeatedly(Return(2)); + EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(2)); EXPECT_DEATH(CreateCng(), "Invalid configuration"); config_.num_channels = 2; EXPECT_DEATH(CreateCng(), "Invalid configuration"); diff --git a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h index a9600044a..55b3db66e 100644 --- a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h +++ b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h @@ -46,9 +46,9 @@ class AudioEncoderCng final : public AudioEncoder { virtual ~AudioEncoderCng(); - virtual int sample_rate_hz() const OVERRIDE; - virtual int num_channels() const OVERRIDE; - int rtp_timestamp_rate_hz() const override; + virtual int SampleRateHz() const OVERRIDE; + virtual int NumChannels() const OVERRIDE; + int RtpTimestampRateHz() const override; virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Max10MsFramesInAPacket() const OVERRIDE; void SetTargetBitrate(int bits_per_second) override; 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 3dd880021..ceef06dc6 100644 --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc @@ -46,10 +46,10 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) AudioEncoderPcm::~AudioEncoderPcm() { } -int AudioEncoderPcm::sample_rate_hz() const { +int AudioEncoderPcm::SampleRateHz() const { return sample_rate_hz_; } -int AudioEncoderPcm::num_channels() const { +int AudioEncoderPcm::NumChannels() const { return num_channels_; } int AudioEncoderPcm::Num10MsFramesInNextPacket() const { @@ -65,7 +65,7 @@ bool AudioEncoderPcm::EncodeInternal(uint32_t rtp_timestamp, size_t max_encoded_bytes, uint8_t* encoded, EncodedInfo* info) { - const int num_samples = sample_rate_hz() / 100 * num_channels(); + const int num_samples = SampleRateHz() / 100 * NumChannels(); if (speech_buffer_.empty()) { first_timestamp_in_buffer_ = rtp_timestamp; } diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h index 83e4aea43..9365c43d0 100644 --- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h +++ b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h @@ -32,8 +32,8 @@ class AudioEncoderPcm : public AudioEncoder { virtual ~AudioEncoderPcm(); - virtual int sample_rate_hz() const OVERRIDE; - virtual int num_channels() const OVERRIDE; + virtual int SampleRateHz() const OVERRIDE; + virtual int NumChannels() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Max10MsFramesInAPacket() const OVERRIDE; 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 08f775371..bd1691fec 100644 --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc @@ -51,15 +51,15 @@ AudioEncoderG722::AudioEncoderG722(const Config& config) AudioEncoderG722::~AudioEncoderG722() {} -int AudioEncoderG722::sample_rate_hz() const { +int AudioEncoderG722::SampleRateHz() const { return kSampleRateHz; } -int AudioEncoderG722::rtp_timestamp_rate_hz() const { +int AudioEncoderG722::RtpTimestampRateHz() const { // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz // codec. return kSampleRateHz / 2; } -int AudioEncoderG722::num_channels() const { +int AudioEncoderG722::NumChannels() const { return num_channels_; } int AudioEncoderG722::Num10MsFramesInNextPacket() const { diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h index 6202f1f81..c314af370 100644 --- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h +++ b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h @@ -30,9 +30,9 @@ class AudioEncoderG722 : public AudioEncoder { explicit AudioEncoderG722(const Config& config); virtual ~AudioEncoderG722(); - virtual int sample_rate_hz() const OVERRIDE; - int rtp_timestamp_rate_hz() const override; - virtual int num_channels() const OVERRIDE; + virtual int SampleRateHz() const OVERRIDE; + int RtpTimestampRateHz() const override; + virtual int NumChannels() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Max10MsFramesInAPacket() const OVERRIDE; 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 da72d678a..a279875b9 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc @@ -43,10 +43,10 @@ AudioEncoderIlbc::~AudioEncoderIlbc() { CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_)); } -int AudioEncoderIlbc::sample_rate_hz() const { +int AudioEncoderIlbc::SampleRateHz() const { return kSampleRateHz; } -int AudioEncoderIlbc::num_channels() const { +int AudioEncoderIlbc::NumChannels() const { return 1; } int AudioEncoderIlbc::Num10MsFramesInNextPacket() const { diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h index 7c2904dfc..7e233bf05 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h +++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h @@ -31,8 +31,8 @@ class AudioEncoderIlbc : public AudioEncoder { explicit AudioEncoderIlbc(const Config& config); virtual ~AudioEncoderIlbc(); - virtual int sample_rate_hz() const OVERRIDE; - virtual int num_channels() const OVERRIDE; + virtual int SampleRateHz() const OVERRIDE; + virtual int NumChannels() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Max10MsFramesInAPacket() const OVERRIDE; 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 65a120449..994d21b44 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 @@ -69,8 +69,8 @@ class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder { void UpdateDecoderSampleRate(int sample_rate_hz); // AudioEncoder public methods. - virtual int sample_rate_hz() const OVERRIDE; - virtual int num_channels() const OVERRIDE; + virtual int SampleRateHz() const OVERRIDE; + virtual int NumChannels() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Max10MsFramesInAPacket() const OVERRIDE; 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 39daa00a8..095bb7b31 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 @@ -166,13 +166,13 @@ void AudioEncoderDecoderIsacT::UpdateDecoderSampleRate(int sample_rate_hz) { } template -int AudioEncoderDecoderIsacT::sample_rate_hz() const { +int AudioEncoderDecoderIsacT::SampleRateHz() const { CriticalSectionScoped cs(state_lock_.get()); return T::EncSampRate(isac_state_); } template -int AudioEncoderDecoderIsacT::num_channels() const { +int AudioEncoderDecoderIsacT::NumChannels() const { return 1; } @@ -181,7 +181,7 @@ int AudioEncoderDecoderIsacT::Num10MsFramesInNextPacket() const { CriticalSectionScoped cs(state_lock_.get()); const int samples_in_next_packet = T::GetNewFrameLen(isac_state_); return rtc::CheckedDivExact(samples_in_next_packet, - rtc::CheckedDivExact(sample_rate_hz(), 100)); + rtc::CheckedDivExact(SampleRateHz(), 100)); } template 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 758625e6a..e424bc6c2 100644 --- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h @@ -21,8 +21,8 @@ class MockAudioEncoder : public AudioEncoder { public: virtual ~MockAudioEncoder() { Die(); } MOCK_METHOD0(Die, void()); - MOCK_CONST_METHOD0(sample_rate_hz, int()); - MOCK_CONST_METHOD0(num_channels, int()); + MOCK_CONST_METHOD0(SampleRateHz, int()); + MOCK_CONST_METHOD0(NumChannels, int()); MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, int()); MOCK_CONST_METHOD0(Max10MsFramesInAPacket, int()); MOCK_METHOD1(SetTargetBitrate, void(int)); 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 693efa5d1..4df92fd30 100644 --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -96,11 +96,11 @@ AudioEncoderOpus::~AudioEncoderOpus() { CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_)); } -int AudioEncoderOpus::sample_rate_hz() const { +int AudioEncoderOpus::SampleRateHz() const { return kSampleRateHz; } -int AudioEncoderOpus::num_channels() const { +int AudioEncoderOpus::NumChannels() const { return num_channels_; } diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h index b615f81b8..245e33430 100644 --- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h +++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h @@ -44,8 +44,8 @@ class AudioEncoderOpus final : public AudioEncoder { explicit AudioEncoderOpus(const Config& config); virtual ~AudioEncoderOpus() OVERRIDE; - virtual int sample_rate_hz() const OVERRIDE; - virtual int num_channels() const OVERRIDE; + virtual int SampleRateHz() const OVERRIDE; + virtual int NumChannels() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Max10MsFramesInAPacket() const OVERRIDE; void SetTargetBitrate(int bits_per_second) override; 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 2b0fb9130..5a256fbe6 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 @@ -26,16 +26,16 @@ AudioEncoderCopyRed::AudioEncoderCopyRed(const Config& config) AudioEncoderCopyRed::~AudioEncoderCopyRed() { } -int AudioEncoderCopyRed::sample_rate_hz() const { - return speech_encoder_->sample_rate_hz(); +int AudioEncoderCopyRed::SampleRateHz() const { + return speech_encoder_->SampleRateHz(); } -int AudioEncoderCopyRed::rtp_timestamp_rate_hz() const { - return speech_encoder_->rtp_timestamp_rate_hz(); +int AudioEncoderCopyRed::RtpTimestampRateHz() const { + return speech_encoder_->RtpTimestampRateHz(); } -int AudioEncoderCopyRed::num_channels() const { - return speech_encoder_->num_channels(); +int AudioEncoderCopyRed::NumChannels() const { + return speech_encoder_->NumChannels(); } int AudioEncoderCopyRed::Num10MsFramesInNextPacket() const { @@ -62,7 +62,7 @@ bool AudioEncoderCopyRed::EncodeInternal(uint32_t rtp_timestamp, uint8_t* encoded, EncodedInfo* info) { if (!speech_encoder_->Encode(rtp_timestamp, audio, - static_cast(sample_rate_hz() / 100), + static_cast(SampleRateHz() / 100), max_encoded_bytes, encoded, info)) return false; if (max_encoded_bytes < info->encoded_bytes + secondary_info_.encoded_bytes) 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 eeda94faf..ea8542d76 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 @@ -35,9 +35,9 @@ class AudioEncoderCopyRed : public AudioEncoder { virtual ~AudioEncoderCopyRed(); - virtual int sample_rate_hz() const OVERRIDE; - int rtp_timestamp_rate_hz() const override; - virtual int num_channels() const OVERRIDE; + virtual int SampleRateHz() const OVERRIDE; + int RtpTimestampRateHz() const override; + virtual int NumChannels() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Max10MsFramesInAPacket() const OVERRIDE; void SetTargetBitrate(int bits_per_second) override; 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 de1339dbb..5373db4a3 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 @@ -41,8 +41,8 @@ class AudioEncoderCopyRedTest : public ::testing::Test { red_.reset(new AudioEncoderCopyRed(config)); memset(encoded_, 0, sizeof(encoded_)); memset(audio_, 0, sizeof(audio_)); - EXPECT_CALL(mock_encoder_, num_channels()).WillRepeatedly(Return(1)); - EXPECT_CALL(mock_encoder_, sample_rate_hz()) + EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1)); + EXPECT_CALL(mock_encoder_, SampleRateHz()) .WillRepeatedly(Return(sample_rate_hz_)); } @@ -103,13 +103,13 @@ TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) { } TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) { - EXPECT_CALL(mock_encoder_, sample_rate_hz()).WillOnce(Return(17)); - EXPECT_EQ(17, red_->sample_rate_hz()); + EXPECT_CALL(mock_encoder_, SampleRateHz()).WillOnce(Return(17)); + EXPECT_EQ(17, red_->SampleRateHz()); } TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) { - EXPECT_CALL(mock_encoder_, num_channels()).WillOnce(Return(17)); - EXPECT_EQ(17, red_->num_channels()); + EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17)); + EXPECT_EQ(17, red_->NumChannels()); } TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) { diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc index 861401588..24fdca31c 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc +++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc @@ -1220,7 +1220,7 @@ int16_t ACMGenericCodecWrapper::Encode( WriteLockScoped wl(codec_wrapper_lock_); CHECK(!input_.empty()); CHECK(encoder_->Encode(rtp_timestamp_, &input_[0], - input_.size() / encoder_->num_channels(), + input_.size() / encoder_->NumChannels(), 2 * MAX_PAYLOAD_SIZE_BYTE, bitstream, encoded_info)); input_.clear(); *bitstream_len_byte = static_cast(encoded_info->encoded_bytes); @@ -1433,7 +1433,7 @@ void ACMGenericCodecWrapper::ResetAudioEncoder() { // Attach CNG if needed. // Reverse-lookup from sample rate to complete key-value pair. auto pt_iter = - FindSampleRateInMap(&cng_pt_, audio_encoder_->sample_rate_hz()); + FindSampleRateInMap(&cng_pt_, audio_encoder_->SampleRateHz()); if (acm_codec_params_.enable_dtx && pt_iter != cng_pt_.end()) { AudioEncoderCng::Config config; config.num_channels = acm_codec_params_.codec_inst.channels; @@ -1475,8 +1475,8 @@ int32_t ACMGenericCodecWrapper::Add10MsData(const uint32_t timestamp, const uint8_t audio_channel) { WriteLockScoped wl(codec_wrapper_lock_); CHECK(input_.empty()); - CHECK_EQ(length_per_channel, encoder_->sample_rate_hz() / 100); - for (int i = 0; i < length_per_channel * encoder_->num_channels(); ++i) { + CHECK_EQ(length_per_channel, encoder_->SampleRateHz() / 100); + for (int i = 0; i < length_per_channel * encoder_->NumChannels(); ++i) { input_.push_back(data[i]); } rtp_timestamp_ = first_frame_ @@ -1485,13 +1485,13 @@ int32_t ACMGenericCodecWrapper::Add10MsData(const uint32_t timestamp, rtc::CheckedDivExact( timestamp - last_timestamp_, static_cast(rtc::CheckedDivExact( - audio_encoder_->sample_rate_hz(), - audio_encoder_->rtp_timestamp_rate_hz()))); + audio_encoder_->SampleRateHz(), + audio_encoder_->RtpTimestampRateHz()))); last_timestamp_ = timestamp; last_rtp_timestamp_ = rtp_timestamp_; first_frame_ = false; - CHECK_EQ(audio_channel, encoder_->num_channels()); + CHECK_EQ(audio_channel, encoder_->NumChannels()); return 0; } diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc index bcbc4c29e..95805d30d 100644 --- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -105,7 +105,7 @@ class AudioDecoderTest : public ::testing::Test { virtual void SetUp() { if (audio_encoder_) - codec_input_rate_hz_ = audio_encoder_->sample_rate_hz(); + codec_input_rate_hz_ = audio_encoder_->SampleRateHz(); // Create arrays. ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0"; // Longest encoded data is produced by PCM16b with 2 bytes per sample. @@ -136,7 +136,7 @@ class AudioDecoderTest : public ::testing::Test { size_t input_len_samples, uint8_t* output) { encoded_info_.encoded_bytes = 0; - const size_t samples_per_10ms = audio_encoder_->sample_rate_hz() / 100; + const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100; CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(), input_len_samples); scoped_ptr interleaved_input( @@ -151,7 +151,7 @@ class AudioDecoderTest : public ::testing::Test { interleaved_input.get()); EXPECT_TRUE(audio_encoder_->Encode( - 0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100, + 0, interleaved_input.get(), audio_encoder_->SampleRateHz() / 100, data_length_ * 2, output, &encoded_info_)); } EXPECT_EQ(payload_type_, encoded_info_.payload_type);