From 05211277798ca4791fbdc508e24d7fd06d5ee6ff Mon Sep 17 00:00:00 2001 From: "kwiberg@webrtc.org" Date: Wed, 18 Feb 2015 12:00:32 +0000 Subject: [PATCH] AudioEncoder: Rename virtual accessors to CamelCase Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz() are simple accessors for almost all implementations of AudioEncoder, they are virtual and not guaranteed to be just simple accessors. Thus, it makes more sense to use the normal CamelCase naming scheme. BUG=4235 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34239004 Cr-Commit-Position: refs/heads/master@{#8407} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../audio_coding/codecs/audio_encoder.cc | 6 ++--- .../audio_coding/codecs/audio_encoder.h | 6 ++--- .../codecs/cng/audio_encoder_cng.cc | 26 +++++++++---------- .../codecs/cng/audio_encoder_cng_unittest.cc | 6 ++--- .../codecs/cng/include/audio_encoder_cng.h | 6 ++--- .../codecs/g711/audio_encoder_pcm.cc | 6 ++--- .../codecs/g711/include/audio_encoder_pcm.h | 4 +-- .../codecs/g722/audio_encoder_g722.cc | 6 ++--- .../codecs/g722/include/audio_encoder_g722.h | 6 ++--- .../codecs/ilbc/audio_encoder_ilbc.cc | 4 +-- .../ilbc/interface/audio_encoder_ilbc.h | 4 +-- .../codecs/isac/audio_encoder_isac_t.h | 4 +-- .../codecs/isac/audio_encoder_isac_t_impl.h | 6 ++--- .../codecs/mock/mock_audio_encoder.h | 4 +-- .../codecs/opus/audio_encoder_opus.cc | 4 +-- .../opus/interface/audio_encoder_opus.h | 4 +-- .../codecs/red/audio_encoder_copy_red.cc | 14 +++++----- .../codecs/red/audio_encoder_copy_red.h | 6 ++--- .../red/audio_encoder_copy_red_unittest.cc | 12 ++++----- .../main/acm2/acm_generic_codec.cc | 14 +++++----- .../neteq/audio_decoder_unittest.cc | 6 ++--- 21 files changed, 77 insertions(+), 77 deletions(-) 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);