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
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index ae1bce1..ae82509 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -26,15 +26,15 @@
uint8_t* encoded,
EncodedInfo* info) {
CHECK_EQ(num_samples_per_channel,
- static_cast<size_t>(sample_rate_hz() / 100));
+ static_cast<size_t>(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 91fff13..c02c3ef 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -70,12 +70,12 @@
// 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 c85f604..bb53e8c 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 @@
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 @@
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() {
}
-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 @@
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 @@
}
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<size_t>(frames_in_buffer_) * samples_per_10ms_frame);
@@ -139,12 +139,12 @@
// 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::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::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 4338c56..2b2c047 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 @@
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 @@
// 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, 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 a960004..55b3db6 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 @@
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 3dd8800..ceef06d 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() {
}
-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 @@
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 83e4aea..9365c43 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 @@
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 08f7753..bd1691f 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() {}
-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 6202f1f..c314af3 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 @@
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 da72d67..a279875 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 @@
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 7c2904d..7e233bf 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 @@
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 65a1204..994d21b 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 @@
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 39daa00..095bb7b 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 @@
}
template <typename T>
-int AudioEncoderDecoderIsacT<T>::sample_rate_hz() const {
+int AudioEncoderDecoderIsacT<T>::SampleRateHz() const {
CriticalSectionScoped cs(state_lock_.get());
return T::EncSampRate(isac_state_);
}
template <typename T>
-int AudioEncoderDecoderIsacT<T>::num_channels() const {
+int AudioEncoderDecoderIsacT<T>::NumChannels() const {
return 1;
}
@@ -181,7 +181,7 @@
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 <typename T>
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 758625e..e424bc6 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 @@
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 693efa5..4df92fd 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 @@
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 b615f81..245e334 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 @@
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 2b0fb91..5a256fb 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() {
}
-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 @@
uint8_t* encoded,
EncodedInfo* info) {
if (!speech_encoder_->Encode(rtp_timestamp, audio,
- static_cast<size_t>(sample_rate_hz() / 100),
+ static_cast<size_t>(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 eeda94f..ea8542d 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 @@
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 de1339d..5373db4 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 @@
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, 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 8614015..24fdca3 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 @@
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<int16_t>(encoded_info->encoded_bytes);
@@ -1433,7 +1433,7 @@
// 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 @@
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 @@
rtc::CheckedDivExact(
timestamp - last_timestamp_,
static_cast<uint32_t>(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 bcbc4c2..95805d3 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 @@
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 @@
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<int16_t[]> interleaved_input(
@@ -151,7 +151,7 @@
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);
