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AudioEncoder: Rename virtual accessors to CamelCase

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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
This commit is contained in:
kwiberg@webrtc.org 2015-02-18 12:00:32 +00:00
parent cc483b7379
commit 0521127779
21 changed files with 77 additions and 77 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/audio_coding/codecs/audio_encoder.cc
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@ -26,15 +26,15 @@ bool AudioEncoder::Encode(uint32_t rtp_timestamp,
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

6
webrtc/modules/audio_coding/codecs/audio_encoder.h
View File

@ -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.,

26
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
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@ -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<size_t>(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],

6
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
View File

@ -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");

6
webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
View File

@ -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;

6
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
View File

@ -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;
}

4
webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
View File

@ -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;

6
webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
View File

@ -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 {

6
webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
View File

@ -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;

4
webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
View File

@ -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 {

4
webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
View File

@ -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;

4
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
View File

@ -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;

6
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
View File

@ -166,13 +166,13 @@ void AudioEncoderDecoderIsacT<T>::UpdateDecoderSampleRate(int sample_rate_hz) {
}
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 @@ int AudioEncoderDecoderIsacT<T>::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 <typename T>

4
webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
View File

@ -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));

4
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -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_;
}

4
webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
View File

@ -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;

14
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
View File

@ -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<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)

6
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
View File

@ -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;

12
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
View File

@ -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) {

14
webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
View File

@ -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<int16_t>(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<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;
}

6
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -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<int16_t[]> 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);