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Rename internal AudioEncoder::Encode method to EncodeInternal

Browse Source 
BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-12-03 20:36:03 +00:00
parent d1fac61e8f
commit 8dc21dc238
7 changed files with 48 additions and 48 deletions
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24
webrtc/modules/audio_coding/codecs/audio_encoder.h
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@ -45,12 +45,12 @@ class AudioEncoder {
EncodedInfo* info) { EncodedInfo* info) {
CHECK_EQ(num_samples_per_channel, CHECK_EQ(num_samples_per_channel,
static_cast<size_t>(sample_rate_hz() / 100)); static_cast<size_t>(sample_rate_hz() / 100));
bool ret = Encode(timestamp, bool ret = EncodeInternal(timestamp,
audio, audio,
max_encoded_bytes, max_encoded_bytes,
encoded, encoded,
encoded_bytes, encoded_bytes,
info); info);
CHECK_LE(*encoded_bytes, max_encoded_bytes); CHECK_LE(*encoded_bytes, max_encoded_bytes);
return ret; return ret;
} }
@ -68,12 +68,12 @@ class AudioEncoder {
virtual int Num10MsFramesInNextPacket() const = 0; virtual int Num10MsFramesInNextPacket() const = 0;
protected: protected:
virtual bool Encode(uint32_t timestamp, virtual bool EncodeInternal(uint32_t timestamp,
const int16_t* audio, const int16_t* audio,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
EncodedInfo* info) = 0; EncodedInfo* info) = 0;
}; };
} // namespace webrtc } // namespace webrtc

12
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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@ -53,12 +53,12 @@ int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_; return num_10ms_frames_per_packet_;
} }
bool AudioEncoderPcm::Encode(uint32_t timestamp, bool AudioEncoderPcm::EncodeInternal(uint32_t timestamp,
const int16_t* audio, const int16_t* audio,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
EncodedInfo* info) { EncodedInfo* info) {
const int num_samples = sample_rate_hz() / 100 * num_channels(); const int num_samples = sample_rate_hz() / 100 * num_channels();
if (speech_buffer_.empty()) { if (speech_buffer_.empty()) {
first_timestamp_in_buffer_ = timestamp; first_timestamp_in_buffer_ = timestamp;

12
webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
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@ -39,12 +39,12 @@ class AudioEncoderPcm : public AudioEncoder {
virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE;
protected: protected:
virtual bool Encode(uint32_t timestamp, virtual bool EncodeInternal(uint32_t timestamp,
const int16_t* audio, const int16_t* audio,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
EncodedInfo* info) OVERRIDE; EncodedInfo* info) OVERRIDE;
virtual int16_t EncodeCall(const int16_t* audio, virtual int16_t EncodeCall(const int16_t* audio,
size_t input_len, size_t input_len,

12
webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
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@ -61,12 +61,12 @@ int AudioEncoderG722::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_; return num_10ms_frames_per_packet_;
} }
bool AudioEncoderG722::Encode(uint32_t timestamp, bool AudioEncoderG722::EncodeInternal(uint32_t timestamp,
const int16_t* audio, const int16_t* audio,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
EncodedInfo* info) { EncodedInfo* info) {
const int samples_per_channel = const int samples_per_channel =
kSampleRateHz / 100 * num_10ms_frames_per_packet_; kSampleRateHz / 100 * num_10ms_frames_per_packet_;
CHECK_GE(max_encoded_bytes, CHECK_GE(max_encoded_bytes,

12
webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
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@ -35,12 +35,12 @@ class AudioEncoderG722 : public AudioEncoder {
virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE;
protected: protected:
virtual bool Encode(uint32_t timestamp, virtual bool EncodeInternal(uint32_t timestamp,
const int16_t* audio, const int16_t* audio,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
EncodedInfo* info) OVERRIDE; EncodedInfo* info) OVERRIDE;
private: private:
// The encoder state for one channel. // The encoder state for one channel.

12
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -74,12 +74,12 @@ int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_; return num_10ms_frames_per_packet_;
} }
bool AudioEncoderOpus::Encode(uint32_t timestamp, bool AudioEncoderOpus::EncodeInternal(uint32_t timestamp,
const int16_t* audio, const int16_t* audio,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
EncodedInfo* info) { EncodedInfo* info) {
if (input_buffer_.empty()) if (input_buffer_.empty())
first_timestamp_in_buffer_ = timestamp; first_timestamp_in_buffer_ = timestamp;
input_buffer_.insert(input_buffer_.end(), audio, input_buffer_.insert(input_buffer_.end(), audio,

12
webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
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@ -36,12 +36,12 @@ class AudioEncoderOpus : public AudioEncoder {
virtual int Num10MsFramesInNextPacket() const OVERRIDE; virtual int Num10MsFramesInNextPacket() const OVERRIDE;
protected: protected:
virtual bool Encode(uint32_t timestamp, virtual bool EncodeInternal(uint32_t timestamp,
const int16_t* audio, const int16_t* audio,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
EncodedInfo* info) OVERRIDE; EncodedInfo* info) OVERRIDE;
private: private:
const int num_10ms_frames_per_packet_; const int num_10ms_frames_per_packet_;