Index: webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc |
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc |
index 28a7b1090af9c1a1ed4c90e0732f3361e11be26b..b6d8a3a1db914eeacb08f8229f3e52d53bf8bab9 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc |
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc |
@@ -19,7 +19,7 @@ |
namespace { |
class OpusFrame : public AudioDecoder::EncodedAudioFrame { |
public: |
- OpusFrame(AudioDecoderOpusImpl* decoder, |
+ OpusFrame(AudioDecoderOpus* decoder, |
rtc::Buffer&& payload, |
bool is_primary_payload) |
: decoder_(decoder), |
@@ -57,25 +57,25 @@ |
} |
private: |
- AudioDecoderOpusImpl* const decoder_; |
+ AudioDecoderOpus* const decoder_; |
const rtc::Buffer payload_; |
const bool is_primary_payload_; |
}; |
} // namespace |
-AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels) |
+AudioDecoderOpus::AudioDecoderOpus(size_t num_channels) |
: channels_(num_channels) { |
RTC_DCHECK(num_channels == 1 || num_channels == 2); |
WebRtcOpus_DecoderCreate(&dec_state_, channels_); |
WebRtcOpus_DecoderInit(dec_state_); |
} |
-AudioDecoderOpusImpl::~AudioDecoderOpusImpl() { |
+AudioDecoderOpus::~AudioDecoderOpus() { |
WebRtcOpus_DecoderFree(dec_state_); |
} |
-std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload( |
+std::vector<AudioDecoder::ParseResult> AudioDecoderOpus::ParsePayload( |
rtc::Buffer&& payload, |
uint32_t timestamp) { |
std::vector<ParseResult> results; |
@@ -95,11 +95,11 @@ |
return results; |
} |
-int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded, |
- size_t encoded_len, |
- int sample_rate_hz, |
- int16_t* decoded, |
- SpeechType* speech_type) { |
+int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded, |
+ size_t encoded_len, |
+ int sample_rate_hz, |
+ int16_t* decoded, |
+ SpeechType* speech_type) { |
RTC_DCHECK_EQ(sample_rate_hz, 48000); |
int16_t temp_type = 1; // Default is speech. |
int ret = |
@@ -110,11 +110,11 @@ |
return ret; |
} |
-int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded, |
- size_t encoded_len, |
- int sample_rate_hz, |
- int16_t* decoded, |
- SpeechType* speech_type) { |
+int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded, |
+ size_t encoded_len, |
+ int sample_rate_hz, |
+ int16_t* decoded, |
+ SpeechType* speech_type) { |
if (!PacketHasFec(encoded, encoded_len)) { |
// This packet is a RED packet. |
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, |
@@ -131,17 +131,17 @@ |
return ret; |
} |
-void AudioDecoderOpusImpl::Reset() { |
+void AudioDecoderOpus::Reset() { |
WebRtcOpus_DecoderInit(dec_state_); |
} |
-int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded, |
- size_t encoded_len) const { |
+int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, |
+ size_t encoded_len) const { |
return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len); |
} |
-int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded, |
- size_t encoded_len) const { |
+int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded, |
+ size_t encoded_len) const { |
if (!PacketHasFec(encoded, encoded_len)) { |
// This packet is a RED packet. |
return PacketDuration(encoded, encoded_len); |
@@ -150,18 +150,18 @@ |
return WebRtcOpus_FecDurationEst(encoded, encoded_len); |
} |
-bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded, |
- size_t encoded_len) const { |
+bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded, |
+ size_t encoded_len) const { |
int fec; |
fec = WebRtcOpus_PacketHasFec(encoded, encoded_len); |
return (fec == 1); |
} |
-int AudioDecoderOpusImpl::SampleRateHz() const { |
+int AudioDecoderOpus::SampleRateHz() const { |
return 48000; |
} |
-size_t AudioDecoderOpusImpl::Channels() const { |
+size_t AudioDecoderOpus::Channels() const { |
return channels_; |
} |