AudioEncoder: change Encode and EncodeInternal return type to void
After code cleanup done on issues: https://webrtc-codereview.appspot.com/34259004/ https://webrtc-codereview.appspot.com/43409004/ https://webrtc-codereview.appspot.com/34309004/ https://webrtc-codereview.appspot.com/34309004/ https://webrtc-codereview.appspot.com/36209004/ https://webrtc-codereview.appspot.com/40899004/ https://webrtc-codereview.appspot.com/39279004/ https://webrtc-codereview.appspot.com/42099005/ and the similar work done for AudioEncoderDecoderIsacT, methods AudioEncoder::Encode and AudioEncoder::EncodeInternal will always succeed. Therefore, there is no need for them to return bool value that represents success or failure. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38279004 Cr-Commit-Position: refs/heads/master@{#8518} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8518 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -56,7 +56,7 @@ int AudioEncoderIlbc::Max10MsFramesInAPacket() const {
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return num_10ms_frames_per_packet_;
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}
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bool AudioEncoderIlbc::EncodeInternal(uint32_t rtp_timestamp,
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void AudioEncoderIlbc::EncodeInternal(uint32_t rtp_timestamp,
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const int16_t* audio,
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size_t max_encoded_bytes,
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uint8_t* encoded,
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@ -93,7 +93,7 @@ bool AudioEncoderIlbc::EncodeInternal(uint32_t rtp_timestamp,
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// for now.
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if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
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info->encoded_bytes = 0;
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return true;
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return;
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}
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// Encode buffered input.
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@ -109,7 +109,6 @@ bool AudioEncoderIlbc::EncodeInternal(uint32_t rtp_timestamp,
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info->encoded_bytes = output_len;
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info->encoded_timestamp = first_timestamp_in_buffer_;
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info->payload_type = payload_type_;
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return true;
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}
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} // namespace webrtc
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@ -37,7 +37,7 @@ class AudioEncoderIlbc : public AudioEncoder {
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virtual int Max10MsFramesInAPacket() const OVERRIDE;
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protected:
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virtual bool EncodeInternal(uint32_t rtp_timestamp,
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virtual void EncodeInternal(uint32_t rtp_timestamp,
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const int16_t* audio,
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size_t max_encoded_bytes,
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uint8_t* encoded,
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