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We changed Encode() and EncodeInternal() return type from bool to void in this issue:

Browse Source 
https://webrtc-codereview.appspot.com/38279004/
Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43839004

Cr-Commit-Position: refs/heads/master@{#8749}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
jmarusic@webrtc.org
2015-03-17 12:12:17 +00:00
parent 73d763e71f
commit 0cb612b43b
21 changed files with 222 additions and 237 deletions
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26
webrtc/modules/audio_coding/codecs/audio_encoder.h
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@ -54,20 +54,21 @@ class AudioEncoder {
std::vector<EncodedInfoLeaf> redundant;
};
static const EncodedInfo kZeroEncodedBytes;
virtual ~AudioEncoder() {}
// Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 *
// num_channels() samples). Multi-channel audio must be sample-interleaved.
// The encoder produces zero or more bytes of output in |encoded|,
// and provides additional encoding information in |info|.
// The encoder produces zero or more bytes of output in |encoded| and
// returns additional encoding information.
// The caller is responsible for making sure that |max_encoded_bytes| is
// not smaller than the number of bytes actually produced by the encoder.
void Encode(uint32_t rtp_timestamp,
const int16_t* audio,
size_t num_samples_per_channel,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info);
EncodedInfo Encode(uint32_t rtp_timestamp,
const int16_t* audio,
size_t num_samples_per_channel,
size_t max_encoded_bytes,
uint8_t* encoded);
// Return the input sample rate in Hz and the number of input channels.
// These are constants set at instantiation time.
@ -107,11 +108,10 @@ class AudioEncoder {
virtual void SetProjectedPacketLossRate(double fraction) {}
protected:
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) = 0;
virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded) = 0;
};
} // namespace webrtc