Revert 8749 "We changed Encode() and EncodeInternal() return typ..."
The reason is that this cl adds a static initializer so we can't roll webrtc into Chromium.
See audio_encoder.cc and 'sizes' regression here:
http://build.chromium.org/p/chromium/builders/Linux%20x64/builds/186
> We changed Encode() and EncodeInternal() return type from bool to void in this issue:
> 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
TBR=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49449004
Cr-Commit-Position: refs/heads/master@{#8772}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8772 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 6e29f08..20ac8b9 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -54,21 +54,20 @@
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
- // returns additional encoding information.
+ // The encoder produces zero or more bytes of output in |encoded|,
+ // and provides additional encoding information in |info|.
// The caller is responsible for making sure that |max_encoded_bytes| is
// not smaller than the number of bytes actually produced by the encoder.
- EncodedInfo Encode(uint32_t rtp_timestamp,
- const int16_t* audio,
- size_t num_samples_per_channel,
- size_t max_encoded_bytes,
- uint8_t* encoded);
+ 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);
// Return the input sample rate in Hz and the number of input channels.
// These are constants set at instantiation time.
@@ -108,10 +107,11 @@
virtual void SetProjectedPacketLossRate(double fraction) {}
protected:
- virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
- size_t max_encoded_bytes,
- uint8_t* encoded) = 0;
+ virtual void EncodeInternal(uint32_t rtp_timestamp,
+ const int16_t* audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ EncodedInfo* info) = 0;
};
} // namespace webrtc