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Add new methods to AudioEncoder interface

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The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()

Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.

BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34049004

Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2015-01-27 18:24:45 +00:00
parent 5614cf16e7
commit 478cedc055
19 changed files with 175 additions and 30 deletions
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2
webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
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@ -25,6 +25,8 @@ class MockAudioEncoder : public AudioEncoder {
MOCK_CONST_METHOD0(num_channels, int());
MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, int());
MOCK_CONST_METHOD0(Max10MsFramesInAPacket, int());
MOCK_METHOD1(SetTargetBitrate, void(int));
MOCK_METHOD1(SetProjectedPacketLossRate, void(double));
// Note, we explicitly chose not to create a mock for the Encode method.
MOCK_METHOD5(EncodeInternal,
bool(uint32_t timestamp,