This adds a new file, webrtc/modules/audio_coding/neteq/tools/packet_source.cc, so that I'll have somewhere to put the new non-inlined methods.
NOTRY=true
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2290593002
Cr-Commit-Position: refs/heads/master@{#13956}
The return type of PacketSource::NextPacket() is changed from a naked
pointer to an std::uniqe_ptr. The interface contract was and still is
that the ownership is passed from the callee to the caller, but a
unique_ptr makes this explicit.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2005873002
Cr-Commit-Position: refs/heads/master@{#12884}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.
The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
The NextPacket method should now return NULL when the end of the
source was reached. In the RtpFileSource, this means that when
the end of file is reached, NULL is returned. Also, when an RTCP
packet is encountered, the next packet will be read from file
immediately.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
