They are invoked with the maximum size of the data to be added, and a
callable that generates that data, like this:
buffer.AppendData(10, [] (rtc::ArrayView<uint8_t> av) {
for (uint8_t i = 0; i != 5; ++i)
av[i] = i;
return 5;
});
The callable returns the number of bytes actually written, and the
final Buffer size will be adjusted accordingly. SetData and AppendData
both return the number of bytes added (i.e. the return value of the
callable).
These versions will be useful when converting AudioEncoder::Encode to use Buffer rather than raw pointers.
Also added a few tests for the new functionality.
Review URL: https://codereview.webrtc.org/1717273002
Cr-Commit-Position: refs/heads/master@{#11733}
The current use of rtc::FifoBuffer can lead to reading across DTLS packet
boundaries which could cause packets to not being processed correctly.
This CL introduces the new class rtc::BufferQueue and changes the
StreamInterfaceChannel to use it instead of the rtc::FifoBuffer.
BUG=chromium:447431
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/52509004
Cr-Commit-Position: refs/heads/master@{#9254}