Donate to e Foundation | Murena handsets with /e/OS | Own a part of Murena! Learn more

Commit f6936a3a authored by Chia-chi Yeh's avatar Chia-chi Yeh Committed by Android (Google) Code Review
Browse files

Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread

parents 4d1e0121 557b04de

voip/jni/rtp/AudioGroup.cpp

+2 −2
Original line number Diff line number Diff line
@@ -588,7 +588,7 @@ bool AudioGroup::set(int sampleRate, int sampleCount) 
    // Give device socket a reasonable timeout and buffer size.

    timeval tv;

    tv.tv_sec = 0;

    tv.tv_usec = 1000 * sampleCount / sampleRate * 1000;

    tv.tv_usec = 1000 * sampleCount / sampleRate * 500;

    if (setsockopt(pair[0], SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv)) ||

        setsockopt(pair[0], SOL_SOCKET, SO_RCVBUF, &output, sizeof(output)) ||

        setsockopt(pair[1], SOL_SOCKET, SO_SNDBUF, &output, sizeof(output))) {
@@ -793,7 +793,7 @@ bool AudioGroup::deviceLoop() 


            status_t status = mRecord.obtainBuffer(&buffer, 1);

            if (status == NO_ERROR) {

                int count = (buffer.frameCount < toRead) ?

                int count = ((int)buffer.frameCount < toRead) ?

                        buffer.frameCount : toRead;

                memcpy(&input[mSampleCount - toRead], buffer.i8, count * 2);

                toRead -= count;