Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
index d6c6dc4..10654a7 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -710,10 +710,10 @@
}
// Store the expected packet size in bytes, used to validate the received
- // packet. If variable rate codec (extra_byte == -1), set to -1 (65535).
+ // packet. If variable rate codec (extra_byte == -1), set to -1.
if (extra_byte != -1) {
// Add 0.875 to always round up to a whole byte
- packet_size_bytes_ = static_cast<uint16_t>(static_cast<float>(packet_size
+ packet_size_bytes_ = static_cast<int>(static_cast<float>(packet_size
* rate) / static_cast<float>(sampling_freq_hz * 8) + 0.875)
+ extra_byte;
} else {
@@ -768,8 +768,8 @@
// Verify that the received packet size matches the settings.
receive_size = channel->payload_size();
if (receive_size) {
- if ((receive_size != packet_size_bytes_) &&
- (packet_size_bytes_ < 65535)) {
+ if ((static_cast<int>(receive_size) != packet_size_bytes_) &&
+ (packet_size_bytes_ > -1)) {
error_count++;
}
@@ -777,8 +777,9 @@
// is used to avoid problems when switching codec or frame size in the
// test.
timestamp_diff = channel->timestamp_diff();
- if ((counter > 10) && (timestamp_diff != packet_size_samples_) &&
- (packet_size_samples_ < 65535))
+ if ((counter > 10) &&
+ (static_cast<int>(timestamp_diff) != packet_size_samples_) &&
+ (packet_size_samples_ > -1))
error_count++;
}
@@ -819,4 +820,3 @@
}
} // namespace webrtc
-