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/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index d06cc07..3253bbd 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -125,7 +125,7 @@
void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string out_file_name, int channels) {
- struct CodecInst recvCodec;
+ struct CodecInst recvCodec = CodecInst();
int noOfCodecs;
EXPECT_EQ(0, acm->InitializeReceiver());
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.cc b/webrtc/modules/audio_coding/main/test/RTPFile.cc
index 50aee31..b886bde 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.cc
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.cc
@@ -234,10 +234,10 @@
return 0;
}
if (payloadSize < (lengthBytes - 20)) {
- return -1;
+ return 0;
}
if (lengthBytes < 20) {
- return -1;
+ return 0;
}
lengthBytes -= 20;
EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
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
-
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
index 10d82ae..73eac47 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
@@ -73,8 +73,8 @@
PCMFile infile_a_;
PCMFile outfile_b_;
int test_count_;
- uint16_t packet_size_samples_;
- uint16_t packet_size_bytes_;
+ int packet_size_samples_;
+ int packet_size_bytes_;
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index 00c3631..b9677e3 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -75,7 +75,7 @@
rtp_info);
if (frame_type != kAudioFrameCN) {
- payload_size_ = payload_size;
+ payload_size_ = static_cast<int>(payload_size);
} else {
payload_size_ = -1;
}
@@ -88,7 +88,7 @@
}
uint16_t TestPackStereo::payload_size() {
- return payload_size_;
+ return static_cast<uint16_t>(payload_size_);
}
uint32_t TestPackStereo::timestamp_diff() {
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/main/test/TestStereo.h
index 03f8041..9cb70e9 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.h
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.h
@@ -52,7 +52,7 @@
uint32_t timestamp_diff_;
uint32_t last_in_timestamp_;
uint64_t total_bytes_;
- uint16_t payload_size_;
+ int payload_size_;
StereoMonoMode codec_mode_;
// Simulate packet losses
bool lost_packet_;