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Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798

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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
This commit is contained in:
henrike@webrtc.org
2014-08-11 21:06:30 +00:00
parent 820f8e9ca7
commit 6ac22e6b47
27 changed files with 92 additions and 104 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
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@@ -49,7 +49,7 @@ int main(int argc, char* argv[])
int i, errtype, VADusage = 0, packetLossPercent = 0;
int16_t CodingMode;
int32_t bottleneck;
int32_t bottleneck = 0;
int16_t framesize = 30; /* ms */
int cur_framesmpls, err;
@@ -57,7 +57,7 @@ int main(int argc, char* argv[])
double starttime, runtime, length_file;
int16_t stream_len = 0;
int16_t declen, lostFrame = 0, declenTC = 0;
int16_t declen = 0, lostFrame = 0, declenTC = 0;
int16_t shortdata[SWBFRAMESAMPLES_10ms];
int16_t vaddata[SWBFRAMESAMPLES_10ms*3];
@@ -609,8 +609,8 @@ int main(int argc, char* argv[])
cout << "\n" << flush;
length_file = 0;
int16_t bnIdxTC;
int16_t jitterInfoTC;
int16_t bnIdxTC = 0;
int16_t jitterInfoTC = 0;
while (endfile == 0)
{
/* Call init functions at random, fault test number 7 */

2
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
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@@ -74,7 +74,7 @@ int main(int argc, char* argv[])
ISACStruct* ISAC_main_inst;
int16_t stream_len = 0;
int16_t declen;
int16_t declen = 0;
int16_t err;
int16_t cur_framesmpls;
int endfile;

2
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
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@@ -125,7 +125,7 @@ Receiver::Receiver()
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());

4
webrtc/modules/audio_coding/main/test/RTPFile.cc
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@@ -234,10 +234,10 @@ uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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));

14
webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
View File

@@ -710,10 +710,10 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
}
// 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 @@ void TestAllCodecs::Run(TestPack* channel) {
// 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 @@ void TestAllCodecs::Run(TestPack* channel) {
// 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 @@ void TestAllCodecs::DisplaySendReceiveCodec() {
}
} // namespace webrtc

4
webrtc/modules/audio_coding/main/test/TestAllCodecs.h
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@@ -73,8 +73,8 @@ class TestAllCodecs : public ACMTest {
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

4
webrtc/modules/audio_coding/main/test/TestStereo.cc
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@@ -75,7 +75,7 @@ int32_t TestPackStereo::SendData(const FrameType frame_type,
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 @@ int32_t TestPackStereo::SendData(const FrameType frame_type,
}
uint16_t TestPackStereo::payload_size() {
return payload_size_;
return static_cast<uint16_t>(payload_size_);
}
uint32_t TestPackStereo::timestamp_diff() {

2
webrtc/modules/audio_coding/main/test/TestStereo.h
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@@ -52,7 +52,7 @@ class TestPackStereo : public AudioPacketizationCallback {
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_;