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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

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
View File

@ -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
View File

@ -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_;