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Resurrected test_api_audio.cc

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I'll be doing some changes to code it tests (rtp_receiver_audio,
specifically) and want to make sure there are tests in place before I
touch anything.

Fixed test_api_audio not properly checking payload data. Required a
fix to LoopBackTransport in test_api to as to act like the regular
audio and video parts of WebRTC and separate payload from header data.

Also added a test for CNG and cleaned up constants.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2378403004
Cr-Commit-Position: refs/heads/master@{#14529}
This commit is contained in:
ossu
2016-10-05 07:51:44 -07:00
committed by Commit bot
parent b1fff92644
commit b2d1e0d1da
2 changed files with 148 additions and 120 deletions
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9
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
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@ -14,6 +14,7 @@
#include <memory>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/rate_limiter.h"
#include "webrtc/test/null_transport.h"
@ -44,7 +45,7 @@ bool LoopBackTransport::SendRtp(const uint8_t* data,
}
RTPHeader header;
std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
if (!parser->Parse(data, len, &header)) {
return false;
}
PayloadUnion payload_specific;
@ -52,9 +53,11 @@ bool LoopBackTransport::SendRtp(const uint8_t* data,
&payload_specific)) {
return false;
}
const uint8_t* payload = data + header.headerLength;
RTC_CHECK_GE(len, header.headerLength);
const size_t payload_length = len - header.headerLength;
receive_statistics_->IncomingPacket(header, len, false);
if (!rtp_receiver_->IncomingRtpPacket(header,
static_cast<const uint8_t*>(data), len,
if (!rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
payload_specific, true)) {
return false;
}