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RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs

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This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.

BUG=webrtc:8159

Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
This commit is contained in:
Karl Wiberg
2017-10-04 12:38:53 +02:00
committed by Commit Bot
parent 83ccca1864
commit c62f6c7121
22 changed files with 205 additions and 259 deletions
Download Patch File Download Diff File Expand all files Collapse all files

23
modules/rtp_rtcp/source/rtp_receiver_unittest.cc
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@ -51,12 +51,8 @@ class RtpReceiverTest : public ::testing::Test {
&mock_rtp_data_,
nullptr,
&rtp_payload_registry_)) {
CodecInst voice_codec = {};
voice_codec.pltype = kPcmuPayloadType;
voice_codec.plfreq = 8000;
voice_codec.rate = kTestRate;
memcpy(voice_codec.plname, "PCMU", 5);
rtp_receiver_->RegisterReceivePayload(voice_codec);
rtp_receiver_->RegisterReceivePayload(kPcmuPayloadType,
SdpAudioFormat("PCMU", 8000, 1));
}
~RtpReceiverTest() {}
@ -90,7 +86,8 @@ TEST_F(RtpReceiverTest, GetSources) {
header.numCSRCs = 2;
header.arrOfCSRCs[0] = kCsrc1;
header.arrOfCSRCs[1] = kCsrc2;
const PayloadUnion payload_specific{AudioPayload()};
const PayloadUnion payload_specific{
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
@ -140,7 +137,8 @@ TEST_F(RtpReceiverTest, GetSourcesChangeSSRC) {
header.payloadType = kPcmuPayloadType;
header.ssrc = kSsrc1;
header.timestamp = rtp_timestamp(now_ms);
const PayloadUnion payload_specific{AudioPayload()};
const PayloadUnion payload_specific{
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
@ -191,7 +189,8 @@ TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
RTPHeader header;
header.payloadType = kPcmuPayloadType;
header.timestamp = rtp_timestamp(now_ms);
const PayloadUnion payload_specific{AudioPayload()};
const PayloadUnion payload_specific{
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
header.numCSRCs = 1;
size_t kSourceListSize = 20;
@ -265,7 +264,8 @@ TEST_F(RtpReceiverTest, GetSourcesContainsAudioLevelExtension) {
header.timestamp = rtp_timestamp(time1_ms);
header.extension.hasAudioLevel = true;
header.extension.audioLevel = 10;
const PayloadUnion payload_specific{AudioPayload()};
const PayloadUnion payload_specific{
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
@ -317,7 +317,8 @@ TEST_F(RtpReceiverTest,
header.timestamp = rtp_timestamp(time1_ms);
header.extension.hasAudioLevel = true;
header.extension.audioLevel = 10;
const PayloadUnion payload_specific{AudioPayload()};
const PayloadUnion payload_specific{
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));