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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
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16
modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -13,6 +13,7 @@
#include <vector>
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_receiver_audio.h"
@ -70,10 +71,8 @@ class VerifyingAudioReceiver : public RtpData {
class RTPCallback : public NullRtpFeedback {
public:
int32_t OnInitializeDecoder(int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int frequency,
size_t channels,
int32_t OnInitializeDecoder(int payload_type,
const SdpAudioFormat& audio_format,
uint32_t rate) override {
EXPECT_EQ(0u, rate) << "The rate should be zero";
return 0;
@ -129,9 +128,11 @@ class RtpRtcpAudioTest : public ::testing::Test {
void RegisterPayload(const CodecInst& codec) {
EXPECT_EQ(0, module1->RegisterSendPayload(codec));
EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(codec));
EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(codec.pltype,
CodecInstToSdp(codec)));
EXPECT_EQ(0, module2->RegisterSendPayload(codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(codec.pltype,
CodecInstToSdp(codec)));
}
VerifyingAudioReceiver data_receiver1;
@ -210,7 +211,8 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
memcpy(voice_codec.plname, "telephone-event", 16);
EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(voice_codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
voice_codec.pltype, CodecInstToSdp(voice_codec)));
// Start DTMF test.
int timeStamp = 160;

7
modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
View File

@ -13,6 +13,7 @@
#include <vector>
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@ -133,9 +134,11 @@ class RtpRtcpRtcpTest : public ::testing::Test {
memcpy(voice_codec.plname, "PCMU", 5);
EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(voice_codec));
EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
voice_codec.pltype, CodecInstToSdp(voice_codec)));
EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(voice_codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
voice_codec.pltype, CodecInstToSdp(voice_codec)));
// We need to send one RTP packet to get the RTCP packet to be accepted by
// the receiving module.