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Setting marker bit on DTMF correctly

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BUG=1157
R=braveyao@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7037 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2014-09-03 11:46:54 +00:00
parent 74cf916924
commit 0a214ffa8a
2 changed files with 63 additions and 6 deletions
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14
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -318,13 +318,15 @@ int32_t RTPSenderAudio::SendAudio(
static_cast<uint16_t>(dtmfDurationSamples),
false);
} else {
// set markerBit on the first packet in the burst
if (SendTelephoneEventPacket(
ended,
_dtmfTimestamp,
static_cast<uint16_t>(dtmfDurationSamples),
!_dtmfEventFirstPacketSent) != 0) {
return -1;
}
_dtmfEventFirstPacketSent = true;
return SendTelephoneEventPacket(
ended,
_dtmfTimestamp,
static_cast<uint16_t>(dtmfDurationSamples),
!_dtmfEventFirstPacketSent);
return 0;
}
}
return 0;

55
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -1073,6 +1073,61 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
sizeof(extension)));
}
// As RFC4733, named telephone events are carried as part of the audio stream
// and must use the same sequence number and timestamp base as the regular
// audio channel.
// This test checks the marker bit for the first packet and the consequent
// packets of the same telephone event. Since it is specifically for DTMF
// events, ignoring audio packets and sending kFrameEmpty instead of those.
TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event";
uint8_t payload_type = 126;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 0,
0, 0));
// For Telephone events, payload is not added to the registered payload list,
// it will register only the payload used for audio stream.
// Registering the payload again for audio stream with different payload name.
strcpy(payload_name, "payload_name");
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000,
1, 0));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
// DTMF event key=9, duration=500 and attenuationdB=10
rtp_sender_->SendTelephoneEvent(9, 500, 10);
// During start, it takes the starting timestamp as last sent timestamp.
// The duration is calculated as the difference of current and last sent
// timestamp. So for first call it will skip since the duration is zero.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
capture_time_ms,
0, NULL, 0,
NULL));
// DTMF Sample Length is (Frequency/1000) * Duration.
// So in this case, it is (8000/1000) * 500 = 4000.
// Sending it as two packets.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
capture_time_ms+2000,
0, NULL, 0,
NULL));
scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
ASSERT_TRUE(rtp_parser.get() != NULL);
webrtc::RTPHeader rtp_header;
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
transport_.last_sent_packet_len_,
&rtp_header));
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(rtp_header.markerBit);
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
capture_time_ms+4000,
0, NULL, 0,
NULL));
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
transport_.last_sent_packet_len_,
&rtp_header));
// Marker Bit should be set to 0 for rest of the packets.
EXPECT_FALSE(rtp_header.markerBit);
}
TEST_F(RtpSenderTest, BytesReportedCorrectly) {
const char* kPayloadName = "GENERIC";
const uint8_t kPayloadType = 127;