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Enable audio streams to send padding.

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Useful if bitrate probing is to be used with audio streams.

BUG=webrtc:7043

Review-Url: https://codereview.webrtc.org/2652893004
Cr-Commit-Position: refs/heads/master@{#16404}
This commit is contained in:
stefan 2017-02-01 09:06:25 -08:00 committed by Commit bot
parent 46fbb7d9d5
commit e35f89a484
2 changed files with 50 additions and 8 deletions
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33
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -36,6 +36,7 @@ namespace webrtc {
namespace {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
constexpr size_t kMaxPaddingLength = 224;
constexpr size_t kMinAudioPaddingLength = 50;
constexpr int kSendSideDelayWindowMs = 1000;
constexpr size_t kRtpHeaderLength = 12;
constexpr uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1.
@ -481,11 +482,21 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
}
size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
// Always send full padding packets. This is accounted for by the
// RtpPacketSender, which will make sure we don't send too much padding even
// if a single packet is larger than requested.
size_t padding_bytes_in_packet =
std::min(MaxPayloadSize(), kMaxPaddingLength);
size_t padding_bytes_in_packet;
if (audio_configured_) {
// Allow smaller padding packets for audio.
padding_bytes_in_packet = std::max(std::min(bytes, MaxPayloadSize()),
kMinAudioPaddingLength);
if (padding_bytes_in_packet > kMaxPaddingLength)
padding_bytes_in_packet = kMaxPaddingLength;
} else {
// Always send full padding packets. This is accounted for by the
// RtpPacketSender, which will make sure we don't send too much padding even
// if a single packet is larger than requested.
// We do this to avoid frequently sending small packets on higher bitrates.
padding_bytes_in_packet =
std::min(MaxPayloadSize(), kMaxPaddingLength);
}
size_t bytes_sent = 0;
while (bytes_sent < bytes) {
int64_t now_ms = clock_->TimeInMilliseconds();
@ -502,9 +513,15 @@ size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
timestamp = last_rtp_timestamp_;
capture_time_ms = capture_time_ms_;
if (rtx_ == kRtxOff) {
// Without RTX we can't send padding in the middle of frames.
if (!last_packet_marker_bit_)
if (payload_type_ == -1)
break;
// Without RTX we can't send padding in the middle of frames.
// For audio marker bits doesn't mark the end of a frame and frames
// are usually a single packet, so for now we don't apply this rule
// for audio.
if (!audio_configured_ && !last_packet_marker_bit_) {
break;
}
ssrc = ssrc_;
sequence_number = sequence_number_;
++sequence_number_;
@ -796,7 +813,7 @@ bool RTPSender::IsFecPacket(const RtpPacketToSend& packet) const {
}
size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
if (audio_configured_ || bytes == 0)
if (bytes == 0)
return 0;
size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
if (bytes_sent < bytes)

25
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -1491,4 +1491,29 @@ TEST_F(RtpSenderTest, AddOverheadToTransportFeedbackObserver) {
SendGenericPayload();
}
TEST_F(RtpSenderTest, SendAudioPadding) {
MockTransport transport;
const bool kEnableAudio = true;
rtp_sender_.reset(new RTPSender(
kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr));
rtp_sender_->SetSendPayloadType(kPayload);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
const size_t kPaddingSize = 59;
EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kPaddingSize, rtp_sender_->TimeToSendPadding(
kPaddingSize, PacketInfo::kNotAProbe));
// Requested padding size is too small, will send a larger one.
const size_t kMinPaddingSize = 50;
EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kMinPaddingSize, rtp_sender_->TimeToSendPadding(
kMinPaddingSize - 5, PacketInfo::kNotAProbe));
}
} // namespace webrtc