Enable audio streams to send padding.
Useful if bitrate probing is to be used with audio streams.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2652893004
Cr-Original-Commit-Position: refs/heads/master@{#16404}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: e35f89a484ca376d5c187d166714eba578dfadc3
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index add7c21..c28420c 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -36,6 +36,7 @@
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::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 @@
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 @@
}
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)
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 1b73b65..a6a886b 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1491,4 +1491,29 @@
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
