Revert of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ )
Reason for revert:
Speculatively reverting, since Android end-to-end tests (such as https://build.chromium.org/p/client.webrtc/builders/Android64%20%28M%20Nexus5X%29) started failing.
Original issue's description:
> 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-Commit-Position: refs/heads/master@{#16404}
> Committed: https://chromium.googlesource.com/external/webrtc/+/e35f89a484ca376d5c187d166714eba578dfadc3
TBR=mflodman@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2669033003
Cr-Original-Commit-Position: refs/heads/master@{#16407}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: d3d3ba5159311296d1a190aa84d20cf1c785ba3d
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index c28420c..add7c21 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -36,7 +36,6 @@
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.
@@ -482,21 +481,11 @@
}
size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
- 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);
- }
+ // 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 bytes_sent = 0;
while (bytes_sent < bytes) {
int64_t now_ms = clock_->TimeInMilliseconds();
@@ -513,15 +502,9 @@
timestamp = last_rtp_timestamp_;
capture_time_ms = capture_time_ms_;
if (rtx_ == kRtxOff) {
- 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_) {
+ if (!last_packet_marker_bit_)
break;
- }
ssrc = ssrc_;
sequence_number = sequence_number_;
++sequence_number_;
@@ -813,7 +796,7 @@
}
size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
- if (bytes == 0)
+ if (audio_configured_ || 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 a6a886b..1b73b65 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1491,29 +1491,4 @@
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
