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Modifying MediaChannel to accept CopyOnWriteBuffer by value.

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MediaChannel accepted the RtpPacket buffers through non-const pointer.
This is both unclear and introduces questions regarding if the buffer is
actually copied or not.
This change modifies the method to accept by value to reduce ambiguity.
Usage of the non-const data() method which could potentially copy the
buffer contents is also reduced in favor of cdata() which never copies.

Bug: None
Change-Id: I3b2daef0d31cb6aacceb46c86da3a40ce836242b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27090}
This commit is contained in:
Amit Hilbuch 2019-03-12 11:10:27 -07:00 committed by Commit Bot
parent dfaea9dd98
commit e7a5f7bfae
15 changed files with 61 additions and 65 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
media/base/fake_media_engine.h
View File

@ -267,13 +267,13 @@ class RtpHelper : public Base {
void set_recv_rtcp_parameters(const RtcpParameters& params) {
recv_rtcp_parameters_ = params;
}
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
rtp_packets_.push_back(std::string(packet.cdata<char>(), packet.size()));
}
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
rtcp_packets_.push_back(std::string(packet.cdata<char>(), packet.size()));
}
virtual void OnReadyToSend(bool ready) { ready_to_send_ = ready; }

4
media/base/fake_network_interface.h
View File

@ -169,9 +169,9 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
static_cast<rtc::TypedMessageData<rtc::CopyOnWriteBuffer>*>(msg->pdata);
if (dest_) {
if (msg->message_id == ST_RTP) {
dest_->OnPacketReceived(&msg_data->data(), rtc::TimeMicros());
dest_->OnPacketReceived(msg_data->data(), rtc::TimeMicros());
} else {
dest_->OnRtcpReceived(&msg_data->data(), rtc::TimeMicros());
dest_->OnRtcpReceived(msg_data->data(), rtc::TimeMicros());
}
}
delete msg_data;

4
media/base/media_channel.h
View File

@ -196,10 +196,10 @@ class MediaChannel : public sigslot::has_slots<> {
virtual void SetInterface(NetworkInterface* iface,
webrtc::MediaTransportInterface* media_transport);
// Called when a RTP packet is received.
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) = 0;
// Called when a RTCP packet is received.
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) = 0;
// Called when the socket's ability to send has changed.
virtual void OnReadyToSend(bool ready) = 0;

10
media/base/rtp_data_engine.cc
View File

@ -194,20 +194,20 @@ bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
return true;
}
void RtpDataMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
void RtpDataMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t /* packet_time_us */) {
RtpHeader header;
if (!GetRtpHeader(packet->cdata(), packet->size(), &header)) {
if (!GetRtpHeader(packet.cdata(), packet.size(), &header)) {
return;
}
size_t header_length;
if (!GetRtpHeaderLen(packet->cdata(), packet->size(), &header_length)) {
if (!GetRtpHeaderLen(packet.cdata(), packet.size(), &header_length)) {
return;
}
const char* data =
packet->cdata<char>() + header_length + sizeof(kReservedSpace);
size_t data_len = packet->size() - header_length - sizeof(kReservedSpace);
packet.cdata<char>() + header_length + sizeof(kReservedSpace);
size_t data_len = packet.size() - header_length - sizeof(kReservedSpace);
if (!receiving_) {
RTC_LOG(LS_WARNING) << "Not receiving packet " << header.ssrc << ":"

4
media/base/rtp_data_engine.h
View File

@ -81,9 +81,9 @@ class RtpDataMediaChannel : public DataMediaChannel {
receiving_ = receive;
return true;
}
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us);
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {}
virtual void OnReadyToSend(bool ready) {}
virtual bool SendData(const SendDataParams& params,

10
media/base/rtp_data_engine_unittest.cc
View File

@ -319,13 +319,13 @@ TEST_F(RtpDataMediaChannelTest, ReceiveData) {
std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
// SetReceived not called.
dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
EXPECT_FALSE(HasReceivedData());
dmc->SetReceive(true);
// Unknown payload id
dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
EXPECT_FALSE(HasReceivedData());
cricket::DataCodec codec;
@ -336,7 +336,7 @@ TEST_F(RtpDataMediaChannelTest, ReceiveData) {
ASSERT_TRUE(dmc->SetRecvParameters(parameters));
// Unknown stream
dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
EXPECT_FALSE(HasReceivedData());
cricket::StreamParams stream;
@ -344,7 +344,7 @@ TEST_F(RtpDataMediaChannelTest, ReceiveData) {
ASSERT_TRUE(dmc->AddRecvStream(stream));
// Finally works!
dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
EXPECT_TRUE(HasReceivedData());
EXPECT_EQ("abcde", GetReceivedData());
EXPECT_EQ(5U, GetReceivedDataLen());
@ -357,6 +357,6 @@ TEST_F(RtpDataMediaChannelTest, InvalidRtpPackets) {
std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
// Too short
dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
EXPECT_FALSE(HasReceivedData());
}

14
media/engine/webrtc_video_engine.cc
View File

@ -1360,11 +1360,11 @@ void WebRtcVideoChannel::FillSendAndReceiveCodecStats(
}
}
void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
RTC_DCHECK_RUN_ON(&thread_checker_);
const webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet,
packet_time_us);
switch (delivery_result) {
case webrtc::PacketReceiver::DELIVERY_OK:
@ -1380,12 +1380,12 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
}
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
if (!GetRtpSsrc(packet.cdata(), packet.size(), &ssrc)) {
return;
}
int payload_type = 0;
if (!GetRtpPayloadType(packet->cdata(), packet->size(), &payload_type)) {
if (!GetRtpPayloadType(packet.cdata(), packet.size(), &payload_type)) {
return;
}
@ -1412,7 +1412,7 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
break;
}
if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet,
packet_time_us) !=
webrtc::PacketReceiver::DELIVERY_OK) {
RTC_LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
@ -1420,14 +1420,14 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
}
}
void WebRtcVideoChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
void WebRtcVideoChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
RTC_DCHECK_RUN_ON(&thread_checker_);
// TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
// for both audio and video on the same path. Since BundleFilter doesn't
// filter RTCP anymore incoming RTCP packets could've been going to audio (so
// logging failures spam the log).
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet,
packet_time_us);
}

4
media/engine/webrtc_video_engine.h
View File

@ -144,9 +144,9 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
bool GetStats(VideoMediaInfo* info) override;
void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
void OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnReadyToSend(bool ready) override;
void OnNetworkRouteChanged(const std::string& transport_name,

32
media/engine/webrtc_video_engine_unittest.cc
View File

@ -1729,7 +1729,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSink) {
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
channel_->OnPacketReceived(&packet1, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet1, /* packet_time_us */ -1);
EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
@ -4835,7 +4835,7 @@ TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) {
memset(data, 0, sizeof(data));
rtc::SetBE32(&data[8], ssrcs[0]);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
<< "No default receive stream created.";
@ -4993,7 +4993,7 @@ TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
memset(data, 0, sizeof(data));
rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// The stream should now be created with the appropriate sync label.
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
@ -5006,7 +5006,7 @@ TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
ASSERT_TRUE(channel_->RemoveRecvStream(kIncomingUnsignalledSsrc));
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_TRUE(
fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group.empty());
@ -5045,7 +5045,7 @@ TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMsUnsignaledRecvStream) {
memset(data, 0, sizeof(data));
rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
recv_stream = fake_call_->GetVideoReceiveStream(kIncomingUnsignalledSsrc);
EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 200);
@ -5082,7 +5082,7 @@ void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket(
rtc::Set8(data, 1, payload_type);
rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
rtc::CopyOnWriteBuffer packet(data, kDataLength);
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
if (expect_created_receive_stream) {
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
@ -5169,7 +5169,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
rtpHeader.ssrc = kIncomingUnsignalledSsrc + 1;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet(data, sizeof(data));
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// VP8 packet should create default receive stream.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
@ -5190,7 +5190,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
rtpHeader.ssrc = kIncomingUnsignalledSsrc + 2;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet2(data, sizeof(data));
channel_->OnPacketReceived(&packet2, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet2, /* packet_time_us */ -1);
// VP9 packet should replace the default receive SSRC.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
@ -5212,7 +5212,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
rtpHeader.ssrc = kIncomingUnsignalledSsrc + 3;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet3(data, sizeof(data));
channel_->OnPacketReceived(&packet3, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet3, /* packet_time_us */ -1);
// H264 packet should replace the default receive SSRC.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
@ -5251,7 +5251,7 @@ TEST_F(WebRtcVideoChannelTest,
rtp_header.ssrc = kSsrcs3[0];
cricket::SetRtpHeader(data, sizeof(data), rtp_header);
rtc::CopyOnWriteBuffer packet(data, sizeof(data));
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// Default receive stream should be created.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
FakeVideoReceiveStream* recv_stream0 =
@ -5269,7 +5269,7 @@ TEST_F(WebRtcVideoChannelTest,
rtp_header.ssrc = kSsrcs3[1];
cricket::SetRtpHeader(data, sizeof(data), rtp_header);
packet.SetData(data, sizeof(data));
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// New default receive stream should be created, but old stream should remain.
ASSERT_EQ(2u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_EQ(recv_stream0, fake_call_->GetVideoReceiveStreams()[0]);
@ -6891,7 +6891,7 @@ TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersWithUnsignaledSsrc) {
rtpHeader.ssrc = kIncomingUnsignalledSsrc;
cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
rtc::CopyOnWriteBuffer packet(data, sizeof(data));
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// The |ssrc| member should still be unset.
rtp_parameters = channel_->GetRtpReceiveParameters(0);
@ -7191,7 +7191,7 @@ TEST_F(WebRtcVideoChannelTestWithClock, GetSources) {
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
channel_->OnPacketReceived(&packet1, /*packet_time_us=*/-1);
channel_->OnPacketReceived(packet1, /*packet_time_us=*/-1);
std::vector<webrtc::RtpSource> sources = channel_->GetSources(kSsrc);
EXPECT_EQ(1u, sources.size());
@ -7201,7 +7201,7 @@ TEST_F(WebRtcVideoChannelTestWithClock, GetSources) {
// a new packet.
int64_t timeDeltaMs = 1;
fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(timeDeltaMs));
channel_->OnPacketReceived(&packet1, /*packet_time_us=*/-1);
channel_->OnPacketReceived(packet1, /*packet_time_us=*/-1);
int64_t timestamp2 = channel_->GetSources(kSsrc)[0].timestamp_ms();
EXPECT_EQ(timestamp2, timestamp1 + timeDeltaMs);
@ -7226,7 +7226,7 @@ TEST_F(WebRtcVideoChannelTestWithClock, GetContributingSources) {
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
channel_->OnPacketReceived(&packet1, /*packet_time_us=*/-1);
channel_->OnPacketReceived(packet1, /*packet_time_us=*/-1);
{
ASSERT_EQ(2u, channel_->GetSources(kSsrc).size());
@ -7251,7 +7251,7 @@ TEST_F(WebRtcVideoChannelTestWithClock, GetContributingSources) {
int64_t timeDeltaMs = 1;
fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(timeDeltaMs));
channel_->OnPacketReceived(&packet2, /*packet_time_us=*/-1);
channel_->OnPacketReceived(packet2, /*packet_time_us=*/-1);
{
ASSERT_EQ(2u, channel_->GetSources(kSsrc).size());

12
media/engine/webrtc_voice_engine.cc
View File

@ -2058,12 +2058,12 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
event, duration);
}
void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, packet,
packet_time_us);
if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
@ -2074,7 +2074,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
// If there already is N unsignaled receive streams, delete the oldest.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
if (!GetRtpSsrc(packet.cdata(), packet.size(), &ssrc)) {
return;
}
RTC_DCHECK(!absl::c_linear_search(unsignaled_recv_ssrcs_, ssrc));
@ -2117,16 +2117,16 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
}
delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
*packet, packet_time_us);
packet, packet_time_us);
RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
void WebRtcVoiceMediaChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Forward packet to Call as well.
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, packet,
packet_time_us);
}

4
media/engine/webrtc_voice_engine.h
View File

@ -201,9 +201,9 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
bool CanInsertDtmf() override;
bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
void OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;

6
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -255,7 +255,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
void DeliverPacket(const void* data, int len) {
rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len);
channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
}
void TearDown() override { delete channel_; }
@ -3322,9 +3322,9 @@ TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
const cricket::FakeAudioReceiveStream* s =
call_.GetAudioReceiveStream(kAudioSsrc);
EXPECT_EQ(0, s->received_packets());
channel_->OnPacketReceived(&kPcmuPacket, /* packet_time_us */ -1);
channel_->OnPacketReceived(kPcmuPacket, /* packet_time_us */ -1);
EXPECT_EQ(1, s->received_packets());
channel_->OnRtcpReceived(&kRtcpPacket, /* packet_time_us */ -1);
channel_->OnRtcpReceived(kRtcpPacket, /* packet_time_us */ -1);
EXPECT_EQ(2, s->received_packets());
}

7
pc/channel.cc
View File

@ -539,13 +539,10 @@ void BaseChannel::ProcessPacket(bool rtcp,
int64_t packet_time_us) {
RTC_DCHECK(worker_thread_->IsCurrent());
// Need to copy variable because OnRtcpReceived/OnPacketReceived
// requires non-const pointer to buffer. This doesn't memcpy the actual data.
rtc::CopyOnWriteBuffer data(packet);
if (rtcp) {
media_channel_->OnRtcpReceived(&data, packet_time_us);
media_channel_->OnRtcpReceived(packet, packet_time_us);
} else {
media_channel_->OnPacketReceived(&data, packet_time_us);
media_channel_->OnPacketReceived(packet, packet_time_us);
}
}

3
pc/jsep_transport.cc
View File

@ -407,8 +407,7 @@ webrtc::RTCError JsepTransport::SetNegotiatedDtlsParameters(
// Apply remote fingerprint.
if (!remote_fingerprint ||
!dtls_transport->SetRemoteFingerprint(
remote_fingerprint->algorithm,
reinterpret_cast<const uint8_t*>(remote_fingerprint->digest.data()),
remote_fingerprint->algorithm, remote_fingerprint->digest.cdata(),
remote_fingerprint->digest.size())) {
return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
"Failed to apply remote fingerprint.");

4
pc/rtp_transport.cc
View File

@ -129,8 +129,8 @@ bool RtpTransport::SendPacket(bool rtcp,
rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
? rtcp_packet_transport_
: rtp_packet_transport_;
int ret = transport->SendPacket(packet->data<char>(), packet->size(), options,
flags);
int ret = transport->SendPacket(packet->cdata<char>(), packet->size(),
options, flags);
if (ret != static_cast<int>(packet->size())) {
if (transport->GetError() == ENOTCONN) {
RTC_LOG(LS_WARNING) << "Got ENOTCONN from transport.";