talk/session/media/channel.cc - Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle.

Unified Diff: talk/session/media/channel.cc

Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: No changes to const parameters. Created 5 years, 2 months ago

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Index: talk/session/media/channel.cc

diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc

index fc998c2531e0d564447a359da9363d117ffe6d5c..4ff90569f17925e4ae89b9d5ae938421281a98a3 100644

--- a/talk/session/media/channel.cc

+++ b/talk/session/media/channel.cc

@@ -30,6 +30,7 @@

#include "talk/media/base/constants.h"

#include "talk/media/base/rtputils.h"

#include "webrtc/p2p/base/transportchannel.h"

+#include "talk/app/webrtc/mediacontroller.h"

#include "talk/session/media/channelmanager.h"

#include "webrtc/base/bind.h"

#include "webrtc/base/buffer.h"

@@ -37,6 +38,7 @@

#include "webrtc/base/common.h"

#include "webrtc/base/dscp.h"

#include "webrtc/base/logging.h"

+#include "webrtc/common_types.h"

namespace cricket {

@@ -171,12 +173,14 @@ void RtpSendParametersFromMediaDescription(

BaseChannel::BaseChannel(rtc::Thread* thread,

MediaChannel* media_channel,

+ webrtc::MediaControllerInterface* media_controller,

TransportController* transport_controller,

const std::string& content_name,

bool rtcp)

: worker_thread_(thread),

transport_controller_(transport_controller),

media_channel_(media_channel),

+ media_controller_(media_controller),

content_name_(content_name),

rtcp_transport_enabled_(rtcp),

transport_channel_(nullptr),

@@ -343,6 +347,7 @@ void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {

tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState);

tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead);

+ tc->SignalSentPacket.connect(this, &BaseChannel::OnPacketSent);

tc->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend);

}

@@ -431,13 +436,13 @@ bool BaseChannel::IsReadyToSend() const {

}

bool BaseChannel::SendPacket(rtc::Buffer* packet,

- rtc::DiffServCodePoint dscp) {

- return SendPacket(false, packet, dscp);

+ const rtc::PacketOptions& options) {

+ return SendPacket(false, packet, options);

}

bool BaseChannel::SendRtcp(rtc::Buffer* packet,

- rtc::DiffServCodePoint dscp) {

- return SendPacket(true, packet, dscp);

+ const rtc::PacketOptions& options) {

+ return SendPacket(true, packet, options);

}

int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,

@@ -477,6 +482,16 @@ void BaseChannel::OnChannelRead(TransportChannel* channel,

HandlePacket(rtcp, &packet, packet_time);

}

+void BaseChannel::OnPacketSent(TransportChannel* channel,

+ const rtc::SentPacket& sent_packet) {

+ if (!media_controller_)

+ return;

pthatcher1 2015/10/05 18:30:13 {}s please

stefan-webrtc 2015/10/07 16:55:25 This code has now been removed.

+ ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);

+ webrtc::SentPacket webrtc_sent_packet(sent_packet.packet_id,

+ sent_packet.send_time_ms);

pthatcher1 2015/10/05 18:30:13 Similarly, can we have just one SentPacket struct?

+ media_controller_->OnPacketSent(webrtc_sent_packet);

pthatcher1 2015/10/05 18:30:13 Would it make sense for this to be media_controlle

void BaseChannel::OnReadyToSend(TransportChannel* channel) {

ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);

SetReadyToSend(channel == rtcp_transport_channel_, true);

@@ -506,8 +521,9 @@ bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,

rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));

}

-bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,

- rtc::DiffServCodePoint dscp) {

+bool BaseChannel::SendPacket(bool rtcp,

+ rtc::Buffer* packet,

+ const rtc::PacketOptions& options) {

// SendPacket gets called from MediaEngine, typically on an encoder thread.

// If the thread is not our worker thread, we will post to our worker

// so that the real work happens on our worker. This avoids us having to

@@ -520,7 +536,7 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,

int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;

PacketMessageData* data = new PacketMessageData;

data->packet = packet->Pass();

- data->dscp = dscp;

+ data->dscp = options.dscp;

worker_thread_->Post(this, message_id, data);

return true;

}

@@ -543,7 +559,8 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,

return false;

}

- rtc::PacketOptions options(dscp);

+ rtc::PacketOptions updated_options;

+ updated_options = options;

// Protect if needed.

if (srtp_filter_.IsActive()) {

bool res;

@@ -559,21 +576,22 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,

res = srtp_filter_.ProtectRtp(

data, len, static_cast<int>(packet->capacity()), &len);

#else

- options.packet_time_params.rtp_sendtime_extension_id =

+ updated_options.packet_time_params.rtp_sendtime_extension_id =

rtp_abs_sendtime_extn_id_;

res = srtp_filter_.ProtectRtp(

data, len, static_cast<int>(packet->capacity()), &len,

- &options.packet_time_params.srtp_packet_index);

+ &updated_options.packet_time_params.srtp_packet_index);

// If protection succeeds, let's get auth params from srtp.

if (res) {

uint8* auth_key = NULL;

int key_len;

res = srtp_filter_.GetRtpAuthParams(

- &auth_key, &key_len, &options.packet_time_params.srtp_auth_tag_len);

+ &auth_key, &key_len,

+ &updated_options.packet_time_params.srtp_auth_tag_len);

if (res) {

- options.packet_time_params.srtp_auth_key.resize(key_len);

- options.packet_time_params.srtp_auth_key.assign(auth_key,

- auth_key + key_len);

+ updated_options.packet_time_params.srtp_auth_key.resize(key_len);

+ updated_options.packet_time_params.srtp_auth_key.assign(

+ auth_key, auth_key + key_len);

}

#endif

@@ -1278,11 +1296,13 @@ void BaseChannel::FlushRtcpMessages() {

VoiceChannel::VoiceChannel(rtc::Thread* thread,

MediaEngineInterface* media_engine,

VoiceMediaChannel* media_channel,

+ webrtc::MediaControllerInterface* media_controller,

TransportController* transport_controller,

const std::string& content_name,

bool rtcp)

: BaseChannel(thread,

media_channel,

+ media_controller,

transport_controller,

content_name,

rtcp),

@@ -1596,11 +1616,13 @@ void VoiceChannel::GetSrtpCryptoSuiteNames(

VideoChannel::VideoChannel(rtc::Thread* thread,

VideoMediaChannel* media_channel,

+ webrtc::MediaControllerInterface* media_controller,

TransportController* transport_controller,

const std::string& content_name,

bool rtcp)

: BaseChannel(thread,

media_channel,

+ media_controller,

transport_controller,

content_name,

rtcp),

@@ -1991,6 +2013,7 @@ DataChannel::DataChannel(rtc::Thread* thread,

bool rtcp)

: BaseChannel(thread,

media_channel,

+ nullptr,

transport_controller,

content_name,

rtcp),

« talk/media/webrtc/webrtcvideoengine2.cc ('K') | « talk/session/media/channel.h ('k') | talk/session/media/channel_unittest.cc » ('j') | talk/session/media/channel_unittest.cc » ('J')