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Add 'AudioPacket' notification to media transport interface.

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So far, base channel was only notifying about 'first audio packet' when
RTP was used, and it never notified about it when media_transport
interface was used. This change adds a sigslot to notify about a new
media packet to the media transport interface.

Bug: webrtc:9719
Change-Id: Ie9230c407f35b1aaa71ba71008ac34ba8869e2d4
Reviewed-on: https://webrtc-review.googlesource.com/c/117249
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26282}
This commit is contained in:
Piotr (Peter) Slatala
2019-01-15 14:24:34 -08:00
committed by Commit Bot
parent e7d08df837
commit 309aafe351
4 changed files with 59 additions and 7 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
api/media_transport_interface.cc
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@ -157,6 +157,9 @@ MediaTransportFactory::CreateMediaTransport(
return std::unique_ptr<MediaTransportInterface>(nullptr);
}
MediaTransportInterface::MediaTransportInterface() = default;
MediaTransportInterface::~MediaTransportInterface() = default;
void MediaTransportInterface::SetKeyFrameRequestCallback(
MediaTransportKeyFrameRequestCallback* callback) {}
@ -168,6 +171,9 @@ MediaTransportInterface::GetLatestTargetTransferRate() {
void MediaTransportInterface::SetNetworkChangeCallback(
MediaTransportNetworkChangeCallback* callback) {}
void MediaTransportInterface::SetFirstAudioPacketReceivedObserver(
AudioPacketReceivedObserver* observer) {}
void MediaTransportInterface::AddTargetTransferRateObserver(
TargetTransferRateObserver* observer) {}
void MediaTransportInterface::RemoveTargetTransferRateObserver(

21
api/media_transport_interface.h
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@ -40,6 +40,15 @@ namespace webrtc {
class RtcEventLog;
class AudioPacketReceivedObserver {
public:
virtual ~AudioPacketReceivedObserver() = default;
// Invoked for the first received audio packet on a given channel id.
// It will be invoked once for each channel id.
virtual void OnFirstAudioPacketReceived(int64_t channel_id) = 0;
};
// A collection of settings for creation of media transport.
struct MediaTransportSettings final {
MediaTransportSettings();
@ -352,7 +361,8 @@ class DataChannelSink {
// and receiving bandwidth estimate update from congestion control.
class MediaTransportInterface {
public:
virtual ~MediaTransportInterface() = default;
MediaTransportInterface();
virtual ~MediaTransportInterface();
// Start asynchronous send of audio frame. The status returned by this method
// only pertains to the synchronous operations (e.g.
@ -399,6 +409,15 @@ class MediaTransportInterface {
virtual void RemoveTargetTransferRateObserver(
TargetTransferRateObserver* observer);
// Sets audio packets observer, which gets informed about incoming audio
// packets. Before destruction, the observer must be unregistered by setting
// nullptr.
//
// This method may be temporary, when the multiplexer is implemented (or
// multiplexer may use it to demultiplex channel ids).
virtual void SetFirstAudioPacketReceivedObserver(
AudioPacketReceivedObserver* observer);
// Intended for receive side. AddRttObserver registers an observer to be
// called for each RTT measurement, typically once per ACK. Before media
// transport is destructed the observer must be unregistered.

19
pc/channel.cc
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@ -744,6 +744,9 @@ VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
crypto_options) {}
VoiceChannel::~VoiceChannel() {
if (media_transport()) {
media_transport()->SetFirstAudioPacketReceivedObserver(nullptr);
}
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
@ -762,6 +765,19 @@ void BaseChannel::OnNetworkRouteChanged(
OnNetworkRouteChanged(absl::make_optional(network_route));
}
void VoiceChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport) {
BaseChannel::Init_w(rtp_transport, media_transport);
if (BaseChannel::media_transport()) {
this->media_transport()->SetFirstAudioPacketReceivedObserver(this);
}
}
void VoiceChannel::OnFirstAudioPacketReceived(int64_t channel_id) {
has_received_packet_ = true;
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
}
void VoiceChannel::UpdateMediaSendRecvState_w() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
@ -1034,7 +1050,8 @@ RtpDataChannel::~RtpDataChannel() {
Deinit();
}
void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport) {
BaseChannel::Init_w(rtp_transport, /*media_transport=*/nullptr);
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);

18
pc/channel.h
View File

@ -20,6 +20,7 @@
#include "api/call/audio_sink.h"
#include "api/jsep.h"
#include "api/media_transport_interface.h"
#include "api/rtp_receiver_interface.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
@ -87,7 +88,7 @@ class BaseChannel : public ChannelInterface,
bool srtp_required,
webrtc::CryptoOptions crypto_options);
virtual ~BaseChannel();
void Init_w(webrtc::RtpTransportInternal* rtp_transport,
virtual void Init_w(webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport);
// Deinit may be called multiple times and is simply ignored if it's already
@ -292,6 +293,8 @@ class BaseChannel : public ChannelInterface,
bool RegisterRtpDemuxerSink();
bool has_received_packet_ = false;
private:
bool ConnectToRtpTransport();
void DisconnectFromRtpTransport();
@ -301,6 +304,7 @@ class BaseChannel : public ChannelInterface,
// MediaTransportNetworkChangeCallback override.
void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route) override;
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
rtc::Thread* const signaling_thread_;
@ -323,7 +327,6 @@ class BaseChannel : public ChannelInterface,
std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
bool writable_ = false;
bool was_ever_writable_ = false;
bool has_received_packet_ = false;
const bool srtp_required_ = true;
webrtc::CryptoOptions crypto_options_;
@ -346,7 +349,8 @@ class BaseChannel : public ChannelInterface,
// VoiceChannel is a specialization that adds support for early media, DTMF,
// and input/output level monitoring.
class VoiceChannel : public BaseChannel {
class VoiceChannel : public BaseChannel,
public webrtc::AudioPacketReceivedObserver {
public:
VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
@ -366,6 +370,8 @@ class VoiceChannel : public BaseChannel {
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
void Init_w(webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport) override;
private:
// overrides from BaseChannel
@ -377,6 +383,8 @@ class VoiceChannel : public BaseChannel {
webrtc::SdpType type,
std::string* error_desc) override;
void OnFirstAudioPacketReceived(int64_t channel_id) override;
// Last AudioSendParameters sent down to the media_channel() via
// SetSendParameters.
AudioSendParameters last_send_params_;
@ -443,7 +451,9 @@ class RtpDataChannel : public BaseChannel {
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport);
void Init_w(webrtc::RtpTransportInternal* rtp_transport);
void Init_w(
webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport = nullptr) override;
virtual bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,