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VoIP interfaces API enhancement (continuation of 169000)

Browse Source 
Bug: webrtc:11251
Change-Id: Iecde33b86856b14db5abade3301a842d5007568d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169034
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30675}
This commit is contained in:
Tim Na
2020-03-03 09:29:22 -08:00
committed by Commit Bot
parent 5e1ea25189
commit ccefde95b3
6 changed files with 85 additions and 70 deletions
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3
api/voip/BUILD.gn
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@ -17,7 +17,8 @@ rtc_source_set("voip_api") {
"voip_network.h",
]
deps = [
"..:transport_api",
"..:array_view",
"../audio_codecs:audio_codecs_api",
"//third_party/abseil-cpp/absl/types:optional",
]
}

5
api/voip/DEPS Normal file
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@ -0,0 +1,5 @@
specific_include_rules = {
".*\.h": [
"+third_party/absl/types/optional.h",
],
}

79
api/voip/voip_base.h
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@ -11,10 +11,12 @@
#ifndef API_VOIP_VOIP_BASE_H_
#define API_VOIP_VOIP_BASE_H_
#include "api/call/transport.h"
#include "third_party/absl/types/optional.h"
namespace webrtc {
class Transport;
// VoipBase interface
//
// VoipBase provides a management interface on a media session using a
@ -22,7 +24,7 @@ namespace webrtc {
// for application to request various media session operations. This
// notion of channel is used throughout other interfaces as well.
//
// Underneath the interface, a channel handle is mapped into an audio session
// Underneath the interface, a channel id is mapped into an audio session
// object that is capable of sending and receiving a single RTP stream with
// another media endpoint. It's possible to create and use multiple active
// channels simultaneously which would mean that particular application
@ -30,47 +32,56 @@ namespace webrtc {
//
// A typical example for the usage context is outlined in VoipEngine
// header file.
enum class ChannelId : int {};
class VoipBase {
public:
// This config enables application to set webrtc::Transport callback pointer
// to receive rtp/rtcp packets from corresponding media session in VoIP
// engine. VoipEngine framework expects applications to handle network I/O
// directly and injection for incoming RTP from remote endpoint is handled
// via VoipNetwork interface.
struct Config {
Transport* transport = nullptr;
uint32_t local_ssrc = 0;
};
// Creates a channel.
// Each channel handle maps into one audio media session where each has
// its own separate module for send/receive rtp packet with one peer.
// Caller must set |transport|, webrtc::Transport callback pointer to
// receive rtp/rtcp packets from corresponding media session in VoIP engine.
// VoipEngine framework expects applications to handle network I/O directly
// and injection for incoming RTP from remote endpoint is handled via
// VoipNetwork interface. |local_ssrc| is optional and when local_ssrc is not
// set, some random value will be used by voip engine.
// Returns value is optional as to indicate the failure to create channel.
virtual absl::optional<ChannelId> CreateChannel(
Transport* transport,
absl::optional<uint32_t> local_ssrc) = 0;
// Create a channel handle.
// Valid handle value is zero or greater integer whereas -1 represents error
// during media session construction. Each channel handle maps into one
// audio media session where each has its own separate module for
// send/receive rtp packet with one peer.
virtual int CreateChannel(const Config& config) = 0;
// Releases |channel_id| that has served the purpose.
// Released channel will be re-allocated again that invoking operations
// on released |channel_id| will lead to undefined behavior.
virtual void ReleaseChannel(ChannelId channel_id) = 0;
// Following methods return boolean to indicate if the operation is succeeded.
// API is subject to expand to reflect error condition to application later.
// Starts sending on |channel_id|. This will start microphone if first to
// start. Returns false if initialization has failed on selected microphone
// device. API is subject to expand to reflect error condition to application
// later.
virtual bool StartSend(ChannelId channel_id) = 0;
// Release |channel| that has served the purpose.
// Released channel handle will be re-allocated again. Invoking
// an operation on released channel will lead to undefined behavior.
virtual bool ReleaseChannel(int channel) = 0;
// Start sending on |channel|. This will start microphone if first to start.
virtual bool StartSend(int channel) = 0;
// Stop sending on |channel|. If this is the last active channel, it will
// Stops sending on |channel_id|. If this is the last active channel, it will
// stop microphone input from underlying audio platform layer.
virtual bool StopSend(int channel) = 0;
// Returns false if termination logic has failed on selected microphone
// device. API is subject to expand to reflect error condition to application
// later.
virtual bool StopSend(ChannelId channel_id) = 0;
// Start playing on speaker device for |channel|.
// Starts playing on speaker device for |channel_id|.
// This will start underlying platform speaker device if not started.
virtual bool StartPlayout(int channel) = 0;
// Returns false if initialization has failed
// on selected speaker device. API is subject to expand to reflect error
// condition to application later.
virtual bool StartPlayout(ChannelId channel_id) = 0;
// Stop playing on speaker device for |channel|. If this is the last
// active channel playing, then it will stop speaker from the platform layer.
virtual bool StopPlayout(int channel) = 0;
// Stops playing on speaker device for |channel_id|.
// If this is the last active channel playing, then it will stop speaker
// from the platform layer.
// Returns false if termination logic has failed on selected speaker device.
// API is subject to expand to reflect error condition to application later.
virtual bool StopPlayout(ChannelId channel_id) = 0;
protected:
virtual ~VoipBase() = default;

9
api/voip/voip_codec.h
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@ -14,6 +14,7 @@
#include <map>
#include "api/audio_codecs/audio_format.h"
#include "api/voip/voip_base.h"
namespace webrtc {
@ -24,11 +25,11 @@ namespace webrtc {
// are used with negotiated codecs. This interface is subject to expand
// as needed in future.
//
// This interface requires a channel handle created via VoipBase interface.
// This interface requires a channel id created via VoipBase interface.
class VoipCodec {
public:
// Set encoder type here along with its payload type to use.
virtual bool SetSendCodec(int channel,
virtual void SetSendCodec(ChannelId channel_id,
int payload_type,
const SdpAudioFormat& encoder_spec) = 0;
@ -36,8 +37,8 @@ class VoipCodec {
// this should be called after payload type has been agreed in media
// session. Note that payload type can differ with same codec in each
// direction.
virtual bool SetReceiveCodecs(
int channel,
virtual void SetReceiveCodecs(
ChannelId channel_id,
const std::map<int, SdpAudioFormat>& decoder_specs) = 0;
protected:

46
api/voip/voip_engine.h
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@ -11,14 +11,12 @@
#ifndef API_VOIP_VOIP_ENGINE_H_
#define API_VOIP_VOIP_ENGINE_H_
#include <memory>
#include "api/voip/voip_base.h"
#include "api/voip/voip_codec.h"
#include "api/voip/voip_network.h"
namespace webrtc {
class VoipBase;
class VoipCodec;
class VoipNetwork;
// VoipEngine interfaces
//
// These pointer interfaces are valid as long as VoipEngine is available.
@ -31,47 +29,47 @@ namespace webrtc {
// .SetAudioDecoderFactory(CreateBuiltinAudioDecoderFactory())
// .Create();
//
// auto* voip_base = voip_engine->Base();
// auto* voip_codec = voip_engine->Codec();
// auto* voip_network = voip_engine->Network();
// auto voip_base = voip_engine->Base();
// auto voip_codec = voip_engine->Codec();
// auto voip_network = voip_engine->Network();
//
// VoipChannel::Config config = { &app_transport_, 0xdeadc0de };
// int channel = voip_base->CreateChannel(config);
// int channel = voip_base.CreateChannel(config);
//
// // After SDP offer/answer, payload type and codec usage have been
// // decided through negotiation.
// voip_codec->SetSendCodec(channel, ...);
// voip_codec->SetReceiveCodecs(channel, ...);
// voip_codec.SetSendCodec(channel, ...);
// voip_codec.SetReceiveCodecs(channel, ...);
//
// // Start Send/Playout on voip channel.
// voip_base->StartSend(channel);
// voip_base->StartPlayout(channel);
// voip_base.StartSend(channel);
// voip_base.StartPlayout(channel);
//
// // Inject received rtp/rtcp thru voip network interface.
// voip_network->ReceivedRTPPacket(channel, rtp_data, rtp_size);
// voip_network->ReceivedRTCPPacket(channel, rtcp_data, rtcp_size);
// voip_network.ReceivedRTPPacket(channel, rtp_data, rtp_size);
// voip_network.ReceivedRTCPPacket(channel, rtcp_data, rtcp_size);
//
// // Stop and release voip channel.
// voip_base->StopSend(channel);
// voip_base->StopPlayout(channel);
// voip_base.StopSend(channel);
// voip_base.StopPlayout(channel);
//
// voip_base->ReleaseChannel(channel);
// voip_base.ReleaseChannel(channel);
//
class VoipEngine {
public:
virtual ~VoipEngine() = default;
// VoipBase is the audio session management interface that
// create/release/start/stop one-to-one audio media session.
virtual VoipBase* Base() = 0;
virtual VoipBase& Base() = 0;
// VoipNetwork provides injection APIs that would enable application
// to send and receive RTP/RTCP packets. There is no default network module
// that provides RTP transmission and reception.
virtual VoipNetwork* Network() = 0;
virtual VoipNetwork& Network() = 0;
// VoipCodec provides codec configuration APIs for encoder and decoders.
virtual VoipCodec* Codec() = 0;
virtual ~VoipEngine() = default;
virtual VoipCodec& Codec() = 0;
};
} // namespace webrtc

13
api/voip/voip_network.h
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@ -11,7 +11,8 @@
#ifndef API_VOIP_VOIP_NETWORK_H_
#define API_VOIP_VOIP_NETWORK_H_
#include "api/call/transport.h"
#include "api/array_view.h"
#include "api/voip/voip_base.h"
namespace webrtc {
@ -25,16 +26,14 @@ class VoipNetwork {
// The packets received from the network should be passed to this
// function. Note that the data including the RTP-header must also be
// given to the VoipEngine.
virtual bool ReceivedRTPPacket(int channel,
const uint8_t* data,
size_t length) = 0;
virtual void ReceivedRTPPacket(ChannelId channel_id,
rtc::ArrayView<const uint8_t> data) = 0;
// The packets received from the network should be passed to this
// function. Note that the data including the RTCP-header must also be
// given to the VoipEngine.
virtual bool ReceivedRTCPPacket(int channel,
const uint8_t* data,
size_t length) = 0;
virtual void ReceivedRTCPPacket(ChannelId channel_id,
rtc::ArrayView<const uint8_t> data) = 0;
protected:
virtual ~VoipNetwork() = default;