blob: 74ea2eb8adc7f99ab42b1d8f63ab16898ee00b9a [file] [log] [blame]
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <algorithm>
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/rtcerror.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
#include "webrtc/call/audio_state.h"
#include "webrtc/call/flexfec_receive_stream.h"
#include "webrtc/call/rtp_transport_controller_send_interface.h"
#include "webrtc/call/video_receive_stream.h"
#include "webrtc/call/video_send_stream.h"
#include "webrtc/common_types.h"
#include "webrtc/rtc_base/networkroute.h"
#include "webrtc/rtc_base/platform_file.h"
#include "webrtc/rtc_base/socket.h"
namespace webrtc {
class AudioProcessing;
class RtcEventLog;
enum class MediaType {
// Like std::min, but considers non-positive values to be unset.
// TODO(zstein): Remove once all callers use rtc::Optional.
template <typename T>
static T MinPositive(T a, T b) {
if (a <= 0) {
return b;
if (b <= 0) {
return a;
return std::min(a, b);
class PacketReceiver {
enum DeliveryStatus {
virtual DeliveryStatus DeliverPacket(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) = 0;
virtual ~PacketReceiver() {}
// A Call instance can contain several send and/or receive streams. All streams
// are assumed to have the same remote endpoint and will share bitrate estimates
// etc.
class Call {
struct Config {
explicit Config(RtcEventLog* event_log) : event_log(event_log) {
static constexpr int kDefaultStartBitrateBps = 300000;
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used. This comes from the remote connection.
struct BitrateConfig {
int min_bitrate_bps = 0;
int start_bitrate_bps = kDefaultStartBitrateBps;
int max_bitrate_bps = -1;
} bitrate_config;
// The local client's bitrate preferences. The actual configuration used
// is a combination of this and |bitrate_config|. The combination is
// currently more complicated than a simple mask operation (see
// SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
// start <= max holds for set parameters.
struct BitrateConfigMask {
rtc::Optional<int> min_bitrate_bps;
rtc::Optional<int> start_bitrate_bps;
rtc::Optional<int> max_bitrate_bps;
// AudioState which is possibly shared between multiple calls.
// TODO(solenberg): Change this to a shared_ptr once we can use C++11.
rtc::scoped_refptr<AudioState> audio_state;
// Audio Processing Module to be used in this call.
// TODO(solenberg): Change this to a shared_ptr once we can use C++11.
AudioProcessing* audio_processing = nullptr;
// RtcEventLog to use for this call. Required.
// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
RtcEventLog* event_log = nullptr;
struct Stats {
std::string ToString(int64_t time_ms) const;
int send_bandwidth_bps = 0; // Estimated available send bandwidth.
int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
int64_t pacer_delay_ms = 0;
int64_t rtt_ms = -1;
static Call* Create(const Call::Config& config);
// Allows mocking |transport_send| for testing.
static Call* Create(
const Call::Config& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
virtual AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) = 0;
virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
virtual AudioReceiveStream* CreateAudioReceiveStream(
const AudioReceiveStream::Config& config) = 0;
virtual void DestroyAudioReceiveStream(
AudioReceiveStream* receive_stream) = 0;
virtual VideoSendStream* CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config) = 0;
virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
virtual VideoReceiveStream* CreateVideoReceiveStream(
VideoReceiveStream::Config configuration) = 0;
virtual void DestroyVideoReceiveStream(
VideoReceiveStream* receive_stream) = 0;
// In order for a created VideoReceiveStream to be aware that it is
// protected by a FlexfecReceiveStream, the latter should be created before
// the former.
virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) = 0;
virtual void DestroyFlexfecReceiveStream(
FlexfecReceiveStream* receive_stream) = 0;
// All received RTP and RTCP packets for the call should be inserted to this
// PacketReceiver. The PacketReceiver pointer is valid as long as the
// Call instance exists.
virtual PacketReceiver* Receiver() = 0;
// Returns the call statistics, such as estimated send and receive bandwidth,
// pacing delay, etc.
virtual Stats GetStats() const = 0;
// The greater min and smaller max set by this and SetBitrateConfigMask will
// be used. The latest non-negative start value from either call will be used.
// Specifying a start bitrate (>0) will reset the current bitrate estimate.
// This is due to how the 'x-google-start-bitrate' flag is currently
// implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
// guaranteed for other negative values or 0.
virtual void SetBitrateConfig(
const Config::BitrateConfig& bitrate_config) = 0;
// The greater min and smaller max set by this and SetBitrateConfig will be
// used. The latest non-negative start value form either call will be used.
// Specifying a start bitrate will reset the current bitrate estimate.
// Assumes 0 <= min <= start <= max holds for set parameters.
virtual void SetBitrateConfigMask(
const Config::BitrateConfigMask& bitrate_mask) = 0;
// TODO(skvlad): When the unbundled case with multiple streams for the same
// media type going over different networks is supported, track the state
// for each stream separately. Right now it's global per media type.
virtual void SignalChannelNetworkState(MediaType media,
NetworkState state) = 0;
virtual void OnTransportOverheadChanged(
MediaType media,
int transport_overhead_per_packet) = 0;
virtual void OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) = 0;
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
virtual ~Call() {}
} // namespace webrtc