blob: 6dd13ab756b44a8f58e634040bd39a0f58cfa5da [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 <map>
#include <string>
#include <utility>
#include <vector>
#include "webrtc/api/call/transport.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/call/rtp_config.h"
#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/rtc_base/platform_file.h"
namespace webrtc {
class VideoEncoder;
class VideoSendStream {
struct StreamStats {
std::string ToString() const;
FrameCounts frame_counts;
bool is_rtx = false;
bool is_flexfec = false;
int width = 0;
int height = 0;
// TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
int total_bitrate_bps = 0;
int retransmit_bitrate_bps = 0;
int avg_delay_ms = 0;
int max_delay_ms = 0;
StreamDataCounters rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
RtcpStatistics rtcp_stats;
struct Stats {
std::string ToString(int64_t time_ms) const;
std::string encoder_implementation_name = "unknown";
int input_frame_rate = 0;
int encode_frame_rate = 0;
int avg_encode_time_ms = 0;
int encode_usage_percent = 0;
uint32_t frames_encoded = 0;
rtc::Optional<uint64_t> qp_sum;
// Bitrate the encoder is currently configured to use due to bandwidth
// limitations.
int target_media_bitrate_bps = 0;
// Bitrate the encoder is actually producing.
int media_bitrate_bps = 0;
// Media bitrate this VideoSendStream is configured to prefer if there are
// no bandwidth limitations.
int preferred_media_bitrate_bps = 0;
bool suspended = false;
bool bw_limited_resolution = false;
bool cpu_limited_resolution = false;
bool bw_limited_framerate = false;
bool cpu_limited_framerate = false;
// Total number of times resolution as been requested to be changed due to
// CPU/quality adaptation.
int number_of_cpu_adapt_changes = 0;
int number_of_quality_adapt_changes = 0;
std::map<uint32_t, StreamStats> substreams;
webrtc::VideoContentType content_type =
struct Config {
Config() = delete;
explicit Config(Transport* send_transport);
Config& operator=(Config&&);
Config& operator=(const Config&) = delete;
// Mostly used by tests. Avoid creating copies if you can.
Config Copy() const { return Config(*this); }
std::string ToString() const;
struct EncoderSettings {
EncoderSettings() = default;
EncoderSettings(std::string payload_name,
int payload_type,
VideoEncoder* encoder)
: payload_name(std::move(payload_name)),
encoder(encoder) {}
std::string ToString() const;
std::string payload_name;
int payload_type = -1;
// TODO(sophiechang): Delete this field when no one is using internal
// sources anymore.
bool internal_source = false;
// Allow 100% encoder utilization. Used for HW encoders where CPU isn't
// expected to be the limiting factor, but a chip could be running at
// 30fps (for example) exactly.
bool full_overuse_time = false;
// Uninitialized VideoEncoder instance to be used for encoding. Will be
// initialized from inside the VideoSendStream.
VideoEncoder* encoder = nullptr;
} encoder_settings;
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
struct Rtp {
Rtp(const Rtp&);
std::string ToString() const;
std::vector<uint32_t> ssrcs;
// See RtcpMode for description.
RtcpMode rtcp_mode = RtcpMode::kCompound;
// Max RTP packet size delivered to send transport from VideoEngine.
size_t max_packet_size = kDefaultMaxPacketSize;
// RTP header extensions to use for this send stream.
std::vector<RtpExtension> extensions;
// See NackConfig for description.
NackConfig nack;
// See UlpfecConfig for description.
UlpfecConfig ulpfec;
struct Flexfec {
Flexfec(const Flexfec&);
// Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
int payload_type = -1;
// SSRC of FlexFEC stream.
uint32_t ssrc = 0;
// Vector containing a single element, corresponding to the SSRC of the
// media stream being protected by this FlexFEC stream.
// The vector MUST have size 1.
// TODO(brandtr): Update comment above when we support
// multistream protection.
std::vector<uint32_t> protected_media_ssrcs;
} flexfec;
// Settings for RTP retransmission payload format, see RFC 4588 for
// details.
struct Rtx {
Rtx(const Rtx&);
std::string ToString() const;
// SSRCs to use for the RTX streams.
std::vector<uint32_t> ssrcs;
// Payload type to use for the RTX stream.
int payload_type = -1;
} rtx;
// RTCP CNAME, see RFC 3550.
std::string c_name;
} rtp;
// Transport for outgoing packets.
Transport* send_transport = nullptr;
// Called for each I420 frame before encoding the frame. Can be used for
// effects, snapshots etc. 'nullptr' disables the callback.
rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
// Called for each encoded frame, e.g. used for file storage. 'nullptr'
// disables the callback. Also measures timing and passes the time
// spent on encoding. This timing will not fire if encoding takes longer
// than the measuring window, since the sample data will have been dropped.
EncodedFrameObserver* post_encode_callback = nullptr;
// Expected delay needed by the renderer, i.e. the frame will be delivered
// this many milliseconds, if possible, earlier than expected render time.
// Only valid if |local_renderer| is set.
int render_delay_ms = 0;
// Target delay in milliseconds. A positive value indicates this stream is
// used for streaming instead of a real-time call.
int target_delay_ms = 0;
// True if the stream should be suspended when the available bitrate fall
// below the minimum configured bitrate. If this variable is false, the
// stream may send at a rate higher than the estimated available bitrate.
bool suspend_below_min_bitrate = false;
// Enables periodic bandwidth probing in application-limited region.
bool periodic_alr_bandwidth_probing = false;
// Access to the copy constructor is private to force use of the Copy()
// method for those exceptional cases where we do use it.
Config(const Config&);
// Starts stream activity.
// When a stream is active, it can receive, process and deliver packets.
virtual void Start() = 0;
// Stops stream activity.
// When a stream is stopped, it can't receive, process or deliver packets.
virtual void Stop() = 0;
// Based on the spec in
// These options are enforced on a best-effort basis. For instance, all of
// these options may suffer some frame drops in order to avoid queuing.
// TODO(sprang): Look into possibility of more strictly enforcing the
// maintain-framerate option.
enum class DegradationPreference {
// Don't take any actions based on over-utilization signals.
// On over-use, request lower frame rate, possibly causing frame drops.
// On over-use, request lower resolution, possibly causing down-scaling.
// Try to strike a "pleasing" balance between frame rate or resolution.
virtual void SetSource(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const DegradationPreference& degradation_preference) = 0;
// Set which streams to send. Must have at least as many SSRCs as configured
// in the config. Encoder settings are passed on to the encoder instance along
// with the VideoStream settings.
virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;
virtual Stats GetStats() = 0;
// Takes ownership of each file, is responsible for closing them later.
// Calling this method will close and finalize any current logs.
// Some codecs produce multiple streams (VP8 only at present), each of these
// streams will log to a separate file. kMaxSimulcastStreams in common_types.h
// gives the max number of such streams. If there is no file for a stream, or
// the file is rtc::kInvalidPlatformFileValue, frames from that stream will
// not be logged.
// If a frame to be written would make the log too large the write fails and
// the log is closed and finalized. A |byte_limit| of 0 means no limit.
virtual void EnableEncodedFrameRecording(
const std::vector<rtc::PlatformFile>& files,
size_t byte_limit) = 0;
inline void DisableEncodedFrameRecording() {
EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
virtual ~VideoSendStream() {}
} // namespace webrtc