blob: 8c9d5b7fecfe0c31b7b450db005631e37d1d1bda [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 "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/frame_callback.h"
#include "webrtc/video_renderer.h"
namespace webrtc {
class VideoEncoder;
// Class to deliver captured frame to the video send stream.
class VideoSendStreamInput {
// These methods do not lock internally and must be called sequentially.
// If your application switches input sources synchronization must be done
// externally to make sure that any old frames are not delivered concurrently.
virtual void SwapFrame(I420VideoFrame* video_frame) = 0;
virtual ~VideoSendStreamInput() {}
class VideoSendStream {
struct Stats {
: input_frame_rate(0),
suspended(false) {}
int input_frame_rate;
int encode_frame_rate;
bool suspended;
std::map<uint32_t, StreamStats> substreams;
struct Config {
: pre_encode_callback(NULL),
suspend_below_min_bitrate(false) {}
std::string ToString() const;
struct EncoderSettings {
EncoderSettings() : payload_type(-1), encoder(NULL) {}
std::string ToString() const;
std::string payload_name;
int payload_type;
// Uninitialized VideoEncoder instance to be used for encoding. Will be
// initialized from inside the VideoSendStream.
webrtc::VideoEncoder* encoder;
} encoder_settings;
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
struct Rtp {
: max_packet_size(kDefaultMaxPacketSize),
min_transmit_bitrate_bps(0) {}
std::string ToString() const;
std::vector<uint32_t> ssrcs;
// Max RTP packet size delivered to send transport from VideoEngine.
size_t max_packet_size;
// Padding will be used up to this bitrate regardless of the bitrate
// produced by the encoder. Padding above what's actually produced by the
// encoder helps maintaining a higher bitrate estimate.
int min_transmit_bitrate_bps;
// RTP header extensions to use for this send stream.
std::vector<RtpExtension> extensions;
// See NackConfig for description.
NackConfig nack;
// See FecConfig for description.
FecConfig fec;
// Settings for RTP retransmission payload format, see RFC 4588 for
// details.
struct Rtx {
Rtx() : payload_type(-1), pad_with_redundant_payloads(false) {}
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;
// Use redundant payloads to pad the bitrate. Instead of padding with
// randomized packets, we will preemptively retransmit media packets on
// the RTX stream.
bool pad_with_redundant_payloads;
} rtx;
// RTCP CNAME, see RFC 3550.
std::string c_name;
} rtp;
// Called for each I420 frame before encoding the frame. Can be used for
// effects, snapshots etc. 'NULL' disables the callback.
I420FrameCallback* pre_encode_callback;
// Called for each encoded frame, e.g. used for file storage. 'NULL'
// disables the callback.
EncodedFrameObserver* post_encode_callback;
// Renderer for local preview. The local renderer will be called even if
// sending hasn't started. 'NULL' disables local rendering.
VideoRenderer* local_renderer;
// 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;
// Target delay in milliseconds. A positive value indicates this stream is
// used for streaming instead of a real-time call.
int target_delay_ms;
// 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;
// Gets interface used to insert captured frames. Valid as long as the
// VideoSendStream is valid.
virtual VideoSendStreamInput* Input() = 0;
virtual void Start() = 0;
virtual void Stop() = 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 bool ReconfigureVideoEncoder(const std::vector<VideoStream>& streams,
const void* encoder_settings) = 0;
virtual Stats GetStats() const = 0;
virtual ~VideoSendStream() {}
} // namespace webrtc