The audio and video engine is exposed directly rather via redundant wrapping functions. This reduces the amount of boiler plate code. Bug: webrtc:9883 Change-Id: I203a945ee6079397e24a378966a569cd5626ac4a Reviewed-on: https://webrtc-review.googlesource.com/c/106683 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25673}
139 lines
4.8 KiB
C++
139 lines
4.8 KiB
C++
/*
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* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "media/base/mediaengine.h"
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#include <utility>
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#include "api/video/video_bitrate_allocation.h"
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#include "rtc_base/stringencode.h"
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namespace cricket {
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RtpCapabilities::RtpCapabilities() = default;
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RtpCapabilities::~RtpCapabilities() = default;
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webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
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webrtc::RtpParameters parameters;
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webrtc::RtpEncodingParameters encoding;
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parameters.encodings.push_back(encoding);
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return parameters;
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}
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webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) {
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std::vector<uint32_t> primary_ssrcs;
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sp.GetPrimarySsrcs(&primary_ssrcs);
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size_t encoding_count = primary_ssrcs.size();
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std::vector<webrtc::RtpEncodingParameters> encodings(encoding_count);
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for (size_t i = 0; i < encodings.size(); ++i) {
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encodings[i].ssrc = primary_ssrcs[i];
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}
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webrtc::RtpParameters parameters;
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parameters.encodings = encodings;
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parameters.rtcp.cname = sp.cname;
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return parameters;
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}
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webrtc::RTCError ValidateRtpParameters(
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const webrtc::RtpParameters& old_rtp_parameters,
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const webrtc::RtpParameters& rtp_parameters) {
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using webrtc::RTCErrorType;
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if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) {
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LOG_AND_RETURN_ERROR(
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RTCErrorType::INVALID_MODIFICATION,
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"Attempted to set RtpParameters with different encoding count");
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}
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if (rtp_parameters.rtcp != old_rtp_parameters.rtcp) {
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LOG_AND_RETURN_ERROR(
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RTCErrorType::INVALID_MODIFICATION,
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"Attempted to set RtpParameters with modified RTCP parameters");
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}
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if (rtp_parameters.header_extensions !=
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old_rtp_parameters.header_extensions) {
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LOG_AND_RETURN_ERROR(
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RTCErrorType::INVALID_MODIFICATION,
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"Attempted to set RtpParameters with modified header extensions");
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}
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for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
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if (rtp_parameters.encodings[i].ssrc !=
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old_rtp_parameters.encodings[i].ssrc) {
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
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"Attempted to set RtpParameters with modified SSRC");
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}
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if (rtp_parameters.encodings[i].bitrate_priority <= 0) {
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
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"Attempted to set RtpParameters bitrate_priority to "
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"an invalid number. bitrate_priority must be > 0.");
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}
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if (rtp_parameters.encodings[i].min_bitrate_bps &&
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rtp_parameters.encodings[i].max_bitrate_bps) {
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if (*rtp_parameters.encodings[i].max_bitrate_bps <
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*rtp_parameters.encodings[i].min_bitrate_bps) {
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LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_RANGE,
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"Attempted to set RtpParameters min bitrate "
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"larger than max bitrate.");
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}
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}
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if (rtp_parameters.encodings[i].num_temporal_layers) {
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if (*rtp_parameters.encodings[i].num_temporal_layers < 1 ||
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*rtp_parameters.encodings[i].num_temporal_layers >
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webrtc::kMaxTemporalStreams) {
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
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"Attempted to set RtpParameters "
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"num_temporal_layers to an invalid number.");
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}
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}
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if (i > 0 && (rtp_parameters.encodings[i].num_temporal_layers !=
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rtp_parameters.encodings[i - 1].num_temporal_layers)) {
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LOG_AND_RETURN_ERROR(
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RTCErrorType::INVALID_MODIFICATION,
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"Attempted to set RtpParameters num_temporal_layers "
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"at encoding layer i: " +
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rtc::ToString(i) +
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" to a different value than other encoding layers.");
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}
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}
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return webrtc::RTCError::OK();
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}
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CompositeMediaEngine::CompositeMediaEngine(
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std::unique_ptr<VoiceEngineInterface> voice_engine,
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std::unique_ptr<VideoEngineInterface> video_engine)
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: voice_engine_(std::move(voice_engine)),
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video_engine_(std::move(video_engine)) {}
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CompositeMediaEngine::~CompositeMediaEngine() = default;
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bool CompositeMediaEngine::Init() {
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voice().Init();
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return true;
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}
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VoiceEngineInterface& CompositeMediaEngine::voice() {
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return *voice_engine_.get();
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}
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VideoEngineInterface& CompositeMediaEngine::video() {
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return *video_engine_.get();
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}
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const VoiceEngineInterface& CompositeMediaEngine::voice() const {
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return *voice_engine_.get();
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}
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const VideoEngineInterface& CompositeMediaEngine::video() const {
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return *video_engine_.get();
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}
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}; // namespace cricket
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