This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
177 lines
6.3 KiB
C++
177 lines
6.3 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/media_engine.h"
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#include <stddef.h>
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#include <cstdint>
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#include <string>
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#include <utility>
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#include "absl/algorithm/container.h"
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#include "api/video/video_bitrate_allocation.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/string_encode.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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const std::vector<RidDescription>& rids = sp.rids();
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RTC_DCHECK(rids.size() == 0 || rids.size() == encoding_count);
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for (size_t i = 0; i < rids.size(); ++i) {
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encodings[i].rid = rids[i].rid;
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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 CheckRtpParametersValues(
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const webrtc::RtpParameters& rtp_parameters) {
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using webrtc::RTCErrorType;
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for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
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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].scale_resolution_down_by &&
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*rtp_parameters.encodings[i].scale_resolution_down_by < 1.0) {
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LOG_AND_RETURN_ERROR(
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RTCErrorType::INVALID_RANGE,
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"Attempted to set RtpParameters scale_resolution_down_by to an "
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"invalid number. scale_resolution_down_by must be >= 1.0");
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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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webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
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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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if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
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[](const webrtc::RtpEncodingParameters& encoding1,
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const webrtc::RtpEncodingParameters& encoding2) {
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return encoding1.rid == encoding2.rid;
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})) {
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
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"Attempted to change RID values in the encodings.");
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}
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if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
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[](const webrtc::RtpEncodingParameters& encoding1,
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const webrtc::RtpEncodingParameters& encoding2) {
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return encoding1.ssrc == encoding2.ssrc;
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})) {
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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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return CheckRtpParametersValues(rtp_parameters);
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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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