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Unified Diff: webrtc/config.cc

Issue 3004723002: Move RtpExtension to api/ directory and config.h/.cc to call/. (Closed)
Patch Set: Fix nit. Created 3 years, 3 months ago
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Index: webrtc/config.cc
diff --git a/webrtc/config.cc b/webrtc/config.cc
deleted file mode 100644
index 19a9a96079dc3b90b09cb12ac43fa916a7ccaf4b..0000000000000000000000000000000000000000
--- a/webrtc/config.cc
+++ /dev/null
@@ -1,283 +0,0 @@
-/*
- * Copyright (c) 2014 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 "webrtc/config.h"
-
-#include <algorithm>
-#include <sstream>
-#include <string>
-
-#include "webrtc/rtc_base/checks.h"
-
-namespace webrtc {
-std::string NackConfig::ToString() const {
- std::stringstream ss;
- ss << "{rtp_history_ms: " << rtp_history_ms;
- ss << '}';
- return ss.str();
-}
-
-std::string UlpfecConfig::ToString() const {
- std::stringstream ss;
- ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
- ss << ", red_payload_type: " << red_payload_type;
- ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
- ss << '}';
- return ss.str();
-}
-
-bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
- return ulpfec_payload_type == other.ulpfec_payload_type &&
- red_payload_type == other.red_payload_type &&
- red_rtx_payload_type == other.red_rtx_payload_type;
-}
-
-std::string RtpExtension::ToString() const {
- std::stringstream ss;
- ss << "{uri: " << uri;
- ss << ", id: " << id;
- if (encrypt) {
- ss << ", encrypt";
- }
- ss << '}';
- return ss.str();
-}
-
-const char RtpExtension::kAudioLevelUri[] =
- "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
-const int RtpExtension::kAudioLevelDefaultId = 1;
-
-const char RtpExtension::kTimestampOffsetUri[] =
- "urn:ietf:params:rtp-hdrext:toffset";
-const int RtpExtension::kTimestampOffsetDefaultId = 2;
-
-const char RtpExtension::kAbsSendTimeUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
-const int RtpExtension::kAbsSendTimeDefaultId = 3;
-
-const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation";
-const int RtpExtension::kVideoRotationDefaultId = 4;
-
-const char RtpExtension::kTransportSequenceNumberUri[] =
- "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
-const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
-
-// This extension allows applications to adaptively limit the playout delay
-// on frames as per the current needs. For example, a gaming application
-// has very different needs on end-to-end delay compared to a video-conference
-// application.
-const char RtpExtension::kPlayoutDelayUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
-const int RtpExtension::kPlayoutDelayDefaultId = 6;
-
-const char RtpExtension::kVideoContentTypeUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
-const int RtpExtension::kVideoContentTypeDefaultId = 7;
-
-const char RtpExtension::kVideoTimingUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
-const int RtpExtension::kVideoTimingDefaultId = 8;
-
-const char RtpExtension::kEncryptHeaderExtensionsUri[] =
- "urn:ietf:params:rtp-hdrext:encrypt";
-
-const int RtpExtension::kMinId = 1;
-const int RtpExtension::kMaxId = 14;
-
-bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
- return uri == webrtc::RtpExtension::kAudioLevelUri ||
- uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
-}
-
-bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
- return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
- uri == webrtc::RtpExtension::kAbsSendTimeUri ||
- uri == webrtc::RtpExtension::kVideoRotationUri ||
- uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
- uri == webrtc::RtpExtension::kPlayoutDelayUri ||
- uri == webrtc::RtpExtension::kVideoContentTypeUri ||
- uri == webrtc::RtpExtension::kVideoTimingUri;
-}
-
-bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
- return uri == webrtc::RtpExtension::kAudioLevelUri ||
- uri == webrtc::RtpExtension::kTimestampOffsetUri ||
-#if !defined(ENABLE_EXTERNAL_AUTH)
- // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
- // here and filter out later if external auth is really used in
- // srtpfilter. External auth is used by Chromium and replaces the
- // extension header value of "kAbsSendTimeUri", so it must not be
- // encrypted (which can't be done by Chromium).
- uri == webrtc::RtpExtension::kAbsSendTimeUri ||
-#endif
- uri == webrtc::RtpExtension::kVideoRotationUri ||
- uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
- uri == webrtc::RtpExtension::kPlayoutDelayUri ||
- uri == webrtc::RtpExtension::kVideoContentTypeUri;
-}
-
-const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
- const std::vector<RtpExtension>& extensions,
- const std::string& uri) {
- for (const auto& extension : extensions) {
- if (extension.uri == uri) {
- return &extension;
- }
- }
- return nullptr;
-}
-
-std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted(
- const std::vector<RtpExtension>& extensions) {
- std::vector<RtpExtension> filtered;
- for (auto extension = extensions.begin(); extension != extensions.end();
- ++extension) {
- if (extension->encrypt) {
- filtered.push_back(*extension);
- continue;
- }
-
- // Only add non-encrypted extension if no encrypted with the same URI
- // is also present...
- if (std::find_if(extension + 1, extensions.end(),
- [extension](const RtpExtension& check) {
- return extension->uri == check.uri;
- }) != extensions.end()) {
- continue;
- }
-
- // ...and has not been added before.
- if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
- filtered.push_back(*extension);
- }
- }
- return filtered;
-}
-
-VideoStream::VideoStream()
- : width(0),
- height(0),
- max_framerate(-1),
- min_bitrate_bps(-1),
- target_bitrate_bps(-1),
- max_bitrate_bps(-1),
- max_qp(-1) {}
-
-VideoStream::~VideoStream() = default;
-
-std::string VideoStream::ToString() const {
- std::stringstream ss;
- ss << "{width: " << width;
- ss << ", height: " << height;
- ss << ", max_framerate: " << max_framerate;
- ss << ", min_bitrate_bps:" << min_bitrate_bps;
- ss << ", target_bitrate_bps:" << target_bitrate_bps;
- ss << ", max_bitrate_bps:" << max_bitrate_bps;
- ss << ", max_qp: " << max_qp;
-
- ss << ", temporal_layer_thresholds_bps: [";
- for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
- ss << temporal_layer_thresholds_bps[i];
- if (i != temporal_layer_thresholds_bps.size() - 1)
- ss << ", ";
- }
- ss << ']';
-
- ss << '}';
- return ss.str();
-}
-
-VideoEncoderConfig::VideoEncoderConfig()
- : content_type(ContentType::kRealtimeVideo),
- encoder_specific_settings(nullptr),
- min_transmit_bitrate_bps(0),
- max_bitrate_bps(0),
- number_of_streams(0) {}
-
-VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default;
-
-VideoEncoderConfig::~VideoEncoderConfig() = default;
-
-std::string VideoEncoderConfig::ToString() const {
- std::stringstream ss;
- ss << "{content_type: ";
- switch (content_type) {
- case ContentType::kRealtimeVideo:
- ss << "kRealtimeVideo";
- break;
- case ContentType::kScreen:
- ss << "kScreenshare";
- break;
- }
- ss << ", encoder_specific_settings: ";
- ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
-
- ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
- ss << '}';
- return ss.str();
-}
-
-VideoEncoderConfig::VideoEncoderConfig(const VideoEncoderConfig&) = default;
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillEncoderSpecificSettings(
- VideoCodec* codec) const {
- if (codec->codecType == kVideoCodecH264) {
- FillVideoCodecH264(codec->H264());
- } else if (codec->codecType == kVideoCodecVP8) {
- FillVideoCodecVp8(codec->VP8());
- } else if (codec->codecType == kVideoCodecVP9) {
- FillVideoCodecVp9(codec->VP9());
- } else {
- RTC_NOTREACHED() << "Encoder specifics set/used for unknown codec type.";
- }
-}
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecH264(
- VideoCodecH264* h264_settings) const {
- RTC_NOTREACHED();
-}
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp8(
- VideoCodecVP8* vp8_settings) const {
- RTC_NOTREACHED();
-}
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp9(
- VideoCodecVP9* vp9_settings) const {
- RTC_NOTREACHED();
-}
-
-VideoEncoderConfig::H264EncoderSpecificSettings::H264EncoderSpecificSettings(
- const VideoCodecH264& specifics)
- : specifics_(specifics) {}
-
-void VideoEncoderConfig::H264EncoderSpecificSettings::FillVideoCodecH264(
- VideoCodecH264* h264_settings) const {
- *h264_settings = specifics_;
-}
-
-VideoEncoderConfig::Vp8EncoderSpecificSettings::Vp8EncoderSpecificSettings(
- const VideoCodecVP8& specifics)
- : specifics_(specifics) {}
-
-void VideoEncoderConfig::Vp8EncoderSpecificSettings::FillVideoCodecVp8(
- VideoCodecVP8* vp8_settings) const {
- *vp8_settings = specifics_;
-}
-
-VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
- const VideoCodecVP9& specifics)
- : specifics_(specifics) {}
-
-void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
- VideoCodecVP9* vp9_settings) const {
- *vp9_settings = specifics_;
-}
-
-} // namespace webrtc
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