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 |