Fix circular dependencies in webrtc_common.
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}diff --git a/.gn b/.gn
index ac4ef80..5e3c6ca 100644
--- a/.gn
+++ b/.gn
@@ -21,6 +21,7 @@
# their includes checked for proper dependencies when you run either
# "gn check" or "gn gen --check".
check_targets = [
+ ":webrtc_common",
"//api/*",
"//audio/*",
"//backup/*",
diff --git a/BUILD.gn b/BUILD.gn
index 6153e4f..9b4248b 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -350,11 +350,24 @@
}
}
+rtc_source_set("typedefs") {
+ sources = [
+ "typedefs.h",
+ ]
+}
+
rtc_static_library("webrtc_common") {
sources = [
"common_types.cc",
"common_types.h",
- "typedefs.h",
+ ]
+ deps = [
+ ":typedefs",
+ "api:array_view",
+ "api:optional",
+ "rtc_base:checks",
+ "rtc_base:deprecation",
+ "rtc_base:stringutils",
]
if (!build_with_chromium && is_clang) {
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 592039f..94dd865 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -59,6 +59,8 @@
"proxy.h",
"rtcerror.cc",
"rtcerror.h",
+ "rtp_headers.cc",
+ "rtp_headers.h",
"rtpparameters.cc",
"rtpparameters.h",
"rtpreceiverinterface.h",
@@ -84,6 +86,7 @@
]
deps = [
+ ":array_view",
":optional",
":rtc_stats_api",
":video_frame_api",
@@ -93,10 +96,14 @@
# Basically, don't add stuff here. You might break sensitive downstream
# targets like pnacl. API should not depend on anything outside of this
# file, really. All these should arguably go away in time.
+ "..:typedefs",
"..:webrtc_common",
"../modules/audio_processing:audio_processing_statistics",
+ "../rtc_base:checks",
+ "../rtc_base:deprecation",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
+ "../rtc_base:stringutils",
]
# This is needed until bugs.webrtc.org/7504 is removed so this target can
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index d2c074d..01dc124 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -29,7 +29,9 @@
"..:array_view",
"..:optional",
"../..:webrtc_common",
+ "../../:typedefs",
"../../rtc_base:checks",
+ "../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:sanitizer",
]
diff --git a/api/ortc/rtptransportinterface.h b/api/ortc/rtptransportinterface.h
index 3c58fad..716a297 100644
--- a/api/ortc/rtptransportinterface.h
+++ b/api/ortc/rtptransportinterface.h
@@ -16,6 +16,7 @@
#include "api/optional.h"
#include "api/ortc/packettransportinterface.h"
#include "api/rtcerror.h"
+#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
diff --git a/api/rtp_headers.cc b/api/rtp_headers.cc
new file mode 100644
index 0000000..a0b1a15
--- /dev/null
+++ b/api/rtp_headers.cc
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2017 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 "api/rtp_headers.h"
+
+#include <string.h>
+#include <algorithm>
+#include <limits>
+#include <type_traits>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/stringutils.h"
+
+namespace webrtc {
+
+RTPHeaderExtension::RTPHeaderExtension()
+ : hasTransmissionTimeOffset(false),
+ transmissionTimeOffset(0),
+ hasAbsoluteSendTime(false),
+ absoluteSendTime(0),
+ hasTransportSequenceNumber(false),
+ transportSequenceNumber(0),
+ hasAudioLevel(false),
+ voiceActivity(false),
+ audioLevel(0),
+ hasVideoRotation(false),
+ videoRotation(kVideoRotation_0),
+ hasVideoContentType(false),
+ videoContentType(VideoContentType::UNSPECIFIED),
+ has_video_timing(false) {}
+
+RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
+ default;
+
+RTPHeaderExtension& RTPHeaderExtension::operator=(
+ const RTPHeaderExtension& other) = default;
+
+RTPHeader::RTPHeader()
+ : markerBit(false),
+ payloadType(0),
+ sequenceNumber(0),
+ timestamp(0),
+ ssrc(0),
+ numCSRCs(0),
+ arrOfCSRCs(),
+ paddingLength(0),
+ headerLength(0),
+ payload_type_frequency(0),
+ extension() {}
+
+RTPHeader::RTPHeader(const RTPHeader& other) = default;
+
+RTPHeader& RTPHeader::operator=(const RTPHeader& other) = default;
+
+} // namespace webrtc
diff --git a/api/rtp_headers.h b/api/rtp_headers.h
new file mode 100644
index 0000000..2624a33
--- /dev/null
+++ b/api/rtp_headers.h
@@ -0,0 +1,178 @@
+/*
+ * Copyright (c) 2017 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.
+ */
+
+#ifndef API_RTP_HEADERS_H_
+#define API_RTP_HEADERS_H_
+
+#include <stddef.h>
+#include <string.h>
+#include <ostream>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/optional.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/deprecation.h"
+#include "common_types.h" // NOLINT(build/include)
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+// Class to represent the value of RTP header extensions that are
+// variable-length strings (e.g., RtpStreamId and RtpMid).
+// Unlike std::string, it can be copied with memcpy and cleared with memset.
+//
+// Empty value represents unset header extension (use empty() to query).
+class StringRtpHeaderExtension {
+ public:
+ // String RTP header extensions are limited to 16 bytes because it is the
+ // maximum length that can be encoded with one-byte header extensions.
+ static constexpr size_t kMaxSize = 16;
+
+ static bool IsLegalName(rtc::ArrayView<const char> name);
+
+ StringRtpHeaderExtension() { value_[0] = 0; }
+ explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
+ Set(value.data(), value.size());
+ }
+ StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default;
+ StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) =
+ default;
+
+ bool empty() const { return value_[0] == 0; }
+ const char* data() const { return value_; }
+ size_t size() const { return strnlen(value_, kMaxSize); }
+
+ void Set(rtc::ArrayView<const uint8_t> value) {
+ Set(reinterpret_cast<const char*>(value.data()), value.size());
+ }
+ void Set(const char* data, size_t size);
+
+ friend bool operator==(const StringRtpHeaderExtension& lhs,
+ const StringRtpHeaderExtension& rhs) {
+ return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
+ }
+ friend bool operator!=(const StringRtpHeaderExtension& lhs,
+ const StringRtpHeaderExtension& rhs) {
+ return !(lhs == rhs);
+ }
+
+ private:
+ char value_[kMaxSize];
+};
+
+// StreamId represents RtpStreamId which is a string.
+typedef StringRtpHeaderExtension StreamId;
+
+// Mid represents RtpMid which is a string.
+typedef StringRtpHeaderExtension Mid;
+
+struct RTPHeaderExtension {
+ RTPHeaderExtension();
+ RTPHeaderExtension(const RTPHeaderExtension& other);
+ RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
+
+ bool hasTransmissionTimeOffset;
+ int32_t transmissionTimeOffset;
+ bool hasAbsoluteSendTime;
+ uint32_t absoluteSendTime;
+ bool hasTransportSequenceNumber;
+ uint16_t transportSequenceNumber;
+
+ // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
+ // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
+ bool hasAudioLevel;
+ bool voiceActivity;
+ uint8_t audioLevel;
+
+ // For Coordination of Video Orientation. See
+ // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+ // ts_126114v120700p.pdf
+ bool hasVideoRotation;
+ VideoRotation videoRotation;
+
+ // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
+ // a corresponding bool flag.
+ bool hasVideoContentType;
+ VideoContentType videoContentType;
+
+ bool has_video_timing;
+ VideoSendTiming video_timing;
+
+ PlayoutDelay playout_delay = {-1, -1};
+
+ // For identification of a stream when ssrc is not signaled. See
+ // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
+ // TODO(danilchap): Update url from draft to release version.
+ StreamId stream_id;
+ StreamId repaired_stream_id;
+
+ // For identifying the media section used to interpret this RTP packet. See
+ // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
+ Mid mid;
+};
+
+struct RTPHeader {
+ RTPHeader();
+ RTPHeader(const RTPHeader& other);
+ RTPHeader& operator=(const RTPHeader& other);
+
+ bool markerBit;
+ uint8_t payloadType;
+ uint16_t sequenceNumber;
+ uint32_t timestamp;
+ uint32_t ssrc;
+ uint8_t numCSRCs;
+ uint32_t arrOfCSRCs[kRtpCsrcSize];
+ size_t paddingLength;
+ size_t headerLength;
+ int payload_type_frequency;
+ RTPHeaderExtension extension;
+};
+
+// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
+// RTCP mode is described by RFC 5506.
+enum class RtcpMode { kOff, kCompound, kReducedSize };
+
+enum NetworkState {
+ kNetworkUp,
+ kNetworkDown,
+};
+
+struct RtpKeepAliveConfig final {
+ // If no packet has been sent for |timeout_interval_ms|, send a keep-alive
+ // packet. The keep-alive packet is an empty (no payload) RTP packet with a
+ // payload type of 20 as long as the other end has not negotiated the use of
+ // this value. If this value has already been negotiated, then some other
+ // unused static payload type from table 5 of RFC 3551 shall be used and set
+ // in |payload_type|.
+ int64_t timeout_interval_ms = -1;
+ uint8_t payload_type = 20;
+
+ bool operator==(const RtpKeepAliveConfig& o) const {
+ return timeout_interval_ms == o.timeout_interval_ms &&
+ payload_type == o.payload_type;
+ }
+ bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
+};
+
+// Currently only VP8/VP9 specific.
+struct RtpPayloadState {
+ int16_t picture_id = -1;
+};
+
+} // namespace webrtc
+
+#endif // API_RTP_HEADERS_H_
diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
index f3cb7f1..6f3623d 100644
--- a/api/video_codecs/BUILD.gn
+++ b/api/video_codecs/BUILD.gn
@@ -26,6 +26,7 @@
"..:optional",
"..:video_frame_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_video",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
diff --git a/audio/utility/BUILD.gn b/audio/utility/BUILD.gn
index f38e0eb..4e01710 100644
--- a/audio/utility/BUILD.gn
+++ b/audio/utility/BUILD.gn
@@ -21,6 +21,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../modules:module_api",
"../../modules/audio_coding:audio_format_conversion",
"../../rtc_base:checks",
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 82f6ca1..6f936b8 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -24,6 +24,7 @@
":rtp_interfaces",
":video_stream_api",
"..:webrtc_common",
+ "../:typedefs",
"../api:audio_mixer_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
@@ -70,6 +71,7 @@
":rtp_interfaces",
"..:webrtc_common",
"../api:array_view",
+ "../api:libjingle_peerconnection_api",
"../api:optional",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
@@ -176,6 +178,7 @@
"../api:optional",
"../api:transport_api",
"../common_video:common_video",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
]
}
@@ -204,6 +207,7 @@
":rtp_sender",
"..:webrtc_common",
"../api:array_view",
+ "../api:libjingle_peerconnection_api",
"../api:mock_audio_mixer",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../logging:rtc_event_log_api",
diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h
index a0b7a11..98ce351 100644
--- a/call/flexfec_receive_stream.h
+++ b/call/flexfec_receive_stream.h
@@ -16,6 +16,7 @@
#include <string>
#include <vector>
+#include "api/rtp_headers.h"
#include "api/call/transport.h"
#include "api/rtpparameters.h"
#include "call/rtp_packet_sink_interface.h"
diff --git a/call/rtcp_demuxer.cc b/call/rtcp_demuxer.cc
index 0adbdf3..0e78dde 100644
--- a/call/rtcp_demuxer.cc
+++ b/call/rtcp_demuxer.cc
@@ -10,6 +10,7 @@
#include "call/rtcp_demuxer.h"
+#include "api/rtp_headers.h"
#include "call/rtcp_packet_sink_interface.h"
#include "call/rtp_rtcp_demuxer_helper.h"
#include "common_types.h" // NOLINT(build/include)
diff --git a/call/rtcp_demuxer_unittest.cc b/call/rtcp_demuxer_unittest.cc
index dd5aa55..0e1c95b 100644
--- a/call/rtcp_demuxer_unittest.cc
+++ b/call/rtcp_demuxer_unittest.cc
@@ -13,6 +13,7 @@
#include <memory>
#include <set>
+#include "api/rtp_headers.h"
#include "call/rtcp_packet_sink_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index de6f712..924d69a 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -16,12 +16,16 @@
#include <string>
#include <vector>
+#include "api/rtp_headers.h"
#include "api/call/transport.h"
#include "api/rtpparameters.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_timing.h"
#include "call/rtp_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "common_video/include/frame_callback.h"
#include "media/base/videosinkinterface.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/platform_file.h"
namespace webrtc {
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index e57307d..890a767 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -18,12 +18,14 @@
#include "api/call/transport.h"
#include "api/rtpparameters.h"
+#include "api/rtp_headers.h"
#include "call/rtp_config.h"
#include "call/video_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "common_video/include/frame_callback.h"
#include "media/base/videosinkinterface.h"
#include "media/base/videosourceinterface.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/platform_file.h"
namespace webrtc {
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index f601731..27287ec 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -62,6 +62,7 @@
deps = [
":sinc_resampler",
"..:webrtc_common",
+ "../:typedefs",
"../api:optional",
"../rtc_base:checks",
"../rtc_base:gtest_prod",
@@ -223,6 +224,7 @@
":common_audio_c_arm_asm",
":common_audio_cc",
"..:webrtc_common",
+ "../:typedefs",
"../rtc_base:checks",
"../rtc_base:compile_assert_c",
"../rtc_base:rtc_base_approved",
@@ -241,6 +243,7 @@
public_configs = [ ":common_audio_config" ]
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
]
@@ -252,6 +255,7 @@
]
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../rtc_base:gtest_prod",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
@@ -447,6 +451,7 @@
":fir_filter_factory",
":sinc_resampler",
"..:webrtc_common",
+ "../:typedefs",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
diff --git a/common_types.cc b/common_types.cc
index f1b2f1e..07f77b7 100644
--- a/common_types.cc
+++ b/common_types.cc
@@ -20,66 +20,6 @@
namespace webrtc {
-StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
-
-constexpr size_t StreamId::kMaxSize;
-
-bool StreamId::IsLegalName(rtc::ArrayView<const char> name) {
- return (name.size() <= kMaxSize && name.size() > 0 &&
- std::all_of(name.data(), name.data() + name.size(), isalnum));
-}
-
-void StreamId::Set(const char* data, size_t size) {
- // If |data| contains \0, the stream id size might become less than |size|.
- RTC_CHECK_LE(size, kMaxSize);
- memcpy(value_, data, size);
- if (size < kMaxSize)
- value_[size] = 0;
-}
-
-// StreamId is used as member of RTPHeader that is sometimes copied with memcpy
-// and thus assume trivial destructibility.
-static_assert(std::is_trivially_destructible<StreamId>::value, "");
-
-RTPHeaderExtension::RTPHeaderExtension()
- : hasTransmissionTimeOffset(false),
- transmissionTimeOffset(0),
- hasAbsoluteSendTime(false),
- absoluteSendTime(0),
- hasTransportSequenceNumber(false),
- transportSequenceNumber(0),
- hasAudioLevel(false),
- voiceActivity(false),
- audioLevel(0),
- hasVideoRotation(false),
- videoRotation(kVideoRotation_0),
- hasVideoContentType(false),
- videoContentType(VideoContentType::UNSPECIFIED),
- has_video_timing(false) {}
-
-RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
- default;
-
-RTPHeaderExtension& RTPHeaderExtension::operator=(
- const RTPHeaderExtension& other) = default;
-
-RTPHeader::RTPHeader()
- : markerBit(false),
- payloadType(0),
- sequenceNumber(0),
- timestamp(0),
- ssrc(0),
- numCSRCs(0),
- arrOfCSRCs(),
- paddingLength(0),
- headerLength(0),
- payload_type_frequency(0),
- extension() {}
-
-RTPHeader::RTPHeader(const RTPHeader& other) = default;
-
-RTPHeader& RTPHeader::operator=(const RTPHeader& other) = default;
-
VideoCodec::VideoCodec()
: codecType(kVideoCodecUnknown),
plName(),
diff --git a/common_types.h b/common_types.h
index 21fdf56..9f1ad8f 100644
--- a/common_types.h
+++ b/common_types.h
@@ -19,9 +19,6 @@
#include "api/array_view.h"
#include "api/optional.h"
-#include "api/video/video_content_type.h"
-#include "api/video/video_rotation.h"
-#include "api/video/video_timing.h"
#include "rtc_base/checks.h"
#include "rtc_base/deprecation.h"
#include "typedefs.h" // NOLINT(build/include)
@@ -685,244 +682,6 @@
int max_ms;
};
-// Class to represent the value of RTP header extensions that are
-// variable-length strings (e.g., RtpStreamId and RtpMid).
-// Unlike std::string, it can be copied with memcpy and cleared with memset.
-//
-// Empty value represents unset header extension (use empty() to query).
-class StringRtpHeaderExtension {
- public:
- // String RTP header extensions are limited to 16 bytes because it is the
- // maximum length that can be encoded with one-byte header extensions.
- static constexpr size_t kMaxSize = 16;
-
- static bool IsLegalName(rtc::ArrayView<const char> name);
-
- StringRtpHeaderExtension() { value_[0] = 0; }
- explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
- Set(value.data(), value.size());
- }
- StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default;
- StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) =
- default;
-
- bool empty() const { return value_[0] == 0; }
- const char* data() const { return value_; }
- size_t size() const { return strnlen(value_, kMaxSize); }
-
- void Set(rtc::ArrayView<const uint8_t> value) {
- Set(reinterpret_cast<const char*>(value.data()), value.size());
- }
- void Set(const char* data, size_t size);
-
- friend bool operator==(const StringRtpHeaderExtension& lhs,
- const StringRtpHeaderExtension& rhs) {
- return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
- }
- friend bool operator!=(const StringRtpHeaderExtension& lhs,
- const StringRtpHeaderExtension& rhs) {
- return !(lhs == rhs);
- }
-
- private:
- char value_[kMaxSize];
-};
-
-// StreamId represents RtpStreamId which is a string.
-typedef StringRtpHeaderExtension StreamId;
-
-// Mid represents RtpMid which is a string.
-typedef StringRtpHeaderExtension Mid;
-
-struct RTPHeaderExtension {
- RTPHeaderExtension();
- RTPHeaderExtension(const RTPHeaderExtension& other);
- RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
-
- bool hasTransmissionTimeOffset;
- int32_t transmissionTimeOffset;
- bool hasAbsoluteSendTime;
- uint32_t absoluteSendTime;
- bool hasTransportSequenceNumber;
- uint16_t transportSequenceNumber;
-
- // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
- // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
- bool hasAudioLevel;
- bool voiceActivity;
- uint8_t audioLevel;
-
- // For Coordination of Video Orientation. See
- // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
- // ts_126114v120700p.pdf
- bool hasVideoRotation;
- VideoRotation videoRotation;
-
- // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
- // a corresponding bool flag.
- bool hasVideoContentType;
- VideoContentType videoContentType;
-
- bool has_video_timing;
- VideoSendTiming video_timing;
-
- PlayoutDelay playout_delay = {-1, -1};
-
- // For identification of a stream when ssrc is not signaled. See
- // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
- // TODO(danilchap): Update url from draft to release version.
- StreamId stream_id;
- StreamId repaired_stream_id;
-
- // For identifying the media section used to interpret this RTP packet. See
- // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
- Mid mid;
-};
-
-struct RTPHeader {
- RTPHeader();
- RTPHeader(const RTPHeader& other);
- RTPHeader& operator=(const RTPHeader& other);
-
- bool markerBit;
- uint8_t payloadType;
- uint16_t sequenceNumber;
- uint32_t timestamp;
- uint32_t ssrc;
- uint8_t numCSRCs;
- uint32_t arrOfCSRCs[kRtpCsrcSize];
- size_t paddingLength;
- size_t headerLength;
- int payload_type_frequency;
- RTPHeaderExtension extension;
-};
-
-struct RtpPacketCounter {
- RtpPacketCounter()
- : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
-
- void Add(const RtpPacketCounter& other) {
- header_bytes += other.header_bytes;
- payload_bytes += other.payload_bytes;
- padding_bytes += other.padding_bytes;
- packets += other.packets;
- }
-
- void Subtract(const RtpPacketCounter& other) {
- RTC_DCHECK_GE(header_bytes, other.header_bytes);
- header_bytes -= other.header_bytes;
- RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
- payload_bytes -= other.payload_bytes;
- RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
- padding_bytes -= other.padding_bytes;
- RTC_DCHECK_GE(packets, other.packets);
- packets -= other.packets;
- }
-
- void AddPacket(size_t packet_length, const RTPHeader& header) {
- ++packets;
- header_bytes += header.headerLength;
- padding_bytes += header.paddingLength;
- payload_bytes +=
- packet_length - (header.headerLength + header.paddingLength);
- }
-
- size_t TotalBytes() const {
- return header_bytes + payload_bytes + padding_bytes;
- }
-
- size_t header_bytes; // Number of bytes used by RTP headers.
- size_t payload_bytes; // Payload bytes, excluding RTP headers and padding.
- size_t padding_bytes; // Number of padding bytes.
- uint32_t packets; // Number of packets.
-};
-
-// Data usage statistics for a (rtp) stream.
-struct StreamDataCounters {
- StreamDataCounters();
-
- void Add(const StreamDataCounters& other) {
- transmitted.Add(other.transmitted);
- retransmitted.Add(other.retransmitted);
- fec.Add(other.fec);
- if (other.first_packet_time_ms != -1 &&
- (other.first_packet_time_ms < first_packet_time_ms ||
- first_packet_time_ms == -1)) {
- // Use oldest time.
- first_packet_time_ms = other.first_packet_time_ms;
- }
- }
-
- void Subtract(const StreamDataCounters& other) {
- transmitted.Subtract(other.transmitted);
- retransmitted.Subtract(other.retransmitted);
- fec.Subtract(other.fec);
- if (other.first_packet_time_ms != -1 &&
- (other.first_packet_time_ms > first_packet_time_ms ||
- first_packet_time_ms == -1)) {
- // Use youngest time.
- first_packet_time_ms = other.first_packet_time_ms;
- }
- }
-
- int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
- return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
- }
-
- // Returns the number of bytes corresponding to the actual media payload (i.e.
- // RTP headers, padding, retransmissions and fec packets are excluded).
- // Note this function does not have meaning for an RTX stream.
- size_t MediaPayloadBytes() const {
- return transmitted.payload_bytes - retransmitted.payload_bytes -
- fec.payload_bytes;
- }
-
- int64_t first_packet_time_ms; // Time when first packet is sent/received.
- RtpPacketCounter transmitted; // Number of transmitted packets/bytes.
- RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes.
- RtpPacketCounter fec; // Number of redundancy packets/bytes.
-};
-
-// Callback, called whenever byte/packet counts have been updated.
-class StreamDataCountersCallback {
- public:
- virtual ~StreamDataCountersCallback() {}
-
- virtual void DataCountersUpdated(const StreamDataCounters& counters,
- uint32_t ssrc) = 0;
-};
-
-// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
-// RTCP mode is described by RFC 5506.
-enum class RtcpMode { kOff, kCompound, kReducedSize };
-
-enum NetworkState {
- kNetworkUp,
- kNetworkDown,
-};
-
-struct RtpKeepAliveConfig final {
- // If no packet has been sent for |timeout_interval_ms|, send a keep-alive
- // packet. The keep-alive packet is an empty (no payload) RTP packet with a
- // payload type of 20 as long as the other end has not negotiated the use of
- // this value. If this value has already been negotiated, then some other
- // unused static payload type from table 5 of RFC 3551 shall be used and set
- // in |payload_type|.
- int64_t timeout_interval_ms = -1;
- uint8_t payload_type = 20;
-
- bool operator==(const RtpKeepAliveConfig& o) const {
- return timeout_interval_ms == o.timeout_interval_ms &&
- payload_type == o.payload_type;
- }
- bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
-};
-
-// Currently only VP8/VP9 specific.
-struct RtpPayloadState {
- int16_t picture_id = -1;
-};
-
} // namespace webrtc
#endif // COMMON_TYPES_H_
diff --git a/common_video/include/video_frame.h b/common_video/include/video_frame.h
index 188d5e5..0c145fa 100644
--- a/common_video/include/video_frame.h
+++ b/common_video/include/video_frame.h
@@ -16,6 +16,7 @@
// to refactor and clean up related interfaces, at which point it
// should be moved to somewhere under api/.
+#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "typedefs.h" // NOLINT(build/include)
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index 0ab9bc8..4e733d0 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -509,6 +509,7 @@
deps = [
"../api:video_frame_api_i420",
"../rtc_base:checks",
+ "../rtc_base:stringutils",
]
if (is_win) {
sources += [
@@ -569,6 +570,7 @@
deps = [
"..:webrtc_common",
"../rtc_base:rtc_base_approved",
+ "../rtc_base:stringutils",
"../rtc_tools:command_line_parser",
]
if (!build_with_chromium && is_clang) {
@@ -727,6 +729,7 @@
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
+ "../rtc_base:stringutils",
"../system_wrappers:field_trial_default",
]
}
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index bfa20aa..981231b 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -71,6 +71,7 @@
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:array_view",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
diff --git a/logging/rtc_event_log/rtc_stream_config.h b/logging/rtc_event_log/rtc_stream_config.h
index e43ee7e..771ba66 100644
--- a/logging/rtc_event_log/rtc_stream_config.h
+++ b/logging/rtc_event_log/rtc_stream_config.h
@@ -14,6 +14,7 @@
#include <string>
#include <vector>
+#include "api/rtp_headers.h"
#include "api/rtpparameters.h"
#include "common_types.h" // NOLINT(build/include)
diff --git a/media/BUILD.gn b/media/BUILD.gn
index ac03a7d..2271339 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -57,6 +57,7 @@
deps = [
"../rtc_base:checks",
"../rtc_base:sanitizer",
+ "../rtc_base:stringutils",
]
public_deps = []
sources = [
@@ -133,6 +134,7 @@
"../call:call_interfaces",
"../modules/video_coding:video_coding_utility",
"../rtc_base:checks",
+ "../rtc_base:stringutils",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
"//third_party/libyuv",
@@ -347,6 +349,7 @@
"../modules/video_coding:video_coding_utility",
"../p2p:rtc_p2p",
"../rtc_base:checks",
+ "../rtc_base:stringutils",
]
sources = [
"base/fakemediaengine.h",
@@ -438,6 +441,7 @@
"../pc:rtc_pc",
"../pc:rtc_pc_base",
"../rtc_base:checks",
+ "../rtc_base:stringutils",
"../test:field_trial",
]
sources = [
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 91fe33e..5768944 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -21,6 +21,7 @@
#include "api/optional.h"
#include "api/rtpparameters.h"
#include "api/rtpreceiverinterface.h"
+#include "api/video/video_content_type.h"
#include "api/video/video_timing.h"
#include "call/video_config.h"
#include "media/base/codec.h"
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 2c1a042..33394f4 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -34,6 +34,7 @@
]
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:optional",
]
}
@@ -46,9 +47,12 @@
deps = [
":module_api_public",
"..:webrtc_common",
+ "../:typedefs",
+ "../api:libjingle_peerconnection_api",
"../api:optional",
"../api:video_frame_api",
"../api:video_frame_api_i420",
+ "../rtc_base:deprecation",
"../rtc_base:rtc_base_approved",
"video_coding:codec_globals_headers",
]
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 9c674d4..6365d56 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -65,6 +65,7 @@
"acm2/rent_a_codec.h",
]
deps = [
+ "../../:typedefs",
"../../rtc_base:checks",
"../../api:array_view",
"../../api:optional",
@@ -96,6 +97,7 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
]
}
@@ -132,6 +134,8 @@
}
deps = audio_coding_deps + [
+ "../../rtc_base:deprecation",
+ "../../:typedefs",
"../../rtc_base:checks",
"../../system_wrappers:metrics_api",
"..:module_api",
@@ -177,6 +181,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
@@ -241,6 +246,7 @@
]
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
]
}
@@ -282,6 +288,7 @@
]
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
]
}
@@ -461,6 +468,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../rtc_base:checks",
@@ -569,6 +577,7 @@
deps = [
":isac_common",
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:compile_assert_c",
@@ -675,6 +684,7 @@
deps = [
":isac_common",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../rtc_base:checks",
@@ -799,6 +809,7 @@
public_configs = [ ":pcm16b_config" ]
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
]
}
@@ -856,6 +867,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
@@ -1042,6 +1054,8 @@
":neteq_decoder_enum",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
+ "../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
@@ -1084,6 +1098,8 @@
":neteq",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
+ "../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -1121,7 +1137,9 @@
":pcm16b",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
+ "../../api:libjingle_peerconnection_api",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@@ -1169,6 +1187,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
@@ -1195,6 +1214,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@@ -1308,6 +1328,7 @@
":pcm16b_c",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:rtc_base_approved",
@@ -1339,6 +1360,7 @@
":neteq_test_support",
":neteq_test_tools",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
@@ -1411,6 +1433,7 @@
":audio_coding_module_typedefs",
":audio_format_conversion",
"..:module_api",
+ "../../:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../rtc_base:rtc_base_approved",
@@ -1441,6 +1464,7 @@
":audio_coding",
":audio_format_conversion",
"..:module_api",
+ "../../:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../rtc_base:rtc_base_approved",
@@ -1519,6 +1543,7 @@
defines = []
deps = [
"..:module_api",
+ "../../:typedefs",
"../../rtc_base:checks",
]
sources = [
@@ -1567,7 +1592,9 @@
rtc_test("audio_codec_speed_tests") {
testonly = true
defines = []
- deps = []
+ deps = [
+ "../../:typedefs",
+ ]
sources = [
"codecs/isac/fix/test/isac_speed_test.cc",
"codecs/opus/opus_speed_test.cc",
@@ -1626,6 +1653,7 @@
":pcm16b",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
@@ -1653,6 +1681,7 @@
":neteq_test_tools",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@@ -1665,6 +1694,7 @@
testonly = true
deps = audio_coding_deps + [
+ "../../:typedefs",
":audio_coding",
":neteq_input_audio_tools",
"../../api/audio_codecs/g711:audio_encoder_g711",
@@ -1688,6 +1718,7 @@
testonly = true
deps = audio_coding_deps + [
+ "../../:typedefs",
"../../system_wrappers:system_wrappers_default",
"../rtp_rtcp:rtp_rtcp_format",
"../../api:array_view",
@@ -1768,6 +1799,7 @@
":neteq",
":neteq_test_support",
"../..:webrtc_common",
+ "../../:typedefs",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@@ -1915,6 +1947,7 @@
deps = [
":g722",
"../..:webrtc_common",
+ "../../:typedefs",
]
}
@@ -2098,6 +2131,7 @@
":webrtc_opus",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 00ed7cd..0d47f21 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -18,6 +18,7 @@
#include "api/audio_codecs/audio_decoder.h"
#include "api/optional.h"
+#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
#include "rtc_base/constructormagic.h"
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h
index a72f7d1..94d45c5 100644
--- a/modules/audio_coding/neteq/tools/packet.h
+++ b/modules/audio_coding/neteq/tools/packet.h
@@ -14,6 +14,7 @@
#include <list>
#include <memory>
+#include "api/rtp_headers.h" // NOLINT(build/include)
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h
index b705f83..3b3cca9 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -11,6 +11,7 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 3174ee9..6c19146 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -100,6 +100,7 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../common_audio",
"../../rtc_base:checks",
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index cffd1a8..1318398 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -36,6 +36,7 @@
":audio_frame_manipulator",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api:audio_mixer_api",
"../../audio/utility:audio_frame_operations",
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index b190f02..d879076 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -226,10 +226,12 @@
":audio_processing_statistics",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api:optional",
"../../audio/utility:audio_frame_operations",
"../../rtc_base:checks",
+ "../../rtc_base:deprecation",
"../../rtc_base:gtest_prod",
"../../rtc_base:protobuf_utils",
"../../rtc_base:sanitizer",
@@ -341,6 +343,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@@ -371,6 +374,7 @@
"../../api:array_view",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
+ "../../rtc_base:stringutils",
]
defines = []
if (apm_debug_dump) {
@@ -404,6 +408,7 @@
":apm_logging",
":audio_processing_statistics",
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_audio:common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@@ -582,6 +587,7 @@
":audioproc_test_utils",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api:optional",
"../../common_audio:common_audio",
@@ -768,6 +774,7 @@
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
+ "../../rtc_base:stringutils",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@@ -819,6 +826,7 @@
":audio_processing",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@@ -838,6 +846,7 @@
deps = [
":audio_processing",
"../..:webrtc_common",
+ "../../:typedefs",
"../../system_wrappers",
"../../system_wrappers:metrics_default",
]
@@ -892,6 +901,7 @@
deps = [
":audioproc_debug_proto",
"../..:webrtc_common",
+ "../../:typedefs",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
]
diff --git a/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 9cedd6e..00e4730 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -45,6 +45,7 @@
]
deps = [
"../../../..:webrtc_common",
+ "../../../../:typedefs",
"../../../../api:array_view",
"../../../../common_audio",
"../../../../rtc_base:checks",
@@ -65,6 +66,7 @@
deps = [
":lib",
"../../../..:webrtc_common",
+ "../../../../:typedefs",
"../../../../api:array_view",
"../../../../api:optional",
"../../../../common_audio",
diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn
index 58d2e25..c014934 100644
--- a/modules/congestion_controller/BUILD.gn
+++ b/modules/congestion_controller/BUILD.gn
@@ -49,6 +49,7 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index 277683a..8e8d92d 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -28,6 +28,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
]
@@ -89,6 +90,7 @@
":desktop_capture_mock",
":primitives",
"../..:webrtc_common",
+ "../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:cpu_features_api",
@@ -333,6 +335,7 @@
deps = [
":primitives",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:refcountedbase",
"../../rtc_base:checks",
"../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
diff --git a/modules/include/module_common_types.h b/modules/include/module_common_types.h
index 1d89c7d..94ccbdc 100644
--- a/modules/include/module_common_types.h
+++ b/modules/include/module_common_types.h
@@ -18,6 +18,7 @@
#include <limits>
#include "api/optional.h"
+#include "api/rtp_headers.h"
#include "api/video/video_rotation.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types_public.h"
diff --git a/modules/media_file/BUILD.gn b/modules/media_file/BUILD.gn
index a3f3312..589b281 100644
--- a/modules/media_file/BUILD.gn
+++ b/modules/media_file/BUILD.gn
@@ -35,6 +35,7 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_audio",
"../../rtc_base:rtc_base_approved",
]
diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
index 428a99e..6b76c64 100644
--- a/modules/pacing/BUILD.gn
+++ b/modules/pacing/BUILD.gn
@@ -34,6 +34,7 @@
deps = [
"..:module_api",
+ "../../:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../logging:rtc_event_log_api",
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index f0fac54..a5dd014 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -46,6 +46,7 @@
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../modules:module_api",
"../../modules/rtp_rtcp:rtp_rtcp_format",
@@ -140,6 +141,7 @@
":remote_bitrate_estimator",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../rtc_base:checks",
"../../rtc_base:gtest_prod",
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index e3f35b8..4ae32cd 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -82,12 +82,14 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../common_video",
"../../rtc_base:checks",
+ "../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
@@ -192,6 +194,7 @@
":rtp_rtcp_format",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
@@ -200,11 +203,13 @@
"../../common_video",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
+ "../../rtc_base:deprecation",
"../../rtc_base:gtest_prod",
"../../rtc_base:rate_limiter",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:sequenced_task_checker",
+ "../../rtc_base:stringutils",
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
@@ -398,6 +403,7 @@
":rtp_rtcp_format",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
diff --git a/modules/rtp_rtcp/include/receive_statistics.h b/modules/rtp_rtcp/include/receive_statistics.h
index ee7dbbb..55c2c05 100644
--- a/modules/rtp_rtcp/include/receive_statistics.h
+++ b/modules/rtp_rtcp/include/receive_statistics.h
@@ -17,6 +17,7 @@
#include "modules/include/module.h"
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.cc b/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
index af3714b..d9b082b 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
@@ -12,6 +12,27 @@
namespace webrtc {
+StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
+
+constexpr size_t StreamId::kMaxSize;
+
+bool StreamId::IsLegalName(rtc::ArrayView<const char> name) {
+ return (name.size() <= kMaxSize && name.size() > 0 &&
+ std::all_of(name.data(), name.data() + name.size(), isalnum));
+}
+
+void StreamId::Set(const char* data, size_t size) {
+ // If |data| contains \0, the stream id size might become less than |size|.
+ RTC_CHECK_LE(size, kMaxSize);
+ memcpy(value_, data, size);
+ if (size < kMaxSize)
+ value_[size] = 0;
+}
+
+// StreamId is used as member of RTPHeader that is sometimes copied with memcpy
+// and thus assume trivial destructibility.
+static_assert(std::is_trivially_destructible<StreamId>::value, "");
+
PayloadUnion::PayloadUnion(const AudioPayload& payload)
: audio_payload_(payload) {}
PayloadUnion::PayloadUnion(const VideoPayload& payload)
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 37ef332..97a2a98 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -16,6 +16,7 @@
#include <vector>
#include "api/audio_codecs/audio_format.h"
+#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types.h"
#include "rtc_base/deprecation.h"
@@ -496,5 +497,100 @@
virtual uint16_t AllocateSequenceNumber() = 0;
};
+struct RtpPacketCounter {
+ RtpPacketCounter()
+ : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
+
+ void Add(const RtpPacketCounter& other) {
+ header_bytes += other.header_bytes;
+ payload_bytes += other.payload_bytes;
+ padding_bytes += other.padding_bytes;
+ packets += other.packets;
+ }
+
+ void Subtract(const RtpPacketCounter& other) {
+ RTC_DCHECK_GE(header_bytes, other.header_bytes);
+ header_bytes -= other.header_bytes;
+ RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
+ payload_bytes -= other.payload_bytes;
+ RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
+ padding_bytes -= other.padding_bytes;
+ RTC_DCHECK_GE(packets, other.packets);
+ packets -= other.packets;
+ }
+
+ void AddPacket(size_t packet_length, const RTPHeader& header) {
+ ++packets;
+ header_bytes += header.headerLength;
+ padding_bytes += header.paddingLength;
+ payload_bytes +=
+ packet_length - (header.headerLength + header.paddingLength);
+ }
+
+ size_t TotalBytes() const {
+ return header_bytes + payload_bytes + padding_bytes;
+ }
+
+ size_t header_bytes; // Number of bytes used by RTP headers.
+ size_t payload_bytes; // Payload bytes, excluding RTP headers and padding.
+ size_t padding_bytes; // Number of padding bytes.
+ uint32_t packets; // Number of packets.
+};
+
+// Data usage statistics for a (rtp) stream.
+struct StreamDataCounters {
+ StreamDataCounters();
+
+ void Add(const StreamDataCounters& other) {
+ transmitted.Add(other.transmitted);
+ retransmitted.Add(other.retransmitted);
+ fec.Add(other.fec);
+ if (other.first_packet_time_ms != -1 &&
+ (other.first_packet_time_ms < first_packet_time_ms ||
+ first_packet_time_ms == -1)) {
+ // Use oldest time.
+ first_packet_time_ms = other.first_packet_time_ms;
+ }
+ }
+
+ void Subtract(const StreamDataCounters& other) {
+ transmitted.Subtract(other.transmitted);
+ retransmitted.Subtract(other.retransmitted);
+ fec.Subtract(other.fec);
+ if (other.first_packet_time_ms != -1 &&
+ (other.first_packet_time_ms > first_packet_time_ms ||
+ first_packet_time_ms == -1)) {
+ // Use youngest time.
+ first_packet_time_ms = other.first_packet_time_ms;
+ }
+ }
+
+ int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
+ return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
+ }
+
+ // Returns the number of bytes corresponding to the actual media payload (i.e.
+ // RTP headers, padding, retransmissions and fec packets are excluded).
+ // Note this function does not have meaning for an RTX stream.
+ size_t MediaPayloadBytes() const {
+ return transmitted.payload_bytes - retransmitted.payload_bytes -
+ fec.payload_bytes;
+ }
+
+ int64_t first_packet_time_ms; // Time when first packet is sent/received.
+ RtpPacketCounter transmitted; // Number of transmitted packets/bytes.
+ RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes.
+ RtpPacketCounter fec; // Number of redundancy packets/bytes.
+};
+
+// Callback, called whenever byte/packet counts have been updated.
+class StreamDataCountersCallback {
+ public:
+ virtual ~StreamDataCountersCallback() {}
+
+ virtual void DataCountersUpdated(const StreamDataCounters& counters,
+ uint32_t ssrc) = 0;
+};
+
} // namespace webrtc
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index 7002fd2..e3de364 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -32,6 +32,7 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../audio/utility:audio_frame_operations",
"../../common_audio",
"../../rtc_base:checks",
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index f5dfa05..2953053 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -28,10 +28,12 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:video_frame_api_i420",
"../../common_video",
"../../media:rtc_media_base",
"../../rtc_base:rtc_base_approved",
+ "../../rtc_base:stringutils",
"../../system_wrappers",
"//third_party/libyuv",
]
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 4e4ab75..02b0a1b 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -99,6 +99,7 @@
"..:module_api",
"..:module_api_public",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../api:video_frame_api_i420",
"../../call:video_stream_api",
@@ -127,6 +128,7 @@
deps = [
":video_coding_utility",
"../..:webrtc_common",
+ "../../:typedefs",
"../../test:test_support",
]
}
@@ -169,6 +171,7 @@
deps = [
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
@@ -239,6 +242,7 @@
deps = [
":video_coding_utility",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:video_frame_api_i420",
"../../common_video:common_video",
"../../rtc_base:checks",
@@ -301,6 +305,7 @@
":video_coding_utility",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
@@ -431,6 +436,7 @@
":video_coding_utility",
":webrtc_vp8",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:video_frame_api_i420",
"../../api/video_codecs:video_codecs_api",
"../../common_video:common_video",
@@ -599,6 +605,7 @@
":webrtc_vp9",
"..:module_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:video_frame_api",
"../../api:video_frame_api_i420",
"../../api/video_codecs:video_codecs_api",
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index e6b9156..cd54b7d 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -96,6 +96,7 @@
"../api:ortc_api",
"../rtc_base:checks",
"../rtc_base:rtc_base",
+ "../rtc_base:stringutils",
"../system_wrappers:field_trial_api",
]
@@ -183,6 +184,7 @@
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
+ "../rtc_base:stringutils",
"../test:test_support",
"//testing/gmock",
"//testing/gtest",
@@ -204,6 +206,7 @@
deps = [
":rtc_p2p",
"..:webrtc_common",
+ "../:typedefs",
"../rtc_base:checks",
"../rtc_base:rtc_base",
]
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 225f7f4..536dd23 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -78,6 +78,7 @@
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_task_queue",
+ "../rtc_base:stringutils",
]
if (rtc_build_libsrtp) {
@@ -186,6 +187,7 @@
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
+ "../rtc_base:stringutils",
"../stats",
"../system_wrappers",
"../system_wrappers:field_trial_api",
@@ -441,6 +443,7 @@
":peerconnection",
":rtc_pc_base",
"../rtc_base:checks",
+ "../rtc_base:stringutils",
]
if (is_android) {
deps += [ ":android_black_magic" ]
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index e64e444..0949f4e 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -96,13 +96,18 @@
]
deps = [
":safe_compare",
- "..:webrtc_common",
+ "..:typedefs",
]
}
-rtc_source_set("type_traits") {
+rtc_source_set("rate_limiter") {
sources = [
- "type_traits.h",
+ "rate_limiter.cc",
+ "rate_limiter.h",
+ ]
+ deps = [
+ ":rtc_base_approved",
+ "../system_wrappers",
]
}
@@ -121,14 +126,25 @@
]
}
-rtc_source_set("rate_limiter") {
+rtc_source_set("stringutils") {
sources = [
- "rate_limiter.cc",
- "rate_limiter.h",
+ "stringutils.cc",
+ "stringutils.h",
]
deps = [
- ":rtc_base_approved",
- "../system_wrappers",
+ ":checks",
+ ]
+}
+
+rtc_source_set("type_traits") {
+ sources = [
+ "type_traits.h",
+ ]
+}
+
+rtc_source_set("deprecation") {
+ sources = [
+ "deprecation.h",
]
}
@@ -146,7 +162,9 @@
deps = [
":checks",
":safe_compare",
+ ":stringutils",
":type_traits",
+ "../:typedefs",
]
sources = [
@@ -171,7 +189,6 @@
"copyonwritebuffer.h",
"criticalsection.cc",
"criticalsection.h",
- "deprecation.h",
"event_tracer.cc",
"event_tracer.h",
"file.cc",
@@ -216,8 +233,6 @@
"stringencode.cc",
"stringencode.h",
"stringize_macros.h",
- "stringutils.cc",
- "stringutils.h",
"swap_queue.h",
"template_util.h",
"thread_annotations.h",
@@ -526,6 +541,7 @@
defines = []
deps = [
":checks",
+ ":stringutils",
"..:webrtc_common",
"../api:optional",
]
@@ -845,6 +861,7 @@
deps = [
":checks",
":rtc_base",
+ ":stringutils",
"../test:field_trial",
"../test:test_support",
]
@@ -956,6 +973,7 @@
":rtc_base_tests_utils",
":rtc_task_queue",
":safe_compare",
+ ":stringutils",
"../api:array_view",
"../system_wrappers:system_wrappers",
"../test:test_support",
@@ -1080,6 +1098,7 @@
":checks",
":rtc_base_tests_main",
":rtc_base_tests_utils",
+ ":stringutils",
"../api:optional",
"../test:test_support",
]
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 2b6d850..771d6d9 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -157,6 +157,7 @@
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../common_video",
]
}
@@ -379,6 +380,7 @@
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../common_audio",
"../modules/audio_processing",
"../modules/audio_processing:audioproc_debug_proto",
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 8657203..3faca81 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -401,6 +401,7 @@
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
+ "../../rtc_base:stringutils",
"../../system_wrappers:field_trial_api",
]
}
diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn
index dc2baf6..f999e21 100644
--- a/system_wrappers/BUILD.gn
+++ b/system_wrappers/BUILD.gn
@@ -54,6 +54,7 @@
":field_trial_api",
":metrics_api",
"..:webrtc_common",
+ "../:typedefs",
"../api:optional",
"../modules:module_api_public",
"../rtc_base:checks",
@@ -112,6 +113,7 @@
]
deps = [
"..:webrtc_common",
+ "../:typedefs",
]
}
@@ -215,6 +217,7 @@
":metrics_default",
":system_wrappers",
"..:webrtc_common",
+ "../:typedefs",
"../rtc_base:rtc_base_approved",
"../test:test_main",
"//testing/gtest",
diff --git a/system_wrappers/include/ntp_time.h b/system_wrappers/include/ntp_time.h
index e963ecf..1c32184 100644
--- a/system_wrappers/include/ntp_time.h
+++ b/system_wrappers/include/ntp_time.h
@@ -12,6 +12,8 @@
#include <stdint.h>
+#include "rtc_base/numerics/safe_conversions.h"
+
namespace webrtc {
class NtpTime {
@@ -40,8 +42,12 @@
// NTP standard (RFC1305, section 3.1) explicitly state value 0 is invalid.
bool Valid() const { return value_ != 0; }
- uint32_t seconds() const { return value_ / kFractionsPerSecond; }
- uint32_t fractions() const { return value_ % kFractionsPerSecond; }
+ uint32_t seconds() const {
+ return rtc::dchecked_cast<uint32_t>(value_ / kFractionsPerSecond);
+ }
+ uint32_t fractions() const {
+ return rtc::dchecked_cast<uint32_t>(value_ % kFractionsPerSecond);
+ }
private:
uint64_t value_;
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 98c74e9..8841ae0 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -60,6 +60,7 @@
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:optional",
"../api:video_frame_api_i420",
"../api/video_codecs:video_codecs_api",
@@ -151,6 +152,7 @@
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:array_view",
"../common_video",
"../rtc_base:gtest_prod",
@@ -215,6 +217,7 @@
":test_support",
":video_test_common",
"..:webrtc_common",
+ "../:typedefs",
"../api:video_frame_api",
"../api:video_frame_api_i420",
"../common_video",
@@ -362,6 +365,7 @@
"testsupport/iosfileutils.mm",
]
deps = [
+ "..:typedefs",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
@@ -384,6 +388,7 @@
]
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:optional",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
@@ -470,6 +475,7 @@
}
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:transport_api",
"../call",
"../call:call_interfaces",
@@ -507,6 +513,7 @@
}
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:array_view",
"../common_audio:common_audio",
"../modules/audio_device:audio_device",
@@ -565,6 +572,7 @@
":test_support",
":video_test_common",
"..:webrtc_common",
+ "../:typedefs",
"../api:transport_api",
"../api:video_frame_api",
"../api:video_frame_api_i420",
@@ -668,6 +676,7 @@
deps = [
":test_support",
"..:webrtc_common",
+ "../:typedefs",
"../common_video",
"../media:rtc_media_base",
"../modules/media_file",
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 475aeaa..19d1302 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -245,6 +245,7 @@
]
deps = [
"../..:webrtc_common",
+ "../../:typedefs",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../modules/rtp_rtcp:rtp_rtcp_format",
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 95b6e31..fe03c9a 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -55,6 +55,7 @@
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api:optional",
"../api:transport_api",
"../api:video_frame_api_i420",
@@ -216,6 +217,7 @@
]
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../common_video",
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 31d95fd..e993931 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -52,6 +52,7 @@
deps = [
":audio_level",
"..:webrtc_common",
+ "../:typedefs",
"../api:array_view",
"../api:audio_mixer_api",
"../api:call_api",
@@ -94,6 +95,7 @@
deps = [
"..:webrtc_common",
+ "../:typedefs",
"../common_audio",
"../modules:module_api",
"../rtc_base:rtc_base_approved",