[Cleanup] Add missing #include. Remove useless ones.

This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}

api/audio/BUILD.gn

@@ -40,6 +40,7 @@ rtc_source_set("aec3_config") {
     "echo_canceller3_config.h",
   ]
   deps = [
+    "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:safe_minmax",
     "../../rtc_base/system:rtc_export",

api/audio/echo_canceller3_config.cc

@@ -12,7 +12,7 @@
 #include <algorithm>
 #include <cmath>
 
-#include "rtc_base/logging.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
 namespace webrtc {

api/audio_codecs/audio_codec_pair_id.cc

@@ -11,6 +11,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 
 #include <atomic>
+#include <cstdint>
 
 #include "rtc_base/checks.h"
 

api/audio_codecs/audio_decoder.h

@@ -11,6 +11,8 @@
 #ifndef API_AUDIO_CODECS_AUDIO_DECODER_H_
 #define API_AUDIO_CODECS_AUDIO_DECODER_H_
 
+#include <stddef.h>
+#include <stdint.h>
 #include <memory>
 #include <vector>
 

api/audio_codecs/audio_format.h

@@ -11,6 +11,7 @@
 #ifndef API_AUDIO_CODECS_AUDIO_FORMAT_H_
 #define API_AUDIO_CODECS_AUDIO_FORMAT_H_
 
+#include <stddef.h>
 #include <map>
 #include <string>
 

api/audio_options.h

@@ -11,6 +11,7 @@
 #ifndef API_AUDIO_OPTIONS_H_
 #define API_AUDIO_OPTIONS_H_
 
+#include <stdint.h>
 #include <string>
 
 #include "absl/types/optional.h"

api/call/transport.cc

@@ -10,6 +10,8 @@
 
 #include "api/call/transport.h"
 
+#include <cstdint>
+
 namespace webrtc {
 
 PacketOptions::PacketOptions() = default;

api/candidate.cc

@@ -11,6 +11,7 @@
 #include "api/candidate.h"
 
 #include "rtc_base/helpers.h"
+#include "rtc_base/ipaddress.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace cricket {

api/crypto/cryptooptions.h

@@ -12,7 +12,6 @@
 #define API_CRYPTO_CRYPTOOPTIONS_H_
 
 #include <vector>
-#include "absl/types/optional.h"
 
 namespace webrtc {
 

api/datachannelinterface.h

@@ -14,6 +14,8 @@
 #ifndef API_DATACHANNELINTERFACE_H_
 #define API_DATACHANNELINTERFACE_H_
 
+#include <stddef.h>
+#include <stdint.h>
 #include <string>
 
 #include "rtc_base/checks.h"

api/jsepicecandidate.cc

@@ -10,6 +10,9 @@
 
 #include "api/jsepicecandidate.h"
 
+#include <algorithm>
+#include <utility>
+
 namespace webrtc {
 
 std::string JsepIceCandidate::sdp_mid() const {

api/jsepicecandidate.h

@@ -14,6 +14,7 @@
 #ifndef API_JSEPICECANDIDATE_H_
 #define API_JSEPICECANDIDATE_H_
 
+#include <stddef.h>
 #include <string>
 #include <vector>
 

api/media_transport_interface.cc

@@ -17,6 +17,9 @@
 
 #include "api/media_transport_interface.h"
 
+#include <cstdint>
+#include <utility>
+
 namespace webrtc {
 
 MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}

api/media_transport_interface.h

@@ -21,6 +21,7 @@
 #include <utility>
 #include <vector>
 
+#include "api/array_view.h"
 #include "api/rtcerror.h"
 #include "api/video/encoded_image.h"
 #include "common_types.h"  // NOLINT(build/include)

api/mediaconstraintsinterface.cc

@@ -10,7 +10,9 @@
 
 #include "api/mediaconstraintsinterface.h"
 
+#include "absl/types/optional.h"
 #include "api/peerconnectioninterface.h"
+#include "media/base/mediaconfig.h"
 #include "rtc_base/stringencode.h"
 
 namespace {

api/mediaconstraintsinterface.h

@@ -21,9 +21,11 @@
 #ifndef API_MEDIACONSTRAINTSINTERFACE_H_
 #define API_MEDIACONSTRAINTSINTERFACE_H_
 
+#include <stddef.h>
 #include <string>
 #include <vector>
 
+#include "api/audio_options.h"
 #include "api/peerconnectioninterface.h"
 
 namespace webrtc {

api/rtp_headers.h

@@ -12,6 +12,7 @@
 #define API_RTP_HEADERS_H_
 
 #include <stddef.h>
+#include <stdint.h>
 #include <string.h>
 
 #include "api/array_view.h"
@@ -19,7 +20,6 @@
 #include "api/video/video_frame_marking.h"
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
-
 #include "common_types.h"  // NOLINT(build/include)
 
 namespace webrtc {

api/rtpparameters.cc

@@ -12,6 +12,7 @@
 #include <algorithm>
 #include <string>
 
+#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {

api/rtpparameters.h

@@ -11,6 +11,7 @@
 #ifndef API_RTPPARAMETERS_H_
 #define API_RTPPARAMETERS_H_
 
+#include <stdint.h>
 #include <string>
 #include <unordered_map>
 #include <vector>

api/rtptransceiverinterface.cc

@@ -10,6 +10,8 @@
 
 #include "api/rtptransceiverinterface.h"
 
+#include "rtc_base/checks.h"
+
 namespace webrtc {
 
 RtpTransceiverInit::RtpTransceiverInit() = default;

api/rtptransceiverinterface.h

@@ -16,9 +16,12 @@
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
+#include "api/mediatypes.h"
+#include "api/rtpparameters.h"
 #include "api/rtpreceiverinterface.h"
 #include "api/rtpsenderinterface.h"
 #include "rtc_base/refcount.h"
+#include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
 

api/units/data_rate.h

@@ -20,12 +20,12 @@
 #include <cmath>
 #include <limits>
 #include <string>
-
-#include "rtc_base/checks.h"
-#include "rtc_base/numerics/safe_conversions.h"
+#include <type_traits>
 
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
 
 namespace webrtc {
 namespace data_rate_impl {

api/units/time_delta.h

@@ -20,6 +20,7 @@
 #include <cstdlib>
 #include <limits>
 #include <string>
+#include <type_traits>
 
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"

api/units/timestamp.h

@@ -15,9 +15,11 @@
 #include <ostream>  // no-presubmit-check TODO(webrtc:8982)
 #endif              // UNIT_TEST
 
+#include <math.h>
 #include <stdint.h>
 #include <limits>
 #include <string>
+#include <type_traits>
 
 #include "api/units/time_delta.h"
 #include "rtc_base/checks.h"

api/video/BUILD.gn

@@ -75,6 +75,9 @@ rtc_source_set("encoded_image") {
   deps = [
     ":video_frame",
     "../..:webrtc_common",
+
+    # TODO(bugs.webrtc.org/8311) Cleanup when kMaxSpatialLayers is moved to proper include.
+    "../../api/video:video_bitrate_allocation",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",

api/video/encoded_image.cc

@@ -12,11 +12,6 @@
 
 #include <string.h>
 
-#include <algorithm>  // swap
-
-#include "rtc_base/bind.h"
-#include "rtc_base/checks.h"
-
 namespace webrtc {
 
 // FFmpeg's decoder, used by H264DecoderImpl, requires up to 8 bytes padding due

api/video/encoded_image.h

@@ -11,11 +11,15 @@
 #ifndef API_VIDEO_ENCODED_IMAGE_H_
 #define API_VIDEO_ENCODED_IMAGE_H_
 
+#include <stdint.h>
+
 #include "absl/types/optional.h"
+#include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
 #include "common_types.h"  // NOLINT(build/include)
+#include "rtc_base/checks.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {

api/video/i420_buffer.h

@@ -11,11 +11,13 @@
 #ifndef API_VIDEO_I420_BUFFER_H_
 #define API_VIDEO_I420_BUFFER_H_
 
+#include <stdint.h>
 #include <memory>
 
 #include "api/video/video_frame_buffer.h"
 #include "api/video/video_rotation.h"
 #include "rtc_base/memory/aligned_malloc.h"
+#include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
 

api/video/video_bitrate_allocation.cc

@@ -10,6 +10,8 @@
 
 #include "api/video/video_bitrate_allocation.h"
 
+#include <cstdint>
+
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/strings/string_builder.h"

api/video/video_bitrate_allocation.h

@@ -11,6 +11,8 @@
 #ifndef API_VIDEO_VIDEO_BITRATE_ALLOCATION_H_
 #define API_VIDEO_VIDEO_BITRATE_ALLOCATION_H_
 
+#include <stddef.h>
+#include <stdint.h>
 #include <limits>
 #include <string>
 #include <vector>

api/video/video_frame.h

@@ -17,6 +17,7 @@
 #include "api/video/color_space.h"
 #include "api/video/video_frame_buffer.h"
 #include "api/video/video_rotation.h"
+#include "rtc_base/scoped_ref_ptr.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {

api/video_codecs/video_codec.cc

@@ -11,14 +11,11 @@
 #include "api/video_codecs/video_codec.h"
 
 #include <string.h>
-#include <algorithm>
-#include <limits>
 #include <string>
-#include <type_traits>
 
 #include "absl/strings/match.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/stringutils.h"
 
 namespace webrtc {
 

api/video_codecs/video_codec.h

@@ -11,8 +11,11 @@
 #ifndef API_VIDEO_CODECS_VIDEO_CODEC_H_
 #define API_VIDEO_CODECS_VIDEO_CODEC_H_
 
+#include <stddef.h>
+#include <stdint.h>
 #include <string>
 
+#include "api/video/video_bitrate_allocation.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "rtc_base/system/rtc_export.h"
 

api/video_codecs/video_encoder.cc

@@ -10,6 +10,10 @@
 
 #include "api/video_codecs/video_encoder.h"
 
+#include <string.h>
+
+#include "rtc_base/checks.h"
+
 namespace webrtc {
 
 // TODO(mflodman): Add default complexity for VP9 and VP9.

api/video_codecs/video_encoder_config.cc

@@ -9,7 +9,6 @@
  */
 #include "api/video_codecs/video_encoder_config.h"
 
-#include <algorithm>
 #include <string>
 
 #include "rtc_base/checks.h"

api/video_codecs/video_encoder_config.h

@@ -11,12 +11,14 @@
 #ifndef API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
 #define API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
 
+#include <stddef.h>
 #include <string>
 #include <vector>
 
 #include "absl/types/optional.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_codec.h"
+#include "common_types.h"  // NOLINT(build/include)
 #include "rtc_base/refcount.h"
 #include "rtc_base/scoped_ref_ptr.h"
 

audio/audio_receive_stream.cc

@@ -13,14 +13,19 @@
 #include <string>
 #include <utility>
 
+#include "absl/memory/memory.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_format.h"
 #include "api/call/audio_sink.h"
+#include "api/rtpparameters.h"
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
+#include "audio/channel_receive.h"
 #include "audio/channel_receive_proxy.h"
 #include "audio/conversion.h"
+#include "call/rtp_config.h"
 #include "call/rtp_stream_receiver_controller_interface.h"
-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "common_types.h"  // NOLINT(build/include)
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"

audio/audio_receive_stream.h

@@ -18,7 +18,6 @@
 #include "api/rtp_headers.h"
 #include "audio/audio_state.h"
 #include "call/audio_receive_stream.h"
-#include "call/rtp_packet_sink_interface.h"
 #include "call/syncable.h"
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/thread_checker.h"

audio/audio_send_stream.cc

@@ -15,12 +15,21 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/transport.h"
+#include "api/crypto/frameencryptorinterface.h"
 #include "audio/audio_state.h"
+#include "audio/channel_send.h"
 #include "audio/channel_send_proxy.h"
 #include "audio/conversion.h"
+#include "call/rtp_config.h"
 #include "call/rtp_transport_controller_send_interface.h"
+#include "common_audio/vad/include/vad.h"
+#include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_processing/include/audio_processing.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/function_view.h"

audio/audio_transport_impl.cc

@@ -17,7 +17,7 @@
 #include "audio/remix_resample.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "call/audio_send_stream.h"
-#include "rtc_base/logging.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 

audio/channel_receive.cc

@@ -26,6 +26,7 @@
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/pacing/packet_router.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"

audio/channel_receive_proxy.cc

@@ -14,7 +14,6 @@
 
 #include "api/call/audio_sink.h"
 #include "audio/channel_send_proxy.h"
-#include "call/rtp_transport_controller_send_interface.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"

audio/channel_send_proxy.cc

@@ -12,11 +12,8 @@
 
 #include <utility>
 
-#include "api/call/audio_sink.h"
 #include "call/rtp_transport_controller_send_interface.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/numerics/safe_minmax.h"
 
 namespace webrtc {
 namespace voe {

audio/null_audio_poller.cc

@@ -9,9 +9,13 @@
  */
 
 #include "audio/null_audio_poller.h"
-#include "rtc_base/logging.h"
+
+#include <stddef.h>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/location.h"
 #include "rtc_base/thread.h"
-#include "rtc_base/timeutils.h"  // for TimeMillis
+#include "rtc_base/timeutils.h"
 
 namespace webrtc {
 namespace internal {

audio/null_audio_poller.h

@@ -11,8 +11,11 @@
 #ifndef AUDIO_NULL_AUDIO_POLLER_H_
 #define AUDIO_NULL_AUDIO_POLLER_H_
 
+#include <stdint.h>
+
 #include "modules/audio_device/include/audio_device_defines.h"
 #include "rtc_base/messagehandler.h"
+#include "rtc_base/messagequeue.h"
 #include "rtc_base/thread_checker.h"
 
 namespace webrtc {

audio/remix_resample.cc

@@ -13,10 +13,7 @@
 #include "api/audio/audio_frame.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "common_audio/resampler/include/push_resampler.h"
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "common_types.h"  // NOLINT(build/include)
 #include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
 
 namespace webrtc {
 namespace voe {

audio/transport_feedback_packet_loss_tracker.cc

@@ -10,11 +10,11 @@
 
 #include "audio/transport_feedback_packet_loss_tracker.h"
 
+#include <iterator>
 #include <limits>
 #include <utility>
 
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/mod_ops.h"
 

audio/utility/audio_frame_operations.cc

@@ -12,6 +12,7 @@
 
 #include <string.h>
 #include <algorithm>
+#include <cstdint>
 
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"

audio/utility/audio_frame_operations.h

@@ -12,6 +12,7 @@
 #define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
 
 #include <stddef.h>
+#include <stdint.h>
 
 #include "api/audio/audio_frame.h"
 

call/BUILD.gn

@@ -41,6 +41,7 @@ rtc_source_set("call_interfaces") {
     "../modules/audio_processing:audio_processing",
     "../modules/audio_processing:audio_processing_statistics",
     "../rtc_base:audio_format_to_string",
+    "../rtc_base:checks",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
     "../rtc_base/network:sent_packet",

call/audio_send_stream.cc

@@ -9,6 +9,9 @@
  */
 
 #include "call/audio_send_stream.h"
+
+#include <stddef.h>
+
 #include "rtc_base/stringencode.h"
 #include "rtc_base/strings/audio_format_to_string.h"
 #include "rtc_base/strings/string_builder.h"

call/call_config.cc

@@ -10,6 +10,8 @@
 
 #include "call/call_config.h"
 
+#include "rtc_base/checks.h"
+
 namespace webrtc {
 
 CallConfig::CallConfig(RtcEventLog* event_log) : event_log(event_log) {

call/flexfec_receive_stream.cc

@@ -10,6 +10,8 @@
 
 #include "call/flexfec_receive_stream.h"
 
+#include "rtc_base/checks.h"
+
 namespace webrtc {
 
 FlexfecReceiveStream::Config::Config(Transport* rtcp_send_transport)

call/flexfec_receive_stream.h

@@ -12,7 +12,6 @@
 #define CALL_FLEXFEC_RECEIVE_STREAM_H_
 
 #include <stdint.h>
-
 #include <string>
 #include <vector>
 
@@ -20,7 +19,6 @@
 #include "api/rtp_headers.h"
 #include "api/rtpparameters.h"
 #include "call/rtp_packet_sink_interface.h"
-#include "common_types.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

call/rtp_config.cc

@@ -10,6 +10,9 @@
 
 #include "call/rtp_config.h"
 
+#include <cstdint>
+
+#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {

call/rtp_config.h

@@ -11,6 +11,8 @@
 #ifndef CALL_RTP_CONFIG_H_
 #define CALL_RTP_CONFIG_H_
 
+#include <stddef.h>
+#include <stdint.h>
 #include <string>
 #include <vector>
 

call/video_send_stream.cc

@@ -9,6 +9,9 @@
  */
 
 #include "call/video_send_stream.h"
+
+#include <utility>
+
 #include "api/crypto/frameencryptorinterface.h"
 #include "rtc_base/strings/string_builder.h"
 

call/video_send_stream.h

@@ -11,19 +11,21 @@
 #ifndef CALL_VIDEO_SEND_STREAM_H_
 #define CALL_VIDEO_SEND_STREAM_H_
 
+#include <stdint.h>
 #include <map>
 #include <string>
-#include <utility>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/call/transport.h"
 #include "api/crypto/cryptooptions.h"
+#include "api/rtpparameters.h"
+#include "api/video/video_content_type.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
 #include "api/video/video_stream_encoder_settings.h"
 #include "api/video_codecs/video_encoder_config.h"
-#include "api/video_codecs/video_encoder_factory.h"
 #include "call/rtp_config.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"

common_audio/audio_converter.h

@@ -11,6 +11,7 @@
 #ifndef COMMON_AUDIO_AUDIO_CONVERTER_H_
 #define COMMON_AUDIO_AUDIO_CONVERTER_H_
 
+#include <stddef.h>
 #include <memory>
 
 #include "rtc_base/constructormagic.h"

common_audio/channel_buffer.cc

@@ -10,6 +10,9 @@
 
 #include "common_audio/channel_buffer.h"
 
+#include <cstdint>
+
+#include "common_audio/include/audio_util.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {

common_audio/fir_filter_c.cc

@@ -11,11 +11,8 @@
 #include "common_audio/fir_filter_c.h"
 
 #include <string.h>
-
 #include <memory>
 
-#include "common_audio/fir_filter_neon.h"
-#include "common_audio/fir_filter_sse.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {

common_audio/fir_filter_factory.cc

@@ -13,12 +13,12 @@
 #include "common_audio/fir_filter_c.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/system/arch.h"
-#include "system_wrappers/include/cpu_features_wrapper.h"
 
 #if defined(WEBRTC_HAS_NEON)
 #include "common_audio/fir_filter_neon.h"
 #elif defined(WEBRTC_ARCH_X86_FAMILY)
 #include "common_audio/fir_filter_sse.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"  // kSSE2, WebRtc_G...
 #endif
 
 namespace webrtc {

common_audio/fir_filter_sse.h

@@ -11,6 +11,7 @@
 #ifndef COMMON_AUDIO_FIR_FILTER_SSE_H_
 #define COMMON_AUDIO_FIR_FILTER_SSE_H_
 
+#include <stddef.h>
 #include <memory>
 
 #include "common_audio/fir_filter.h"

common_audio/include/audio_util.h

@@ -11,6 +11,7 @@
 #ifndef COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
 #define COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
 
+#include <stdint.h>
 #include <algorithm>
 #include <cmath>
 #include <cstring>

common_audio/real_fourier.h

@@ -11,6 +11,7 @@
 #ifndef COMMON_AUDIO_REAL_FOURIER_H_
 #define COMMON_AUDIO_REAL_FOURIER_H_
 
+#include <stddef.h>
 #include <complex>
 #include <memory>
 

common_audio/real_fourier_ooura.h

@@ -11,6 +11,7 @@
 #ifndef COMMON_AUDIO_REAL_FOURIER_OOURA_H_
 #define COMMON_AUDIO_REAL_FOURIER_OOURA_H_
 
+#include <stddef.h>
 #include <complex>
 #include <memory>
 

common_audio/resampler/push_resampler.cc

@@ -10,12 +10,12 @@
 
 #include "common_audio/resampler/include/push_resampler.h"
 
+#include <stdint.h>
 #include <string.h>
 
 #include "absl/container/inlined_vector.h"
 #include "absl/memory/memory.h"
 #include "common_audio/include/audio_util.h"
-#include "common_audio/resampler/include/resampler.h"
 #include "common_audio/resampler/push_sinc_resampler.h"
 #include "rtc_base/checks.h"
 

common_audio/resampler/push_sinc_resampler.h

@@ -11,6 +11,8 @@
 #ifndef COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
 #define COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
 
+#include <stddef.h>
+#include <stdint.h>
 #include <memory>
 
 #include "common_audio/resampler/sinc_resampler.h"

common_audio/resampler/resampler.cc

@@ -12,6 +12,7 @@
  * A wrapper for resampling a numerous amount of sampling combinations.
  */
 
+#include <stdint.h>
 #include <stdlib.h>
 #include <string.h>
 

common_audio/resampler/sinc_resampler.cc

@@ -88,13 +88,13 @@
 #include "common_audio/resampler/sinc_resampler.h"
 
 #include <math.h>
+#include <stdint.h>
 #include <string.h>
-
 #include <limits>
 
 #include "rtc_base/checks.h"
 #include "rtc_base/system/arch.h"
-#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"  // kSSE2, WebRtc_G...
 
 namespace webrtc {
 

common_audio/resampler/sinc_resampler.h

@@ -14,6 +14,7 @@
 #ifndef COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
 #define COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
 
+#include <stddef.h>
 #include <memory>
 
 #include "rtc_base/constructormagic.h"

common_audio/resampler/sinc_resampler_sse.cc

@@ -11,10 +11,12 @@
 // Modified from the Chromium original:
 // src/media/base/simd/sinc_resampler_sse.cc
 
-#include "common_audio/resampler/sinc_resampler.h"
-
+#include <stddef.h>
+#include <stdint.h>
 #include <xmmintrin.h>
 
+#include "common_audio/resampler/sinc_resampler.h"
+
 namespace webrtc {
 
 float SincResampler::Convolve_SSE(const float* input_ptr,

common_audio/smoothing_filter.cc

@@ -12,6 +12,7 @@
 
 #include <cmath>
 
+#include "rtc_base/checks.h"
 #include "rtc_base/timeutils.h"
 
 namespace webrtc {

common_audio/smoothing_filter.h

@@ -11,9 +11,10 @@
 #ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_
 #define COMMON_AUDIO_SMOOTHING_FILTER_H_
 
+#include <stdint.h>
+
 #include "absl/types/optional.h"
 #include "rtc_base/constructormagic.h"
-#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 

common_audio/vad/vad.cc

@@ -12,6 +12,7 @@
 
 #include <memory>
 
+#include "common_audio/vad/include/webrtc_vad.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {

common_audio/wav_file.cc

@@ -10,15 +10,14 @@
 
 #include "common_audio/wav_file.h"
 
+#include <errno.h>
 #include <algorithm>
 #include <cstdio>
-#include <limits>
 
 #include "common_audio/include/audio_util.h"
 #include "common_audio/wav_header.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/system/arch.h"
 
 namespace webrtc {

common_audio/wav_header.cc

@@ -14,12 +14,10 @@
 
 #include "common_audio/wav_header.h"
 
-#include <algorithm>
 #include <cstring>
 #include <limits>
 #include <string>
 
-#include "common_audio/include/audio_util.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/system/arch.h"
 

common_types.h

@@ -11,12 +11,10 @@
 #ifndef COMMON_TYPES_H_
 #define COMMON_TYPES_H_
 
-#include <stddef.h>
-#include <string.h>
-#include <string>
-#include <vector>
+#include <stddef.h>  // For size_t
+#include <cstdint>
+
 #include "absl/strings/match.h"
-#include "api/array_view.h"
 // TODO(sprang): Remove this include when all usage includes it directly.
 #include "api/video/video_bitrate_allocation.h"
 #include "rtc_base/checks.h"

common_video/bitrate_adjuster.cc

@@ -13,7 +13,6 @@
 #include <algorithm>
 #include <cmath>
 
-#include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/timeutils.h"
 

common_video/h264/h264_bitstream_parser.cc

@@ -9,13 +9,12 @@
  */
 #include "common_video/h264/h264_bitstream_parser.h"
 
-#include <memory>
+#include <stdlib.h>
+#include <cstdint>
 #include <vector>
 
-#include "rtc_base/bitbuffer.h"
-#include "rtc_base/checks.h"
-
 #include "common_video/h264/h264_common.h"
+#include "rtc_base/bitbuffer.h"
 #include "rtc_base/logging.h"
 
 namespace {

common_video/h264/h264_bitstream_parser.h

@@ -17,10 +17,6 @@
 #include "common_video/h264/pps_parser.h"
 #include "common_video/h264/sps_parser.h"
 
-namespace rtc {
-class BitBufferWriter;
-}
-
 namespace webrtc {
 
 // Stateful H264 bitstream parser (due to SPS/PPS). Used to parse out QP values

common_video/h264/h264_common.cc

@@ -10,6 +10,8 @@
 
 #include "common_video/h264/h264_common.h"
 
+#include <cstdint>
+
 namespace webrtc {
 namespace H264 {
 

common_video/h264/h264_common.h

@@ -11,7 +11,8 @@
 #ifndef COMMON_VIDEO_H264_H264_COMMON_H_
 #define COMMON_VIDEO_H264_H264_COMMON_H_
 
-#include <memory>
+#include <stddef.h>
+#include <stdint.h>
 #include <vector>
 
 #include "rtc_base/buffer.h"

common_video/h264/pps_parser.cc

@@ -10,12 +10,12 @@
 
 #include "common_video/h264/pps_parser.h"
 
-#include <memory>
+#include <cstdint>
 #include <vector>
 
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/bitbuffer.h"
-#include "rtc_base/logging.h"
+#include "rtc_base/checks.h"
 
 #define RETURN_EMPTY_ON_FAIL(x) \
   if (!(x)) {                   \

common_video/h264/sps_parser.cc

@@ -10,12 +10,11 @@
 
 #include "common_video/h264/sps_parser.h"
 
-#include <memory>
+#include <cstdint>
 #include <vector>
 
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/bitbuffer.h"
-#include "rtc_base/logging.h"
 
 namespace {
 typedef absl::optional<webrtc::SpsParser::SpsState> OptionalSps;

common_video/h264/sps_vui_rewriter.cc

@@ -11,18 +11,17 @@
 
 #include "common_video/h264/sps_vui_rewriter.h"
 
-#include <algorithm>
-#include <memory>
+#include <string.h>
+#include <cstdint>
 #include <vector>
 
+#include "common_video/h264/h264_common.h"
+#include "common_video/h264/sps_parser.h"
 #include "rtc_base/bitbuffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
-#include "common_video/h264/h264_common.h"
-#include "common_video/h264/sps_parser.h"
-
 namespace webrtc {
 
 // The maximum expected growth from adding a VUI to the SPS. It's actually

common_video/h264/sps_vui_rewriter.h

@@ -12,14 +12,13 @@
 #ifndef COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
 #define COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
 
+#include <stddef.h>
+#include <stdint.h>
+
 #include "absl/types/optional.h"
 #include "common_video/h264/sps_parser.h"
 #include "rtc_base/buffer.h"
 
-namespace rtc {
-class BitBuffer;
-}
-
 namespace webrtc {
 
 // A class that can parse an SPS block of a NAL unit and if necessary

common_video/i420_buffer_pool.cc

@@ -10,6 +10,8 @@
 
 #include "common_video/include/i420_buffer_pool.h"
 
+#include <limits>
+
 #include "rtc_base/checks.h"
 
 namespace webrtc {

common_video/include/bitrate_adjuster.h

@@ -11,10 +11,13 @@
 #ifndef COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
 #define COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
 
-#include <functional>
+#include <stddef.h>
+#include <stdint.h>
 
+#include "absl/types/optional.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/rate_statistics.h"
+#include "rtc_base/thread_annotations.h"
 
 namespace webrtc {
 

common_video/include/i420_buffer_pool.h

@@ -11,12 +11,13 @@
 #ifndef COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
 #define COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
 
-#include <limits>
+#include <stddef.h>
 #include <list>
 
 #include "api/video/i420_buffer.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/refcountedobject.h"
+#include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
 

common_video/include/incoming_video_stream.h

@@ -11,6 +11,9 @@
 #ifndef COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
 #define COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
 
+#include <stdint.h>
+
+#include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "common_video/video_render_frames.h"
 #include "rtc_base/race_checker.h"

common_video/include/video_frame_buffer.h

@@ -11,10 +11,11 @@
 #ifndef COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
 #define COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
 
-#include <memory>
+#include <stdint.h>
 
 #include "api/video/video_frame_buffer.h"
 #include "rtc_base/callback.h"
+#include "rtc_base/refcountedobject.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {

common_video/incoming_video_stream.cc

@@ -11,9 +11,11 @@
 #include "common_video/include/incoming_video_stream.h"
 
 #include <memory>
+#include <utility>
 
+#include "absl/types/optional.h"
 #include "common_video/video_render_frames.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/trace_event.h"
 
 namespace webrtc {

common_video/libyuv/include/webrtc_libyuv.h

@@ -15,11 +15,14 @@
 #ifndef COMMON_VIDEO_LIBYUV_INCLUDE_WEBRTC_LIBYUV_H_
 #define COMMON_VIDEO_LIBYUV_INCLUDE_WEBRTC_LIBYUV_H_
 
+#include <stdint.h>
 #include <stdio.h>
 #include <vector>
 
 #include "api/video/video_frame.h"
-#include "common_types.h"  // NOLINT(build/include)  // VideoTypes.
+#include "api/video/video_frame_buffer.h"
+#include "common_types.h"  // NOLINT(build/include)
+#include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
 

common_video/video_render_frames.cc

@@ -10,8 +10,10 @@
 
 #include "common_video/video_render_frames.h"
 
+#include <type_traits>
 #include <utility>
 
+#include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/timeutils.h"
 #include "system_wrappers/include/metrics.h"

common_video/video_render_frames.h

@@ -11,8 +11,8 @@
 #ifndef COMMON_VIDEO_VIDEO_RENDER_FRAMES_H_
 #define COMMON_VIDEO_VIDEO_RENDER_FRAMES_H_
 
+#include <stddef.h>
 #include <stdint.h>
-
 #include <list>
 
 #include "absl/types/optional.h"

logging/rtc_event_log/events/rtc_event_audio_playout.h

@@ -11,6 +11,7 @@
 #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
 #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
 
+#include <stdint.h>
 #include <memory>
 
 #include "logging/rtc_event_log/events/rtc_event.h"

logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h

@@ -11,6 +11,7 @@
 #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_DELAY_BASED_H_
 #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_DELAY_BASED_H_
 
+#include <stdint.h>
 #include <memory>
 
 #include "logging/rtc_event_log/events/rtc_event.h"

logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h

@@ -11,6 +11,7 @@
 #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_LOSS_BASED_H_
 #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_LOSS_BASED_H_
 
+#include <stdint.h>
 #include <memory>
 
 #include "logging/rtc_event_log/events/rtc_event.h"

logging/rtc_event_log/events/rtc_event_probe_cluster_created.h

@@ -11,6 +11,7 @@
 #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_CLUSTER_CREATED_H_
 #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_CLUSTER_CREATED_H_
 
+#include <stdint.h>
 #include <memory>
 
 #include "logging/rtc_event_log/events/rtc_event.h"