Delete some unneeded references to module.h and module_api target.
Bug: None Change-Id: Iad983efa8666ca29b5e1ac22b3ade8987dbb2841 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242371 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35648}
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@ -993,7 +993,6 @@ rtc_library("neteq") {
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deps = [
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":audio_coding_module_typedefs",
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":webrtc_cng",
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"..:module_api",
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"..:module_api_public",
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"../../api:array_view",
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"../../api:rtp_headers",
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@ -1438,7 +1437,6 @@ if (rtc_include_tests) {
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deps = audio_coding_deps + [
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"../../api:scoped_refptr",
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"..:module_api",
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":audio_coding",
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"../../api/audio_codecs:audio_codecs_api",
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"../../api/audio_codecs:builtin_audio_decoder_factory",
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@ -25,7 +25,6 @@
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#include "api/units/data_size.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "modules/include/module.h"
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#include "modules/pacing/pacing_controller.h"
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#include "modules/pacing/rtp_packet_pacer.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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@ -437,7 +437,6 @@ rtc_library("mock_rtp_rtcp") {
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deps = [
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":rtp_rtcp",
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":rtp_rtcp_format",
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"..:module_api",
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"../../api/video:video_bitrate_allocation",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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@ -19,7 +19,6 @@
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#include "absl/types/optional.h"
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#include "api/video/video_bitrate_allocation.h"
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#include "modules/include/module.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
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