ff88950833
This is a reland of commit 38ddea5ee3320bf3441aeb3654e099b3695c9789 Original change's description: > Add InsertPacket method that takes RtpPacketInfo. > > The version which only passes receive_time will be removed (once migrated). > Keeping the version that only passes header and payload for convenience. > > This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class. > > Bug: webrtc:42223109 > Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43445} Bug: webrtc:42223109 Change-Id: I97d1d3d390e6d3de8bf9355b895ec336339d079f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369260 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43454}
272 lines
8.9 KiB
Plaintext
272 lines
8.9 KiB
Plaintext
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_library("audio") {
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sources = [
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"audio_level.cc",
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"audio_level.h",
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"audio_receive_stream.cc",
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.cc",
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"audio_state.h",
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"audio_transport_impl.cc",
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"audio_transport_impl.h",
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"channel_receive.cc",
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"channel_receive.h",
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"channel_receive_frame_transformer_delegate.cc",
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"channel_receive_frame_transformer_delegate.h",
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"channel_send.cc",
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"channel_send.h",
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"channel_send_frame_transformer_delegate.cc",
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"channel_send_frame_transformer_delegate.h",
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"conversion.h",
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"remix_resample.cc",
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"remix_resample.h",
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]
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deps = [
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"../api:array_view",
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"../api:call_api",
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"../api:field_trials_view",
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"../api:frame_transformer_interface",
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"../api:function_view",
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"../api:make_ref_counted",
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"../api:rtp_headers",
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"../api:rtp_packet_info",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api:sequence_checker",
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"../api:transport_api",
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"../api/audio:aec3_factory",
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"../api/audio:audio_device",
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"../api/audio:audio_frame_api",
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"../api/audio:audio_frame_processor",
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"../api/audio:audio_mixer_api",
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"../api/audio:audio_processing",
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"../api/audio_codecs:audio_codecs_api",
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"../api/crypto:frame_decryptor_interface",
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"../api/crypto:frame_encryptor_interface",
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"../api/crypto:options",
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"../api/environment",
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"../api/neteq:default_neteq_factory",
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"../api/neteq:neteq_api",
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"../api/rtc_event_log",
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"../api/task_queue",
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"../api/task_queue:pending_task_safety_flag",
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"../api/transport/rtp:rtp_source",
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"../api/units:time_delta",
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"../api/units:timestamp",
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"../call:audio_sender_interface",
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"../call:bitrate_allocator",
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"../call:call_interfaces",
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"../call:rtp_interfaces",
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"../common_audio",
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"../common_audio:common_audio_c",
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"../logging:rtc_event_audio",
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"../logging:rtc_stream_config",
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"../media:media_channel",
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"../media:media_channel_impl",
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"../modules/async_audio_processing",
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"../modules/audio_coding",
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"../modules/audio_coding:audio_coding_module_typedefs",
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"../modules/audio_coding:audio_encoder_cng",
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"../modules/audio_coding:audio_network_adaptor_config",
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"../modules/audio_coding:red",
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"../modules/audio_device",
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"../modules/audio_processing",
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"../modules/audio_processing:audio_frame_proxies",
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"../modules/audio_processing:rms_level",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:audio_format_to_string",
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"../rtc_base:buffer",
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"../rtc_base:checks",
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"../rtc_base:event_tracer",
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"../rtc_base:logging",
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"../rtc_base:macromagic",
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"../rtc_base:race_checker",
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"../rtc_base:rate_limiter",
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"../rtc_base:refcount",
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"../rtc_base:rtc_event",
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"../rtc_base:rtc_numerics",
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"../rtc_base:safe_conversions",
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"../rtc_base:safe_minmax",
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"../rtc_base:stringutils",
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"../rtc_base:threading",
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"../rtc_base:timeutils",
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"../rtc_base/containers:flat_set",
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"../rtc_base/experiments:field_trial_parser",
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"../rtc_base/synchronization:mutex",
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"../rtc_base/system:no_unique_address",
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"../rtc_base/task_utils:repeating_task",
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"../system_wrappers",
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"../system_wrappers:metrics",
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"utility:audio_frame_operations",
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"//third_party/abseil-cpp/absl/functional:any_invocable",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/strings:string_view",
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]
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}
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if (rtc_include_tests) {
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rtc_library("audio_end_to_end_test") {
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testonly = true
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sources = [
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"test/audio_end_to_end_test.cc",
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"test/audio_end_to_end_test.h",
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]
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deps = [
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":audio",
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"../api:simulated_network_api",
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"../api/audio:audio_device",
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"../api/task_queue",
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"../call:fake_network",
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"../modules/audio_device:test_audio_device_module",
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"../system_wrappers",
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"../test:test_common",
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"../test:test_support",
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"../test:video_test_constants",
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]
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}
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rtc_library("audio_tests") {
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testonly = true
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sources = [
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"audio_receive_stream_unittest.cc",
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"audio_send_stream_tests.cc",
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"audio_send_stream_unittest.cc",
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"audio_state_unittest.cc",
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"channel_receive_frame_transformer_delegate_unittest.cc",
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"channel_send_frame_transformer_delegate_unittest.cc",
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"channel_send_unittest.cc",
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"mock_voe_channel_proxy.h",
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"remix_resample_unittest.cc",
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"test/audio_stats_test.cc",
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"test/nack_test.cc",
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"test/non_sender_rtt_test.cc",
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]
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deps = [
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":audio",
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":audio_end_to_end_test",
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":channel_receive_unittest",
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"../api:array_view",
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"../api:bitrate_allocation",
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"../api:frame_transformer_factory",
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"../api:frame_transformer_interface",
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"../api:function_view",
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"../api:libjingle_peerconnection_api",
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"../api:make_ref_counted",
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"../api:mock_audio_mixer",
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"../api:mock_frame_decryptor",
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"../api:mock_frame_encryptor",
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"../api:mock_frame_transformer",
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"../api:mock_transformable_audio_frame",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api:transport_api",
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"../api/audio:audio_frame_api",
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"../api/audio:audio_processing_statistics",
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"../api/audio_codecs:audio_codecs_api",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../api/audio_codecs/opus:audio_decoder_opus",
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"../api/audio_codecs/opus:audio_encoder_opus",
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"../api/crypto:frame_decryptor_interface",
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"../api/crypto:frame_encryptor_interface",
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"../api/crypto:options",
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"../api/environment",
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"../api/environment:environment_factory",
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"../api/task_queue:default_task_queue_factory",
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"../api/task_queue/test:mock_task_queue_base",
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"../api/transport:bitrate_settings",
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"../api/transport:network_control",
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"../api/units:data_rate",
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"../api/units:data_size",
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"../api/units:time_delta",
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"../api/units:timestamp",
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"../call:bitrate_allocator",
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"../call:call_interfaces",
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"../call:mock_bitrate_allocator",
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"../call:mock_call_interfaces",
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"../call:mock_rtp_interfaces",
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"../call:rtp_interfaces",
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"../call:rtp_receiver",
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"../call:rtp_sender",
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"../common_audio",
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"../modules/audio_coding:audio_coding_module_typedefs",
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"../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_mixer:audio_mixer_test_utils",
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"../modules/audio_processing:mocks",
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"../modules/pacing",
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"../modules/rtp_rtcp:mock_rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:gunit_helpers",
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"../rtc_base:macromagic",
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"../rtc_base:refcount",
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"../rtc_base:rtc_base_tests_utils",
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"../rtc_base:safe_compare",
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"../rtc_base:task_queue_for_test",
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"../rtc_base:threading",
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"../rtc_base:timeutils",
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"../system_wrappers",
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"../test:audio_codec_mocks",
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"../test:field_trial",
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"../test:mock_transport",
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"../test:rtp_test_utils",
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"../test:run_loop",
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"../test:scoped_key_value_config",
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"../test:test_common",
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"../test:test_support",
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"../test:video_test_constants",
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"../test/time_controller:time_controller",
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"utility:utility_tests",
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"//testing/gtest",
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"//third_party/abseil-cpp/absl/memory",
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]
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}
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rtc_library("channel_receive_unittest") {
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testonly = true
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sources = [ "channel_receive_unittest.cc" ]
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deps = [
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":audio",
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"../api:mock_frame_transformer",
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"../api/audio:audio_device",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/crypto:frame_decryptor_interface",
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"../api/environment:environment_factory",
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"../api/task_queue:default_task_queue_factory",
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"../logging:mocks",
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"../modules/audio_device:mock_audio_device",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:ntp_time_util",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:logging",
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"../rtc_base:threading",
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"../test:audio_codec_mocks",
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"../test:mock_transport",
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"../test:test_support",
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"../test/time_controller",
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"//third_party/abseil-cpp/absl/strings",
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]
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}
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}
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