Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 35e10e6..d569d27 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -128,24 +128,34 @@
"mock_voe_channel_proxy.h",
"remix_resample_unittest.cc",
"test/audio_stats_test.cc",
+ "test/media_transport_test.cc",
"time_interval_unittest.cc",
"transport_feedback_packet_loss_tracker_unittest.cc",
]
deps = [
":audio",
":audio_end_to_end_test",
+ "../api:loopback_media_transport",
"../api:mock_audio_mixer",
"../api:mock_frame_decryptor",
"../api:mock_frame_encryptor",
"../api/audio:audio_frame_api",
+ "../api/audio_codecs:audio_codecs_api",
+ "../api/audio_codecs/opus:audio_decoder_opus",
+ "../api/audio_codecs/opus:audio_encoder_opus",
"../api/units:time_delta",
+ "../call:mock_bitrate_allocator",
"../call:mock_call_interfaces",
"../call:mock_rtp_interfaces",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../common_audio",
"../logging:mocks",
+ "../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
+
+ # For TestAudioDeviceModule
+ "../modules/audio_device:audio_device_impl",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing_statistics",
"../modules/audio_processing:mocks",
@@ -153,6 +163,7 @@
"../modules/pacing:pacing",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
+ "../modules/utility",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",