mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-12 21:30:45 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions

Remove all split channels related code

Browse source 
Bug: webrtc:13931
Change-Id: I93b8ca0ba1ec15bf260236bbc914b41fbb30aa58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310680
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40376}
This commit is contained in:
Florent Castelli 2023-06-27 20:07:09 +00:00 committed by WebRTC LUCI CQ
parent 9c62b32d28
commit d797cb6ca7
32 changed files with 187 additions and 2724 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
BUILD.gn
View file

@ -64,7 +64,6 @@ if (!build_with_chromium) {
"modules/video_capture:video_capture_internal_impl",
"modules/video_coding:video_codec_perf_tests",
"net/dcsctp:dcsctp_unittests",
"pc:peer_connection_mediachannel_split_unittests",
"pc:peerconnection_unittests",
"pc:rtc_pc_unittests",
"pc:slow_peer_connection_unittests",

409
infra/specs/client.webrtc.json
View file

@ -173,27 +173,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -665,27 +644,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -1186,27 +1144,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -1678,27 +1615,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -2435,25 +2351,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -2844,25 +2741,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-20.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -3234,25 +3112,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -3624,25 +3483,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -4033,25 +3873,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -4442,25 +4263,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -4833,25 +4635,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -5225,25 +5008,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -5635,25 +5399,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -6072,26 +5817,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cores": "12",
"cpu": "x86-64",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -6483,26 +6208,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cores": "12",
"cpu": "x86-64",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -6887,25 +6592,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -7297,25 +6983,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "arm64-64-Apple_M1",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -7714,25 +7381,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -8104,25 +7752,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -8494,25 +8123,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -8884,25 +8494,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {

4
infra/specs/gn_isolate_map.pyl
View file

@ -71,10 +71,6 @@
"label": "//modules:modules_unittests",
"type": "windowed_test_launcher",
},
"peer_connection_mediachannel_split_unittests": {
"label": "//pc:peer_connection_mediachannel_split_unittests",
"type": "console_test_launcher",
},
"peerconnection_unittests": {
"label": "//pc:peerconnection_unittests",
"type": "console_test_launcher",

2
infra/specs/test_suites.pyl
View file

@ -25,7 +25,6 @@
'modules_unittests': {
'mixins': ['shards-6'],
},
'peer_connection_mediachannel_split_unittests': {},
'peerconnection_unittests': {
'mixins': ['shards-4'],
},
@ -64,7 +63,6 @@
'modules_unittests': {
'mixins': ['shards-6'],
},
'peer_connection_mediachannel_split_unittests': {},
'peerconnection_unittests': {
'mixins': ['shards-4'],
},

486
infra/specs/tryserver.webrtc.json
View file

@ -173,27 +173,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -690,27 +669,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -1207,27 +1165,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -1750,27 +1687,6 @@
"test": "modules_unittests",
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
},
"resultdb": {
"enable": true,
"has_native_resultdb_integration": true
},
"swarming": {
"dimension_sets": [
{
"android_devices": "1",
"device_type": "walleye",
"os": "Android"
}
],
"service_account": "chromium-tester@chops-service-accounts.iam.gserviceaccount.com"
},
"test": "peer_connection_mediachannel_split_unittests",
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"merge": {
"script": "//testing/merge_scripts/standard_gtest_merge.py"
@ -5699,25 +5615,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -6120,26 +6017,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"isolate_profile_data": true,
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"isolate_profile_data": true,
@ -6617,25 +6494,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -7027,25 +6885,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -7460,25 +7299,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-20.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -7850,25 +7670,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -8329,25 +8130,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -8719,25 +8501,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -9128,25 +8891,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -9537,25 +9281,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -9927,25 +9652,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Ubuntu-18.04"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -10323,26 +10029,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cores": "12",
"cpu": "x86-64",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -10735,26 +10421,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cores": "12",
"cpu": "x86-64",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -11139,25 +10805,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "arm64-64-Apple_M1",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -11529,25 +11176,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -11989,25 +11617,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "arm64-64-Apple_M1",
"os": "Mac-12"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -12379,25 +11988,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -12773,25 +12363,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -13163,25 +12734,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -13553,25 +13105,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {
@ -13943,25 +13476,6 @@
},
"test_id_prefix": "ninja://modules:modules_unittests/"
},
{
"isolate_name": "peer_connection_mediachannel_split_unittests",
"merge": {
"script": "//testing/merge_scripts/standard_isolated_script_merge.py"
},
"name": "peer_connection_mediachannel_split_unittests",
"resultdb": {
"result_format": "json"
},
"swarming": {
"dimension_sets": [
{
"cpu": "x86-64",
"os": "Windows-10-19045"
}
]
},
"test_id_prefix": "ninja://pc:peer_connection_mediachannel_split_unittests/"
},
{
"isolate_name": "peerconnection_unittests",
"merge": {

42
media/BUILD.gn
View file

@ -141,40 +141,6 @@ rtc_library("rtc_media_base") {
]
}
rtc_library("media_channel_shim") {
sources = [
"base/media_channel_shim.cc",
"base/media_channel_shim.h",
]
deps = [
":codec",
":media_channel",
":media_channel_impl",
":stream_params",
"../api:frame_transformer_interface",
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/transport/rtp:rtp_source",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../api/video_codecs:video_codecs_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:network_route",
"../rtc_base/network:sent_packet",
]
absl_deps = [
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("media_channel_impl") {
sources = [
"base/media_channel_impl.cc",
@ -230,7 +196,6 @@ rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
deps = [
":codec",
":delayable",
":media_constants",
":rtp_utils",
":stream_params",
@ -279,11 +244,6 @@ rtc_source_set("media_channel") {
]
}
rtc_source_set("delayable") {
sources = [ "base/delayable.h" ]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("codec") {
sources = [
"base/codec.cc",
@ -474,7 +434,6 @@ rtc_library("rtc_audio_video") {
":codec",
":media_channel",
":media_channel_impl",
":media_channel_shim",
":media_constants",
":rid_description",
":rtc_media_base",
@ -841,7 +800,6 @@ if (rtc_include_tests) {
deps = [
":codec",
":media_channel",
":media_channel_shim",
":media_constants",
":rtc_audio_video",
":rtc_internal_video_codecs",

38
media/base/delayable.h
View file

@ -1,38 +0,0 @@
/*
* Copyright (c) 2019 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 MEDIA_BASE_DELAYABLE_H_
#define MEDIA_BASE_DELAYABLE_H_
#include <stdint.h>
#include "absl/types/optional.h"
namespace cricket {
// Delayable is used by user code through ApplyConstraints algorithm. Its
// methods must take precendence over similar functional in `syncable.h`.
class Delayable {
public:
virtual ~Delayable() {}
// Set base minimum delay of the receive stream with specified ssrc.
// Base minimum delay sets lower bound on minimum delay value which
// determines minimum delay until audio playout.
// Returns false if there is no stream with given ssrc.
virtual bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
virtual absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const = 0;
};
} // namespace cricket
#endif // MEDIA_BASE_DELAYABLE_H_

47
media/base/fake_media_engine.h
View file

@ -778,29 +778,6 @@ class FakeVoiceEngine : public VoiceEngineInterface {
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) override;
VoiceMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) override {
RTC_DCHECK_RUN_ON(call->worker_thread());
std::unique_ptr<VoiceMediaSendChannelInterface> send_channel;
std::unique_ptr<VoiceMediaReceiveChannelInterface> receive_channel;
if (role == MediaChannel::Role::kSend ||
role == MediaChannel::Role::kBoth) {
send_channel = CreateSendChannel(call, config, options, crypto_options,
codec_pair_id);
}
if (role == MediaChannel::Role::kReceive ||
role == MediaChannel::Role::kBoth) {
receive_channel = CreateReceiveChannel(call, config, options,
crypto_options, codec_pair_id);
}
return new VoiceMediaShimChannel(std::move(send_channel),
std::move(receive_channel));
}
// TODO(ossu): For proper testing, These should either individually settable
// or the voice engine should reference mockable factories.
@ -844,30 +821,6 @@ class FakeVideoEngine : public VideoEngineInterface {
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options) override;
VideoMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
override {
RTC_LOG(LS_INFO) << "CreateMediaChannel. Options: " << options.ToString();
std::unique_ptr<VideoMediaSendChannelInterface> send_channel;
std::unique_ptr<VideoMediaReceiveChannelInterface> receive_channel;
if (role == MediaChannel::Role::kSend ||
role == MediaChannel::Role::kBoth) {
send_channel = CreateSendChannel(call, config, options, crypto_options,
video_bitrate_allocator_factory);
}
if (role == MediaChannel::Role::kReceive ||
role == MediaChannel::Role::kBoth) {
receive_channel =
CreateReceiveChannel(call, config, options, crypto_options);
}
return new VideoMediaShimChannel(std::move(send_channel),
std::move(receive_channel));
}
FakeVideoMediaSendChannel* GetSendChannel(size_t index);
FakeVideoMediaReceiveChannel* GetReceiveChannel(size_t index);

14
media/base/media_channel.h
View file

@ -43,7 +43,6 @@
#include "call/video_receive_stream.h"
#include "common_video/include/quality_limitation_reason.h"
#include "media/base/codec.h"
#include "media/base/delayable.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
@ -75,7 +74,6 @@ webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
namespace cricket {
class AudioSource;
class MediaChannel; // TODO(bugs.webrtc.org/13931): Delete when irrelevant
class VideoCapturer;
struct RtpHeader;
struct VideoFormat;
@ -260,7 +258,7 @@ class MediaSendChannelInterface {
absl::AnyInvocable<void()> callback) = 0;
};
class MediaReceiveChannelInterface : public Delayable {
class MediaReceiveChannelInterface {
public:
virtual ~MediaReceiveChannelInterface() = default;
@ -312,6 +310,16 @@ class MediaReceiveChannelInterface : public Delayable {
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer) = 0;
// Set base minimum delay of the receive stream with specified ssrc.
// Base minimum delay sets lower bound on minimum delay value which
// determines minimum delay until audio playout.
// Returns false if there is no stream with given ssrc.
virtual bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
virtual absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const = 0;
};
// The stats information is structured as follows:

15
media/base/media_channel_impl.cc
View file

@ -61,11 +61,6 @@ MediaChannelUtil::MediaChannelUtil(TaskQueueBase* network_thread,
bool enable_dscp)
: transport_(network_thread, enable_dscp) {}
MediaChannel::MediaChannel(Role role,
TaskQueueBase* network_thread,
bool enable_dscp)
: MediaChannelUtil(network_thread, enable_dscp), role_(role) {}
MediaChannelUtil::~MediaChannelUtil() {}
void MediaChannelUtil::SetInterface(MediaChannelNetworkInterface* iface) {
@ -181,10 +176,6 @@ std::map<std::string, std::string> AudioSendParameters::ToStringMap() const {
return params;
}
cricket::MediaType VoiceMediaChannel::media_type() const {
return cricket::MediaType::MEDIA_TYPE_AUDIO;
}
VideoSendParameters::VideoSendParameters() = default;
VideoSendParameters::~VideoSendParameters() = default;
@ -194,12 +185,6 @@ std::map<std::string, std::string> VideoSendParameters::ToStringMap() const {
return params;
}
cricket::MediaType VideoMediaChannel::media_type() const {
return cricket::MediaType::MEDIA_TYPE_VIDEO;
}
void VideoMediaChannel::SetVideoCodecSwitchingEnabled(bool enabled) {}
// --------------------- MediaChannelUtil::TransportForMediaChannels -----
MediaChannelUtil::TransportForMediaChannels::TransportForMediaChannels(

212
media/base/media_channel_impl.h
View file

@ -59,16 +59,12 @@
#include "rtc_base/socket.h"
#include "rtc_base/thread_annotations.h"
// This file contains the base classes for classes that implement
// the MediaChannel interfaces.
// the channel interfaces.
// These implementation classes used to be the exposed interface names,
// but this is in the process of being changed.
// TODO(bugs.webrtc.org/13931): Remove the MediaChannel class.
namespace cricket {
class VoiceMediaChannel;
class VideoMediaChannel;
// The `MediaChannelUtil` class provides functionality that is used by
// multiple MediaChannel-like objects, of both sending and receiving
// types.
@ -195,212 +191,6 @@ class MediaChannelUtil {
TransportForMediaChannels transport_;
};
// The `MediaChannel` class implements both the SendChannel and
// ReceiveChannel interface. It is used in legacy code that does not
// use the split interfaces.
class MediaChannel : public MediaChannelUtil,
public MediaSendChannelInterface,
public MediaReceiveChannelInterface {
public:
// Role of the channel. Used to describe which interface it supports.
// This is temporary until we stop using the same implementation for both
// interfaces.
enum class Role {
kSend,
kReceive,
kBoth // Temporary value for non-converted test and downstream code
// TODO(bugs.webrtc.org/13931): Remove kBoth when usage is removed.
};
MediaChannel(Role role,
webrtc::TaskQueueBase* network_thread,
bool enable_dscp = false);
virtual ~MediaChannel() = default;
Role role() const { return role_; }
// Downcasting to the subclasses.
virtual VideoMediaChannel* AsVideoChannel() {
RTC_CHECK_NOTREACHED();
return nullptr;
}
virtual VoiceMediaChannel* AsVoiceChannel() {
RTC_CHECK_NOTREACHED();
return nullptr;
}
// Must declare the methods inherited from the base interface template,
// even when abstract, to tell the compiler that all instances of the name
// referred to by subclasses of this share the same implementation.
cricket::MediaType media_type() const override = 0;
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override = 0;
void OnPacketSent(const rtc::SentPacket& sent_packet) override = 0;
void OnReadyToSend(bool ready) override = 0;
void OnNetworkRouteChanged(absl::string_view transport_name,
const rtc::NetworkRoute& network_route) override =
0;
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override = 0;
// Methods from the APIs that are implemented in MediaChannelUtil
using MediaChannelUtil::ExtmapAllowMixed;
using MediaChannelUtil::HasNetworkInterface;
using MediaChannelUtil::SetExtmapAllowMixed;
using MediaChannelUtil::SetInterface;
private:
const Role role_;
};
// Base class for implementation classes
class VideoMediaChannel : public MediaChannel,
public VideoMediaSendChannelInterface,
public VideoMediaReceiveChannelInterface {
public:
explicit VideoMediaChannel(MediaChannel::Role role,
webrtc::TaskQueueBase* network_thread,
bool enable_dscp = false)
: MediaChannel(role, network_thread, enable_dscp) {}
~VideoMediaChannel() override {}
// Downcasting to the implemented interfaces.
VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; }
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
return this;
}
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
cricket::MediaType media_type() const override;
// Downcasting to the subclasses.
VideoMediaChannel* AsVideoChannel() override { return this; }
void SetExtmapAllowMixed(bool mixed) override {
MediaChannel::SetExtmapAllowMixed(mixed);
}
bool ExtmapAllowMixed() const override {
return MediaChannel::ExtmapAllowMixed();
}
void SetInterface(MediaChannelNetworkInterface* iface) override {
return MediaChannel::SetInterface(iface);
}
// Declared here in order to avoid "found by multiple paths" compile error
bool AddSendStream(const StreamParams& sp) override = 0;
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override = 0;
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override =
0;
bool AddRecvStream(const StreamParams& sp) override = 0;
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override = 0;
void SetEncoderSelector(uint32_t ssrc,
webrtc::VideoEncoderFactory::EncoderSelectorInterface*
encoder_selector) override {}
// Gets quality stats for the channel.
virtual bool GetSendStats(VideoMediaSendInfo* info) = 0;
virtual bool GetReceiveStats(VideoMediaReceiveInfo* info) = 0;
bool GetStats(VideoMediaSendInfo* info) override {
return GetSendStats(info);
}
bool GetStats(VideoMediaReceiveInfo* info) override {
return GetReceiveStats(info);
}
// TODO(bugs.webrtc.org/13931): Remove when configuration is more sensible
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override = 0;
// Enable network condition based codec switching.
// Note: should have been pure virtual.
void SetVideoCodecSwitchingEnabled(bool enabled) override;
private:
// Functions not implemented on this interface
bool HasNetworkInterface() const override {
return MediaChannel::HasNetworkInterface();
}
};
// Base class for implementation classes
class VoiceMediaChannel : public MediaChannel,
public VoiceMediaSendChannelInterface,
public VoiceMediaReceiveChannelInterface {
public:
MediaType media_type() const override;
VoiceMediaChannel(MediaChannel::Role role,
webrtc::TaskQueueBase* network_thread,
bool enable_dscp = false)
: MediaChannel(role, network_thread, enable_dscp) {}
~VoiceMediaChannel() override {}
// Downcasting to the implemented interfaces.
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
return this;
}
VoiceMediaChannel* AsVoiceChannel() override { return this; }
VideoMediaSendChannelInterface* AsVideoSendChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
// Declared here in order to avoid "found by multiple paths" compile error
bool AddSendStream(const StreamParams& sp) override = 0;
bool AddRecvStream(const StreamParams& sp) override = 0;
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override = 0;
void SetEncoderSelector(uint32_t ssrc,
webrtc::VideoEncoderFactory::EncoderSelectorInterface*
encoder_selector) override {}
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override = 0;
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override =
0;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override = 0;
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback = nullptr) override = 0;
void SetExtmapAllowMixed(bool mixed) override {
MediaChannel::SetExtmapAllowMixed(mixed);
}
bool ExtmapAllowMixed() const override {
return MediaChannel::ExtmapAllowMixed();
}
void SetInterface(MediaChannelNetworkInterface* iface) override {
return MediaChannel::SetInterface(iface);
}
bool HasNetworkInterface() const override {
return MediaChannel::HasNetworkInterface();
}
// Gets quality stats for the channel.
virtual bool GetSendStats(VoiceMediaSendInfo* info) = 0;
virtual bool GetReceiveStats(VoiceMediaReceiveInfo* info,
bool get_and_clear_legacy_stats) = 0;
bool GetStats(VoiceMediaSendInfo* info) override {
return GetSendStats(info);
}
bool GetStats(VoiceMediaReceiveInfo* info,
bool get_and_clear_legacy_stats) override {
return GetReceiveStats(info, get_and_clear_legacy_stats);
}
};
} // namespace cricket
#endif // MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_

57
media/base/media_channel_shim.cc
View file

@ -1,57 +0,0 @@
/*
* Copyright 2023 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 "media/base/media_channel_shim.h"
namespace cricket {
// Note: The VideoMediaChannel default implementations are not used here, and
// should be removed from that interface.
// TODO(bugs.webrtc.org/13931): Remove them.
VoiceMediaShimChannel::VoiceMediaShimChannel(
std::unique_ptr<VoiceMediaSendChannelInterface> send_impl,
std::unique_ptr<VoiceMediaReceiveChannelInterface> receive_impl)
: VoiceMediaChannel(MediaChannel::Role::kBoth, nullptr, false),
send_impl_(std::move(send_impl)),
receive_impl_(std::move(receive_impl)) {
if (send_impl_ && receive_impl_) {
send_impl_->SetSsrcListChangedCallback(
[this](const std::set<uint32_t>& choices) {
receive_impl_->ChooseReceiverReportSsrc(choices);
});
send_impl_->SetSendCodecChangedCallback([this]() {
receive_impl_->SetReceiveNackEnabled(send_impl_->SendCodecHasNack());
receive_impl_->SetReceiveNonSenderRttEnabled(
send_impl_->SenderNonSenderRttEnabled());
});
}
}
VideoMediaShimChannel::VideoMediaShimChannel(
std::unique_ptr<VideoMediaSendChannelInterface> send_impl,
std::unique_ptr<VideoMediaReceiveChannelInterface> receive_impl)
: VideoMediaChannel(MediaChannel::Role::kBoth, nullptr, false),
send_impl_(std::move(send_impl)),
receive_impl_(std::move(receive_impl)) {
if (send_impl_ && receive_impl_) {
send_impl_->SetSendCodecChangedCallback([this]() {
// Adjust receive streams based on send codec.
receive_impl_->SetReceiverFeedbackParameters(
send_impl_->SendCodecHasLntf(), send_impl_->SendCodecHasNack(),
send_impl_->SendCodecRtcpMode(), send_impl_->SendCodecRtxTime());
});
send_impl_->SetSsrcListChangedCallback(
[this](const std::set<uint32_t>& choices) {
receive_impl_->ChooseReceiverReportSsrc(choices);
});
}
}
} // namespace cricket

575
media/base/media_channel_shim.h
View file

@ -1,575 +0,0 @@
/*
* Copyright 2023 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 MEDIA_BASE_MEDIA_CHANNEL_SHIM_H_
#define MEDIA_BASE_MEDIA_CHANNEL_SHIM_H_
#include <stdint.h>
#include <functional>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/recordable_encoded_frame.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_channel_impl.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
namespace cricket {
// The VideoMediaShimChannel is replacing the VideoMediaChannel
// interface.
// If called with both send_impl and receive_impl, it operates in kBoth
// mode; if called with only one, it will shim that one and DCHECK if one
// tries to do functions belonging to the other.
class VoiceMediaShimChannel : public VoiceMediaChannel {
public:
VoiceMediaShimChannel(
std::unique_ptr<VoiceMediaSendChannelInterface> send_impl,
std::unique_ptr<VoiceMediaReceiveChannelInterface> receive_impl);
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
return this;
}
VideoMediaSendChannelInterface* AsVideoSendChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
// SetInterface needs to run on both send and receive channels.
void SetInterface(MediaChannelNetworkInterface* iface) override {
if (send_impl_) {
send_impl()->SetInterface(iface);
}
if (receive_impl_) {
receive_impl()->SetInterface(iface);
}
}
// Implementation of MediaBaseChannelInterface
cricket::MediaType media_type() const override { return MEDIA_TYPE_AUDIO; }
// Implementation of MediaSendChannelInterface
absl::optional<Codec> GetSendCodec() const override {
return send_impl()->GetSendCodec();
}
void OnPacketSent(const rtc::SentPacket& sent_packet) override {
send_impl()->OnPacketSent(sent_packet);
}
void OnReadyToSend(bool ready) override { send_impl()->OnReadyToSend(ready); }
void OnNetworkRouteChanged(absl::string_view transport_name,
const rtc::NetworkRoute& network_route) override {
send_impl()->OnNetworkRouteChanged(transport_name, network_route);
}
void SetExtmapAllowMixed(bool extmap_allow_mixed) override {
send_impl()->SetExtmapAllowMixed(extmap_allow_mixed);
}
bool HasNetworkInterface() const override {
return send_impl()->HasNetworkInterface();
}
bool ExtmapAllowMixed() const override {
return send_impl()->ExtmapAllowMixed();
}
bool AddSendStream(const StreamParams& sp) override {
return send_impl()->AddSendStream(sp);
}
bool RemoveSendStream(uint32_t ssrc) override {
return send_impl()->RemoveSendStream(ssrc);
}
void SetFrameEncryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
frame_encryptor) override {
send_impl()->SetFrameEncryptor(ssrc, frame_encryptor);
}
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback = nullptr) override {
return send_impl()->SetRtpSendParameters(ssrc, parameters,
std::move(callback));
}
void SetEncoderToPacketizerFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override {
return send_impl()->SetEncoderToPacketizerFrameTransformer(
ssrc, frame_transformer);
}
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override {
return send_impl()->GetRtpSendParameters(ssrc);
}
// Implementation of MediaReceiveChannelInterface
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override {
receive_impl()->OnPacketReceived(packet);
}
bool AddRecvStream(const StreamParams& sp) override {
return receive_impl()->AddRecvStream(sp);
}
bool RemoveRecvStream(uint32_t ssrc) override {
return receive_impl()->RemoveRecvStream(ssrc);
}
void ResetUnsignaledRecvStream() override {
return receive_impl()->ResetUnsignaledRecvStream();
}
absl::optional<uint32_t> GetUnsignaledSsrc() const override {
return receive_impl()->GetUnsignaledSsrc();
}
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {
return receive_impl()->ChooseReceiverReportSsrc(choices);
}
void OnDemuxerCriteriaUpdatePending() override {
receive_impl()->OnDemuxerCriteriaUpdatePending();
}
void OnDemuxerCriteriaUpdateComplete() override {
receive_impl()->OnDemuxerCriteriaUpdateComplete();
}
void SetFrameDecryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override {
receive_impl()->SetFrameDecryptor(ssrc, frame_decryptor);
}
void SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override {
receive_impl()->SetDepacketizerToDecoderFrameTransformer(ssrc,
frame_transformer);
}
bool SendCodecHasNack() const override {
return send_impl()->SendCodecHasNack();
}
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {
send_impl()->SetSendCodecChangedCallback(std::move(callback));
}
// Implementation of VoiceMediaSendChannel
bool SetSendParameters(const AudioSendParameters& params) override {
return send_impl()->SetSendParameters(params);
}
void SetSend(bool send) override { return send_impl()->SetSend(send); }
bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) override {
return send_impl()->SetAudioSend(ssrc, enable, options, source);
}
bool CanInsertDtmf() override { return send_impl()->CanInsertDtmf(); }
bool InsertDtmf(uint32_t ssrc, int event, int duration) override {
return send_impl()->InsertDtmf(ssrc, event, duration);
}
bool GetStats(VoiceMediaSendInfo* info) override {
return send_impl()->GetStats(info);
}
bool SenderNackEnabled() const override {
return send_impl()->SenderNackEnabled();
}
bool SenderNonSenderRttEnabled() const override {
return send_impl()->SenderNonSenderRttEnabled();
}
// Implementation of VoiceMediaReceiveChannelInterface
bool SetRecvParameters(const AudioRecvParameters& params) override {
return receive_impl()->SetRecvParameters(params);
}
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override {
return receive_impl()->GetRtpReceiveParameters(ssrc);
}
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override {
return receive_impl()->GetSources(ssrc);
}
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override {
return receive_impl()->GetDefaultRtpReceiveParameters();
}
void SetPlayout(bool playout) override {
return receive_impl()->SetPlayout(playout);
}
bool SetOutputVolume(uint32_t ssrc, double volume) override {
return receive_impl()->SetOutputVolume(ssrc, volume);
}
bool SetDefaultOutputVolume(double volume) override {
return receive_impl()->SetDefaultOutputVolume(volume);
}
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) override {
return receive_impl()->SetRawAudioSink(ssrc, std::move(sink));
}
void SetDefaultRawAudioSink(
std::unique_ptr<webrtc::AudioSinkInterface> sink) override {
return receive_impl()->SetDefaultRawAudioSink(std::move(sink));
}
bool GetStats(VoiceMediaReceiveInfo* info, bool reset_legacy) override {
return receive_impl_->GetStats(info, reset_legacy);
}
void SetReceiveNackEnabled(bool enabled) override {
receive_impl_->SetReceiveNackEnabled(enabled);
}
void SetReceiveNonSenderRttEnabled(bool enabled) override {
receive_impl_->SetReceiveNonSenderRttEnabled(enabled);
}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
send_impl()->SetSsrcListChangedCallback(std::move(callback));
}
// Implementation of Delayable
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override {
return receive_impl()->SetBaseMinimumPlayoutDelayMs(ssrc, delay_ms);
}
absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const override {
return receive_impl()->GetBaseMinimumPlayoutDelayMs(ssrc);
}
bool GetSendStats(VoiceMediaSendInfo* info) override {
return send_impl()->GetStats(info);
}
bool GetReceiveStats(VoiceMediaReceiveInfo* info,
bool reset_legacy) override {
return receive_impl()->GetStats(info, reset_legacy);
}
// Only for testing of implementations - these will be used to static_cast the
// pointers to the implementations, so can only be safely used in conjunction
// with the corresponding create functions.
VoiceMediaSendChannelInterface* SendImplForTesting() {
return send_impl_.get();
}
VoiceMediaReceiveChannelInterface* ReceiveImplForTesting() {
return receive_impl_.get();
}
private:
VoiceMediaSendChannelInterface* send_impl() { return send_impl_.get(); }
VoiceMediaReceiveChannelInterface* receive_impl() {
RTC_DCHECK(receive_impl_);
return receive_impl_.get();
}
const VoiceMediaSendChannelInterface* send_impl() const {
RTC_DCHECK(send_impl_);
return send_impl_.get();
}
const VoiceMediaReceiveChannelInterface* receive_impl() const {
return receive_impl_.get();
}
std::unique_ptr<VoiceMediaSendChannelInterface> send_impl_;
std::unique_ptr<VoiceMediaReceiveChannelInterface> receive_impl_;
};
// The VideoMediaShimChannel is replacing the VideoMediaChannel
// interface.
// If called with both send_impl and receive_impl, it operates in kBoth
// mode; if called with only one, it will shim that one and DCHECK if one
// tries to do functions belonging to the other.
class VideoMediaShimChannel : public VideoMediaChannel {
public:
VideoMediaShimChannel(
std::unique_ptr<VideoMediaSendChannelInterface> send_impl,
std::unique_ptr<VideoMediaReceiveChannelInterface> receive_impl);
VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; }
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
return this;
}
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
// SetInterface needs to run on both send and receive channels.
void SetInterface(MediaChannelNetworkInterface* iface) override {
if (send_impl_) {
send_impl()->SetInterface(iface);
}
if (receive_impl_) {
receive_impl()->SetInterface(iface);
}
}
// Implementation of MediaBaseChannelInterface
cricket::MediaType media_type() const override { return MEDIA_TYPE_VIDEO; }
// Implementation of MediaSendChannelInterface
void OnPacketSent(const rtc::SentPacket& sent_packet) override {
send_impl()->OnPacketSent(sent_packet);
}
void OnReadyToSend(bool ready) override { send_impl()->OnReadyToSend(ready); }
void OnNetworkRouteChanged(absl::string_view transport_name,
const rtc::NetworkRoute& network_route) override {
send_impl()->OnNetworkRouteChanged(transport_name, network_route);
}
void SetExtmapAllowMixed(bool extmap_allow_mixed) override {
send_impl()->SetExtmapAllowMixed(extmap_allow_mixed);
}
bool HasNetworkInterface() const override {
return send_impl()->HasNetworkInterface();
}
bool ExtmapAllowMixed() const override {
return send_impl()->ExtmapAllowMixed();
}
bool AddSendStream(const StreamParams& sp) override {
return send_impl()->AddSendStream(sp);
}
bool RemoveSendStream(uint32_t ssrc) override {
return send_impl()->RemoveSendStream(ssrc);
}
void SetFrameEncryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
frame_encryptor) override {
send_impl()->SetFrameEncryptor(ssrc, frame_encryptor);
}
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback = nullptr) override {
return send_impl()->SetRtpSendParameters(ssrc, parameters,
std::move(callback));
}
void SetEncoderToPacketizerFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override {
return send_impl()->SetEncoderToPacketizerFrameTransformer(
ssrc, frame_transformer);
}
void SetEncoderSelector(uint32_t ssrc,
webrtc::VideoEncoderFactory::EncoderSelectorInterface*
encoder_selector) override {
send_impl()->SetEncoderSelector(ssrc, encoder_selector);
}
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override {
return send_impl()->GetRtpSendParameters(ssrc);
}
// Send_Implementation of VideoMediaSendChannelInterface
bool SetSendParameters(const VideoSendParameters& params) override {
return send_impl()->SetSendParameters(params);
}
absl::optional<Codec> GetSendCodec() const override {
return send_impl()->GetSendCodec();
}
bool SetSend(bool send) override { return send_impl()->SetSend(send); }
bool SetVideoSend(
uint32_t ssrc,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override {
return send_impl()->SetVideoSend(ssrc, options, source);
}
void GenerateSendKeyFrame(uint32_t ssrc,
const std::vector<std::string>& rids) override {
return send_impl()->GenerateSendKeyFrame(ssrc, rids);
}
void SetVideoCodecSwitchingEnabled(bool enabled) override {
return send_impl()->SetVideoCodecSwitchingEnabled(enabled);
}
bool GetStats(VideoMediaSendInfo* info) override {
return send_impl_->GetStats(info);
}
bool GetSendStats(VideoMediaSendInfo* info) override {
return send_impl_->GetStats(info);
}
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override {
return send_impl_->FillBitrateInfo(bwe_info);
}
// Information queries to support SetReceiverFeedbackParameters
webrtc::RtcpMode SendCodecRtcpMode() const override {
return send_impl()->SendCodecRtcpMode();
}
bool SendCodecHasLntf() const override {
return send_impl()->SendCodecHasLntf();
}
bool SendCodecHasNack() const override {
return send_impl()->SendCodecHasNack();
}
absl::optional<int> SendCodecRtxTime() const override {
return send_impl()->SendCodecRtxTime();
}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
send_impl()->SetSsrcListChangedCallback(std::move(callback));
}
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {
// This callback is used internally by the shim, so should not be called by
// users for the "both" case.
if (send_impl_ && receive_impl_) {
RTC_CHECK_NOTREACHED();
}
send_impl()->SetSendCodecChangedCallback(std::move(callback));
}
// Implementation of Delayable
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override {
return receive_impl()->SetBaseMinimumPlayoutDelayMs(ssrc, delay_ms);
}
absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const override {
return receive_impl()->GetBaseMinimumPlayoutDelayMs(ssrc);
}
// Implementation of MediaReceiveChannelInterface
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override {
receive_impl()->OnPacketReceived(packet);
}
bool AddRecvStream(const StreamParams& sp) override {
return receive_impl()->AddRecvStream(sp);
}
bool RemoveRecvStream(uint32_t ssrc) override {
return receive_impl()->RemoveRecvStream(ssrc);
}
void ResetUnsignaledRecvStream() override {
return receive_impl()->ResetUnsignaledRecvStream();
}
absl::optional<uint32_t> GetUnsignaledSsrc() const override {
return receive_impl()->GetUnsignaledSsrc();
}
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {
return receive_impl()->ChooseReceiverReportSsrc(choices);
}
void OnDemuxerCriteriaUpdatePending() override {
receive_impl()->OnDemuxerCriteriaUpdatePending();
}
void OnDemuxerCriteriaUpdateComplete() override {
receive_impl()->OnDemuxerCriteriaUpdateComplete();
}
void SetFrameDecryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override {
receive_impl()->SetFrameDecryptor(ssrc, frame_decryptor);
}
void SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override {
receive_impl()->SetDepacketizerToDecoderFrameTransformer(ssrc,
frame_transformer);
}
// Implementation of VideoMediaReceiveChannelInterface
bool SetRecvParameters(const VideoRecvParameters& params) override {
return receive_impl()->SetRecvParameters(params);
}
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override {
return receive_impl()->GetRtpReceiveParameters(ssrc);
}
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override {
return receive_impl()->GetDefaultRtpReceiveParameters();
}
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {
return receive_impl()->SetSink(ssrc, sink);
}
void SetDefaultSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {
return receive_impl()->SetDefaultSink(sink);
}
void RequestRecvKeyFrame(uint32_t ssrc) override {
return receive_impl()->RequestRecvKeyFrame(ssrc);
}
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override {
return receive_impl()->GetSources(ssrc);
}
// Set recordable encoded frame callback for `ssrc`
void SetRecordableEncodedFrameCallback(
uint32_t ssrc,
std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
override {
return receive_impl()->SetRecordableEncodedFrameCallback(
ssrc, std::move(callback));
}
// Clear recordable encoded frame callback for `ssrc`
void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override {
receive_impl()->ClearRecordableEncodedFrameCallback(ssrc);
}
bool GetStats(VideoMediaReceiveInfo* info) override {
return receive_impl()->GetStats(info);
}
bool GetReceiveStats(VideoMediaReceiveInfo* info) override {
return receive_impl()->GetStats(info);
}
void SetReceiverFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
absl::optional<int> rtx_time) override {
receive_impl()->SetReceiverFeedbackParameters(lntf_enabled, nack_enabled,
rtcp_mode, rtx_time);
}
void SetReceive(bool receive) override {
receive_impl()->SetReceive(receive);
}
bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override {
return receive_impl()->AddDefaultRecvStreamForTesting(sp);
}
// Only for testing of implementations - these will be used to static_cast the
// pointers to the implementations, so can only be safely used in conjunction
// with the corresponding create functions.
VideoMediaSendChannelInterface* SendImplForTesting() {
return send_impl_.get();
}
VideoMediaReceiveChannelInterface* ReceiveImplForTesting() {
return receive_impl_.get();
}
private:
VideoMediaSendChannelInterface* send_impl() { return send_impl_.get(); }
VideoMediaReceiveChannelInterface* receive_impl() {
RTC_DCHECK(receive_impl_);
return receive_impl_.get();
}
const VideoMediaSendChannelInterface* send_impl() const {
RTC_DCHECK(send_impl_);
return send_impl_.get();
}
const VideoMediaReceiveChannelInterface* receive_impl() const {
return receive_impl_.get();
}
std::unique_ptr<VideoMediaSendChannelInterface> send_impl_;
std::unique_ptr<VideoMediaReceiveChannelInterface> receive_impl_;
};
} // namespace cricket
#endif // MEDIA_BASE_MEDIA_CHANNEL_SHIM_H_

90
media/base/media_engine.h
View file

@ -38,9 +38,6 @@ class Call;
namespace cricket {
class VideoMediaChannel;
class VoiceMediaChannel;
// Checks that the scalability_mode value of each encoding is supported by at
// least one video codec of the list. If the list is empty, no check is done.
webrtc::RTCError CheckScalabilityModeValues(
@ -120,42 +117,6 @@ class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
return nullptr;
}
// MediaChannel creation
// Creates a voice media channel. Returns NULL on failure.
virtual VoiceMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) {
// For the case where a subclass overrides the deprecated method
// but not the replacement method, call the deprecated method.
// TODO(bugs.webrtc.org/13931): Remove default implementation
// when downstream has migrated to new API.
RTC_CHECK(!recursion_guard_);
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wdeprecated-declarations"
RTC_LOG(LS_ERROR)
<< "Override of deprecated declaration detected - please update!";
return CreateMediaChannel(call, config, options, crypto_options);
#pragma clang diagnostic pop
}
// Backwards compatible version
[[deprecated("Use version with role parameter")]] virtual VoiceMediaChannel*
CreateMediaChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options) {
recursion_guard_ = true;
auto new_channel =
CreateMediaChannel(MediaChannel::Role::kBoth, call, config, options,
crypto_options, webrtc::AudioCodecPairId::Create());
recursion_guard_ = false;
return new_channel;
}
virtual const std::vector<AudioCodec>& send_codecs() const = 0;
virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
@ -170,11 +131,6 @@ class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
virtual absl::optional<webrtc::AudioDeviceModule::Stats>
GetAudioDeviceStats() = 0;
private:
// Workaround variable for avoiding recursion between old and new APIs.
// TODO(bugs.webrtc.org/13931): Remove when old interface is gone.
bool recursion_guard_ = false;
};
class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
@ -206,47 +162,6 @@ class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
return nullptr;
}
// Creates a video media channel.
// Returns NULL on failure.
virtual VideoMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
// For the case where a subclass overrides the deprecated method
// but not the replacement method, call the deprecated method.
// TODO(bugs.webrtc.org/13931): Remove default implementation
// when downstream has migrated to new API.
RTC_CHECK(!recursion_guard_);
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wdeprecated-declarations"
RTC_LOG(LS_ERROR)
<< "Override of deprecated declaration detected - please update!";
return CreateMediaChannel(call, config, options, crypto_options,
video_bitrate_allocator_factory);
#pragma clang diagnostic pop
}
// Creates a video media channel.
// Returns NULL on failure.
// TODO(bugs.webrtc.org/13931): Stop downstream usage of this function.
[[deprecated("Please specify the role")]] virtual VideoMediaChannel*
CreateMediaChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
recursion_guard_ = true;
auto new_channel =
CreateMediaChannel(MediaChannel::Role::kBoth, call, config, options,
crypto_options, video_bitrate_allocator_factory);
recursion_guard_ = false;
return new_channel;
}
// Retrieve list of supported codecs.
virtual std::vector<VideoCodec> send_codecs() const = 0;
virtual std::vector<VideoCodec> recv_codecs() const = 0;
@ -261,11 +176,6 @@ class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
RTC_DCHECK(include_rtx);
return recv_codecs();
}
private:
// Workaround variable for avoiding recursion between old and new APIs.
// TODO(bugs.webrtc.org/13931): Remove when old interface is gone.
bool recursion_guard_ = false;
};
// MediaEngineInterface is an abstraction of a media engine which can be

64
media/base/media_engine_unittest.cc
View file

@ -83,68 +83,4 @@ class MostlyMockVoiceEngineInterface : public VoiceEngineInterface {
(override));
};
class OldStyleVoiceEngineInterface : public MostlyMockVoiceEngineInterface {
public:
using MostlyMockVoiceEngineInterface::CreateMediaChannel;
// Old style overrides the deprecated API only.
VoiceMediaChannel* CreateMediaChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options) override {
++call_count;
return nullptr;
}
int call_count = 0;
};
class NewStyleVoiceEngineInterface : public MostlyMockVoiceEngineInterface {
// New style overrides the non-deprecated API.
VoiceMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) override {
return nullptr;
}
};
TEST(MediaEngineTest, NewStyleApiCallsOldIfOverridden) {
OldStyleVoiceEngineInterface implementation_under_test;
MediaConfig config;
AudioOptions options;
webrtc::CryptoOptions crypto_options;
// Calling the old-style interface.
implementation_under_test.CreateMediaChannel(nullptr, config, options,
crypto_options);
EXPECT_EQ(implementation_under_test.call_count, 1);
// Calling the new-style interface redirects to the old-style interface.
implementation_under_test.CreateMediaChannel(
MediaChannel::Role::kBoth, nullptr, config, options, crypto_options,
webrtc::AudioCodecPairId::Create());
EXPECT_EQ(implementation_under_test.call_count, 2);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(MediaEngineTest, NoOverrideOfCreateCausesCrash) {
MostlyMockVoiceEngineInterface implementation_under_test;
MediaConfig config;
AudioOptions options;
webrtc::CryptoOptions crypto_options;
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wdeprecated-declarations"
EXPECT_DEATH(implementation_under_test.CreateMediaChannel(
nullptr, config, options, crypto_options),
"Check failed: !recursion_guard_");
#pragma clang diagnostic pop
EXPECT_DEATH(implementation_under_test.CreateMediaChannel(
MediaChannel::Role::kBoth, nullptr, config, options,
crypto_options, webrtc::AudioCodecPairId::Create()),
"Check failed: !recursion_guard_");
}
#endif
} // namespace cricket

13
media/engine/null_webrtc_video_engine.h
View file

@ -24,8 +24,6 @@ class Call;
namespace cricket {
class VideoMediaChannel;
// Video engine implementation that does nothing and can be used in
// CompositeMediaEngine.
class NullWebRtcVideoEngine : public VideoEngineInterface {
@ -49,17 +47,6 @@ class NullWebRtcVideoEngine : public VideoEngineInterface {
const override {
return {};
}
VideoMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
override {
return nullptr;
}
};
} // namespace cricket

23
media/engine/webrtc_video_engine.cc
View file

@ -762,29 +762,6 @@ WebRtcVideoEngine::CreateReceiveChannel(
call, config, options, crypto_options, decoder_factory_.get());
}
VideoMediaChannel* WebRtcVideoEngine::CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
RTC_LOG(LS_INFO) << "CreateMediaChannel. Options: " << options.ToString();
std::unique_ptr<VideoMediaSendChannelInterface> send_channel;
std::unique_ptr<VideoMediaReceiveChannelInterface> receive_channel;
if (role == MediaChannel::Role::kSend || role == MediaChannel::Role::kBoth) {
send_channel = CreateSendChannel(call, config, options, crypto_options,
video_bitrate_allocator_factory);
}
if (role == MediaChannel::Role::kReceive ||
role == MediaChannel::Role::kBoth) {
receive_channel =
CreateReceiveChannel(call, config, options, crypto_options);
}
return new VideoMediaShimChannel(std::move(send_channel),
std::move(receive_channel));
}
std::vector<VideoCodec> WebRtcVideoEngine::send_codecs(bool include_rtx) const {
return GetPayloadTypesAndDefaultCodecs(encoder_factory_.get(),
/*is_decoder_factory=*/false,

10
media/engine/webrtc_video_engine.h
View file

@ -59,7 +59,6 @@
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_channel_impl.h"
#include "media/base/media_channel_shim.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
@ -117,15 +116,6 @@ class WebRtcVideoEngine : public VideoEngineInterface {
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options) override;
VideoMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
override;
std::vector<VideoCodec> send_codecs() const override {
return send_codecs(true);
}

23
media/engine/webrtc_voice_engine.cc
View file

@ -486,29 +486,6 @@ WebRtcVoiceEngine::CreateReceiveChannel(
this, config, options, crypto_options, call, codec_pair_id);
}
VoiceMediaChannel* WebRtcVoiceEngine::CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) {
RTC_DCHECK_RUN_ON(call->worker_thread());
std::unique_ptr<VoiceMediaSendChannelInterface> send_channel;
std::unique_ptr<VoiceMediaReceiveChannelInterface> receive_channel;
if (role == MediaChannel::Role::kSend || role == MediaChannel::Role::kBoth) {
send_channel =
CreateSendChannel(call, config, options, crypto_options, codec_pair_id);
}
if (role == MediaChannel::Role::kReceive ||
role == MediaChannel::Role::kBoth) {
receive_channel = CreateReceiveChannel(call, config, options,
crypto_options, codec_pair_id);
}
return new VoiceMediaShimChannel(std::move(send_channel),
std::move(receive_channel));
}
void WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: "

9
media/engine/webrtc_voice_engine.h
View file

@ -53,7 +53,6 @@
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_channel_impl.h"
#include "media/base/media_channel_shim.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/rtp_utils.h"
@ -121,14 +120,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) override;
VoiceMediaChannel* CreateMediaChannel(
MediaChannel::Role role,
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) override;
const std::vector<AudioCodec>& send_codecs() const override;
const std::vector<AudioCodec>& recv_codecs() const override;
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()

1
media/engine/webrtc_voice_engine_unittest.cc
View file

@ -28,7 +28,6 @@
#include "media/base/fake_network_interface.h"
#include "media/base/fake_rtp.h"
#include "media/base/media_channel.h"
#include "media/base/media_channel_shim.h"
#include "media/base/media_constants.h"
#include "media/engine/fake_webrtc_call.h"
#include "modules/audio_device/include/mock_audio_device.h"

24
pc/BUILD.gn
View file

@ -2450,7 +2450,6 @@ if (rtc_include_tests && !build_with_chromium) {
"../common_video",
"../logging:fake_rtc_event_log",
"../media:codec",
"../media:delayable",
"../media:media_channel",
"../media:media_constants",
"../media:rid_description",
@ -2559,27 +2558,6 @@ if (rtc_include_tests && !build_with_chromium) {
]
}
rtc_test("peer_connection_mediachannel_split_unittests") {
testonly = true
sources = [ "peer_connection_mediachannel_split_unittest.cc" ]
deps = [
":integration_test_helpers",
":session_description",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:scoped_refptr",
"../media:codec",
"../media:media_constants",
"../rtc_base:gunit_helpers",
"../rtc_base:rtc_base_tests_utils",
"../test:test_main",
"../test:test_support",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("data_channel_controller_unittest") {
testonly = true
sources = [ "data_channel_controller_unittest.cc" ]
@ -2758,7 +2736,7 @@ if (rtc_include_tests && !build_with_chromium) {
"test/mock_peer_connection_observers.h",
"test/mock_rtp_receiver_internal.h",
"test/mock_rtp_sender_internal.h",
"test/mock_voice_media_channel.h",
"test/mock_voice_media_receive_channel_interface.h",
"test/peer_connection_test_wrapper.cc",
"test/peer_connection_test_wrapper.h",
"test/rtc_stats_obtainer.h",

40
pc/audio_rtp_receiver_unittest.cc
View file

@ -12,7 +12,7 @@
#include <atomic>
#include "pc/test/mock_voice_media_channel.h"
#include "pc/test/mock_voice_media_receive_channel_interface.h"
#include "rtc_base/gunit.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
@ -38,28 +38,26 @@ class AudioRtpReceiverTest : public ::testing::Test {
rtc::make_ref_counted<AudioRtpReceiver>(worker_,
std::string(),
std::vector<std::string>(),
false)),
media_channel_(cricket::MediaChannel::Role::kReceive,
rtc::Thread::Current()) {
EXPECT_CALL(media_channel_, SetRawAudioSink(kSsrc, _));
EXPECT_CALL(media_channel_, SetBaseMinimumPlayoutDelayMs(kSsrc, _));
false)) {
EXPECT_CALL(receive_channel_, SetRawAudioSink(kSsrc, _));
EXPECT_CALL(receive_channel_, SetBaseMinimumPlayoutDelayMs(kSsrc, _));
}
~AudioRtpReceiverTest() {
EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolumeMuted));
EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolumeMuted));
receiver_->SetMediaChannel(nullptr);
}
rtc::AutoThread main_thread_;
rtc::Thread* worker_;
rtc::scoped_refptr<AudioRtpReceiver> receiver_;
cricket::MockVoiceMediaChannel media_channel_;
cricket::MockVoiceMediaReceiveChannelInterface receive_channel_;
};
TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) {
std::atomic_int set_volume_calls(0);
EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kDefaultVolume))
EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kDefaultVolume))
.WillOnce(InvokeWithoutArgs([&] {
set_volume_calls++;
return true;
@ -67,11 +65,11 @@ TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) {
receiver_->track();
receiver_->track()->set_enabled(true);
receiver_->SetMediaChannel(media_channel_.AsVoiceReceiveChannel());
EXPECT_CALL(media_channel_, SetDefaultRawAudioSink(_)).Times(0);
receiver_->SetMediaChannel(&receive_channel_);
EXPECT_CALL(receive_channel_, SetDefaultRawAudioSink(_)).Times(0);
receiver_->SetupMediaChannel(kSsrc);
EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolume))
EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolume))
.WillOnce(InvokeWithoutArgs([&] {
set_volume_calls++;
return true;
@ -87,11 +85,11 @@ TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) {
receiver_->OnSetVolume(kVolume);
receiver_->track()->set_enabled(true);
receiver_->SetMediaChannel(media_channel_.AsVoiceReceiveChannel());
receiver_->SetMediaChannel(&receive_channel_);
// The previosly set initial volume should be propagated to the provided
// media_channel_ as soon as SetupMediaChannel is called.
EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolume));
EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolume));
receiver_->SetupMediaChannel(kSsrc);
}
@ -102,13 +100,13 @@ TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) {
TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
webrtc::test::RunLoop loop;
auto* thread = rtc::Thread::Current(); // Points to loop's thread.
cricket::MockVoiceMediaChannel media_channel(
cricket::MediaChannel::Role::kReceive, thread);
cricket::MockVoiceMediaReceiveChannelInterface receive_channel;
auto receiver = rtc::make_ref_counted<AudioRtpReceiver>(
thread, std::string(), std::vector<std::string>(), true, &media_channel);
thread, std::string(), std::vector<std::string>(), true,
&receive_channel);
EXPECT_CALL(media_channel, SetDefaultRawAudioSink(_)).Times(1);
EXPECT_CALL(media_channel, SetDefaultOutputVolume(kDefaultVolume)).Times(1);
EXPECT_CALL(receive_channel, SetDefaultRawAudioSink(_)).Times(1);
EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kDefaultVolume)).Times(1);
receiver->SetupUnsignaledMediaChannel();
loop.Flush();
@ -119,10 +117,10 @@ TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
// for the worker thread. This notification should trigger the volume
// of the media channel to be set to kVolumeMuted.
// Flush the worker thread, but set the expectation first for the call.
EXPECT_CALL(media_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
loop.Flush();
EXPECT_CALL(media_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
receiver->SetMediaChannel(nullptr);
}

75
pc/channel.cc
View file

@ -140,32 +140,6 @@ BaseChannel::BaseChannel(
RTC_DLOG(LS_INFO) << "Created channel: " << ToString();
}
BaseChannel::BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: media_channel_(std::move(media_channel)),
worker_thread_(worker_thread),
network_thread_(network_thread),
signaling_thread_(signaling_thread),
alive_(PendingTaskSafetyFlag::Create()),
srtp_required_(srtp_required),
extensions_filter_(
crypto_options.srtp.enable_encrypted_rtp_header_extensions
? webrtc::RtpExtension::kPreferEncryptedExtension
: webrtc::RtpExtension::kDiscardEncryptedExtension),
demuxer_criteria_(mid),
ssrc_generator_(ssrc_generator) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(media_channel_);
RTC_DCHECK(ssrc_generator_);
RTC_DLOG(LS_INFO) << "Created channel: " << ToString();
}
BaseChannel::~BaseChannel() {
TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
RTC_DCHECK_RUN_ON(worker_thread_);
@ -178,15 +152,9 @@ BaseChannel::~BaseChannel() {
}
std::string BaseChannel::ToString() const {
if (media_send_channel_) {
return StringFormat(
"{mid: %s, media_type: %s}", mid().c_str(),
MediaTypeToString(media_send_channel_->media_type()).c_str());
} else {
return StringFormat(
"{mid: %s, media_type: %s}", mid().c_str(),
MediaTypeToString(media_channel_->media_type()).c_str());
}
return StringFormat(
"{mid: %s, media_type: %s}", mid().c_str(),
MediaTypeToString(media_send_channel_->media_type()).c_str());
}
bool BaseChannel::ConnectToRtpTransport_n() {
@ -866,26 +834,6 @@ VoiceChannel::VoiceChannel(
crypto_options,
ssrc_generator) {}
VoiceChannel::VoiceChannel(
rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel_impl),
mid,
srtp_required,
crypto_options,
ssrc_generator) {
InitCallback();
}
VoiceChannel::~VoiceChannel() {
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
// this can't be done in the base class, since it calls a virtual
@ -1041,23 +989,6 @@ VideoChannel::VideoChannel(
send_channel()->SendCodecRtxTime());
});
}
VideoChannel::VideoChannel(
rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel_impl),
mid,
srtp_required,
crypto_options,
ssrc_generator) {}
VideoChannel::~VideoChannel() {
TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel");

57
pc/channel.h
View file

@ -94,16 +94,6 @@ class BaseChannel : public ChannelInterface,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Constructor for use when the MediaChannel is not split
// TODO(bugs.webrtc.org/13931): Delete when split channel project is complete.
BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
virtual ~BaseChannel();
rtc::Thread* worker_thread() const { return worker_thread_; }
@ -316,13 +306,6 @@ class BaseChannel : public ChannelInterface,
// Return description of media channel to facilitate logging
std::string ToString() const;
// MediaChannel implementation pointers.
// Either the `media_channel_` is set, or the `media_send_channel_`
// and the `media_receive_channel_` is set.
// TODO(bugs.webrtc.org/13931): Delete `media_channel_` when split channel
// project is complete.
const std::unique_ptr<MediaChannel> media_channel_;
const std::unique_ptr<MediaSendChannelInterface> media_send_channel_;
const std::unique_ptr<MediaReceiveChannelInterface> media_receive_channel_;
@ -396,16 +379,6 @@ class VoiceChannel : public BaseChannel {
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Constructor for use when the MediaChannel is not split
// TODO(bugs.webrtc.org/13931): Delete when split channel project is complete.
VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
~VoiceChannel();
@ -416,17 +389,11 @@ class VoiceChannel : public BaseChannel {
VoiceChannel* AsVoiceChannel() override { return this; }
VoiceMediaSendChannelInterface* send_channel() {
if (media_send_channel_) {
return media_send_channel_->AsVoiceSendChannel();
}
return media_channel_->AsVoiceSendChannel();
return media_send_channel_->AsVoiceSendChannel();
}
VoiceMediaReceiveChannelInterface* receive_channel() {
if (media_receive_channel_) {
return media_receive_channel_->AsVoiceReceiveChannel();
}
return media_channel_->AsVoiceReceiveChannel();
return media_receive_channel_->AsVoiceReceiveChannel();
}
VoiceMediaSendChannelInterface* media_send_channel() override {
@ -483,16 +450,6 @@ class VideoChannel : public BaseChannel {
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Constructor for use when the MediaChannel is not split
// TODO(bugs.webrtc.org/13931): Delete when split channel project is complete.
VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
~VideoChannel();
VideoChannel* AsVideoChannel() override { return this; }
@ -502,17 +459,11 @@ class VideoChannel : public BaseChannel {
}
VideoMediaSendChannelInterface* send_channel() {
if (media_send_channel_) {
return media_send_channel_->AsVideoSendChannel();
}
return media_channel_->AsVideoSendChannel();
return media_send_channel_->AsVideoSendChannel();
}
VideoMediaReceiveChannelInterface* receive_channel() {
if (media_receive_channel_) {
return media_receive_channel_->AsVideoReceiveChannel();
}
return media_channel_->AsVideoReceiveChannel();
return media_receive_channel_->AsVideoReceiveChannel();
}
VideoMediaSendChannelInterface* media_send_channel() override {

1
pc/channel_interface.h
View file

@ -28,7 +28,6 @@ class VideoBitrateAllocatorFactory;
namespace cricket {
class MediaChannel;
class VoiceChannel;
class VideoChannel;
class MediaContentDescription;

209
pc/peer_connection_mediachannel_split_unittest.cc
View file

@ -1,209 +0,0 @@
/*
* Copyright 2012 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.
*/
// Integration tests for PeerConnection to exercise the options of
// either splitting or not splitting the MediaChannel object.
// These tests exercise a full stack over a simulated network.
// TODO(bugs.webrtc.org/13931): Delete these tests when split is landed.
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/media_stream_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/scoped_refptr.h"
#include "api/stats/rtc_stats.h"
#include "api/stats/rtc_stats_report.h"
#include "api/stats/rtcstats_objects.h"
#include "media/base/codec.h"
#include "media/base/media_constants.h"
#include "pc/session_description.h"
#include "pc/test/integration_test_helpers.h"
#include "rtc_base/gunit.h"
#include "rtc_base/virtual_socket_server.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
class PeerConnectionMediaChannelSplitTest
: public PeerConnectionIntegrationBaseTest,
public ::testing::WithParamInterface<std::string> {
protected:
PeerConnectionMediaChannelSplitTest()
: PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan,
/* field_trials = */ GetParam()) {}
};
int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
auto sender_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
if (sender_stats.size() != 1) {
ADD_FAILURE();
return 0;
}
if (!sender_stats[0]->nack_count.is_defined()) {
return 0;
}
return *sender_stats[0]->nack_count;
}
int NacksSentCount(PeerConnectionIntegrationWrapper& pc) {
rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
auto receiver_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
if (receiver_stats.size() != 1) {
ADD_FAILURE();
return 0;
}
if (!receiver_stats[0]->nack_count.is_defined()) {
return 0;
}
return *receiver_stats[0]->nack_count;
}
// Test disabled because it is flaky.
TEST_P(PeerConnectionMediaChannelSplitTest,
DISABLED_AudioPacketLossCausesNack) {
RTCConfiguration config;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
auto audio_transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
ASSERT_TRUE(audio_transceiver_or_error.ok());
auto send_transceiver = audio_transceiver_or_error.MoveValue();
// Munge the SDP to include NACK and RRTR on Opus, and remove all other
// codecs.
caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* desc) {
for (ContentInfo& content : desc->contents()) {
cricket::AudioContentDescription* media =
content.media_description()->as_audio();
std::vector<cricket::AudioCodec> codecs = media->codecs();
std::vector<cricket::AudioCodec> codecs_out;
for (cricket::AudioCodec codec : codecs) {
if (codec.name == "opus") {
codec.AddFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
codec.AddFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamRrtr, cricket::kParamValueEmpty));
codecs_out.push_back(codec);
}
}
EXPECT_FALSE(codecs_out.empty());
media->set_codecs(codecs_out);
}
});
caller()->CreateAndSetAndSignalOffer();
// Check for failure in helpers
ASSERT_FALSE(HasFailure());
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio(1);
ExpectNewFrames(media_expectations);
ASSERT_FALSE(HasFailure());
virtual_socket_server()->set_drop_probability(0.2);
// Wait until callee has sent at least one NACK.
// Note that due to stats caching, this might only be visible 50 ms
// after the nack was in fact sent.
EXPECT_TRUE_WAIT(NacksSentCount(*callee()) > 0, kDefaultTimeout);
ASSERT_FALSE(HasFailure());
virtual_socket_server()->set_drop_probability(0.0);
// Wait until caller has received at least one NACK
EXPECT_TRUE_WAIT(NacksReceivedCount(*caller()) > 0, kDefaultTimeout);
}
TEST_P(PeerConnectionMediaChannelSplitTest, VideoPacketLossCausesNack) {
RTCConfiguration config;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
auto video_transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());
ASSERT_TRUE(video_transceiver_or_error.ok());
auto send_transceiver = video_transceiver_or_error.MoveValue();
// Munge the SDP to include NACK and RRTR on VP8, and remove all other
// codecs.
caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* desc) {
for (ContentInfo& content : desc->contents()) {
cricket::VideoContentDescription* media =
content.media_description()->as_video();
std::vector<cricket::VideoCodec> codecs = media->codecs();
std::vector<cricket::VideoCodec> codecs_out;
for (cricket::VideoCodec codec : codecs) {
if (codec.name == "VP8") {
ASSERT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)));
codecs_out.push_back(codec);
}
}
EXPECT_FALSE(codecs_out.empty());
media->set_codecs(codecs_out);
}
});
caller()->CreateAndSetAndSignalOffer();
// Check for failure in helpers
ASSERT_FALSE(HasFailure());
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeVideo(1);
ExpectNewFrames(media_expectations);
ASSERT_FALSE(HasFailure());
virtual_socket_server()->set_drop_probability(0.2);
// Wait until callee has sent at least one NACK.
// Note that due to stats caching, this might only be visible 50 ms
// after the nack was in fact sent.
EXPECT_TRUE_WAIT(NacksSentCount(*callee()) > 0, kDefaultTimeout);
ASSERT_FALSE(HasFailure());
// Wait until caller has received at least one NACK
EXPECT_TRUE_WAIT(NacksReceivedCount(*caller()) > 0, kDefaultTimeout);
}
// Test that we can get capture start ntp time.
TEST_P(PeerConnectionMediaChannelSplitTest,
GetCaptureStartNtpTimeWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Get the remote audio track created on the receiver, so they can be used as
// GetStats filters.
auto receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(1u, receivers.size());
auto remote_audio_track = receivers[0]->track();
// Get the audio output level stats. Note that the level is not available
// until an RTCP packet has been received.
EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track.get())
->CaptureStartNtpTime() > 0,
2 * kMaxWaitForFramesMs);
}
INSTANTIATE_TEST_SUITE_P(PeerConnectionMediaChannelSplitTest,
PeerConnectionMediaChannelSplitTest,
Values("WebRTC-SplitMediaChannel/Disabled/",
"WebRTC-SplitMediaChannel/Enabled/"));
} // namespace
} // namespace webrtc

8
pc/rtp_sender_receiver_unittest.cc
View file

@ -37,7 +37,6 @@
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video/video_codec_constants.h"
#include "media/base/codec.h"
#include "media/base/delayable.h"
#include "media/base/fake_media_engine.h"
#include "media/base/media_channel.h"
#include "media/base/media_config.h"
@ -485,9 +484,10 @@ class RtpSenderReceiverTest
// Check that minimum Jitter Buffer delay is propagated to the underlying
// `media_channel`.
void VerifyRtpReceiverDelayBehaviour(cricket::Delayable* media_channel,
RtpReceiverInterface* receiver,
uint32_t ssrc) {
void VerifyRtpReceiverDelayBehaviour(
cricket::MediaReceiveChannelInterface* media_channel,
RtpReceiverInterface* receiver,
uint32_t ssrc) {
receiver->SetJitterBufferMinimumDelay(/*delay_seconds=*/0.5);
absl::optional<int> delay_ms =
media_channel->GetBaseMinimumPlayoutDelayMs(ssrc); // In milliseconds.

140
pc/rtp_transceiver.cc
View file

@ -202,8 +202,6 @@ RTCError RtpTransceiver::CreateChannel(
return RTCError(RTCErrorType::INTERNAL_ERROR,
"No media engine for mid=" + std::string(mid));
}
bool use_split_media_channel =
!context()->field_trials().IsDisabled("WebRTC-SplitMediaChannel");
std::unique_ptr<cricket::ChannelInterface> new_channel;
if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
@ -219,50 +217,34 @@ RTCError RtpTransceiver::CreateChannel(
AudioCodecPairId codec_pair_id = AudioCodecPairId::Create();
if (use_split_media_channel) {
std::unique_ptr<cricket::VoiceMediaSendChannelInterface>
media_send_channel = media_engine()->voice().CreateSendChannel(
call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
if (!media_send_channel) {
// TODO(bugs.webrtc.org/14912): Consider CHECK or reporting failure
return;
}
std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface>
media_receive_channel =
media_engine()->voice().CreateReceiveChannel(
call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel = media_receive_channel.get()](
const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VoiceChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), std::move(media_send_channel),
std::move(media_receive_channel), mid, srtp_required,
crypto_options, context()->ssrc_generator());
} else {
cricket::VoiceMediaChannel* media_channel =
media_engine()->voice().CreateMediaChannel(
cricket::MediaChannel::Role::kBoth, call_ptr, media_config,
audio_options, crypto_options, AudioCodecPairId::Create());
if (!media_channel) {
return;
}
new_channel = std::make_unique<cricket::VoiceChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), absl::WrapUnique(media_channel), mid,
srtp_required, crypto_options, context()->ssrc_generator());
std::unique_ptr<cricket::VoiceMediaSendChannelInterface>
media_send_channel = media_engine()->voice().CreateSendChannel(
call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
if (!media_send_channel) {
// TODO(bugs.webrtc.org/14912): Consider CHECK or reporting failure
return;
}
std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface>
media_receive_channel = media_engine()->voice().CreateReceiveChannel(
call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel =
media_receive_channel.get()](const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VoiceChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), std::move(media_send_channel),
std::move(media_receive_channel), mid, srtp_required, crypto_options,
context()->ssrc_generator());
});
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type());
@ -273,49 +255,33 @@ RTCError RtpTransceiver::CreateChannel(
context()->worker_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(context()->worker_thread());
if (use_split_media_channel) {
std::unique_ptr<cricket::VideoMediaSendChannelInterface>
media_send_channel = media_engine()->video().CreateSendChannel(
call_ptr, media_config, video_options, crypto_options,
video_bitrate_allocator_factory);
if (!media_send_channel) {
return;
}
std::unique_ptr<cricket::VideoMediaReceiveChannelInterface>
media_receive_channel =
media_engine()->video().CreateReceiveChannel(
call_ptr, media_config, video_options, crypto_options);
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel = media_receive_channel.get()](
const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VideoChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), std::move(media_send_channel),
std::move(media_receive_channel), mid, srtp_required,
crypto_options, context()->ssrc_generator());
} else {
cricket::VideoMediaChannel* media_channel =
media_engine()->video().CreateMediaChannel(
cricket::MediaChannel::Role::kBoth, call_ptr, media_config,
video_options, crypto_options, video_bitrate_allocator_factory);
if (!media_channel) {
return;
}
new_channel = std::make_unique<cricket::VideoChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), absl::WrapUnique(media_channel), mid,
srtp_required, crypto_options, context()->ssrc_generator());
std::unique_ptr<cricket::VideoMediaSendChannelInterface>
media_send_channel = media_engine()->video().CreateSendChannel(
call_ptr, media_config, video_options, crypto_options,
video_bitrate_allocator_factory);
if (!media_send_channel) {
return;
}
std::unique_ptr<cricket::VideoMediaReceiveChannelInterface>
media_receive_channel = media_engine()->video().CreateReceiveChannel(
call_ptr, media_config, video_options, crypto_options);
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel =
media_receive_channel.get()](const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VideoChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), std::move(media_send_channel),
std::move(media_receive_channel), mid, srtp_required, crypto_options,
context()->ssrc_generator());
});
}
if (!new_channel) {

4
pc/rtp_transceiver_unittest.cc
View file

@ -382,7 +382,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions,
EXPECT_CALL(*mock_channel, SetFirstPacketReceivedCallback(_));
EXPECT_CALL(*mock_channel, media_type())
.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
EXPECT_CALL(*mock_channel, media_send_channel())
EXPECT_CALL(*mock_channel, voice_media_send_channel())
.WillRepeatedly(Return(nullptr));
EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name));
EXPECT_CALL(*mock_channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
@ -415,7 +415,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, ReturnsNegotiatedHdrExts) {
EXPECT_CALL(*mock_channel, SetFirstPacketReceivedCallback(_));
EXPECT_CALL(*mock_channel, media_type())
.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
EXPECT_CALL(*mock_channel, media_send_channel())
EXPECT_CALL(*mock_channel, voice_media_send_channel())
.WillRepeatedly(Return(nullptr));
EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name));
EXPECT_CALL(*mock_channel, SetRtpTransport(_)).WillRepeatedly(Return(true));

28
pc/test/mock_channel_interface.h
View file

@ -14,6 +14,7 @@
#include <string>
#include <vector>
#include "media/base/media_channel.h"
#include "pc/channel_interface.h"
#include "test/gmock.h"
@ -27,12 +28,27 @@ class MockChannelInterface : public cricket::ChannelInterface {
MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
MOCK_METHOD(VideoChannel*, AsVideoChannel, (), (override));
MOCK_METHOD(VoiceChannel*, AsVoiceChannel, (), (override));
MOCK_METHOD(MediaChannel*, media_send_channel, (), (override));
MOCK_METHOD(VoiceMediaChannel*, voice_media_send_channel, (), (override));
MOCK_METHOD(VideoMediaChannel*, video_media_send_channel, (), (override));
MOCK_METHOD(MediaChannel*, media_receive_channel, (), (override));
MOCK_METHOD(VoiceMediaChannel*, voice_media_receive_channel, (), (override));
MOCK_METHOD(VideoMediaChannel*, video_media_receive_channel, (), (override));
MOCK_METHOD(MediaSendChannelInterface*, media_send_channel, (), (override));
MOCK_METHOD(VoiceMediaSendChannelInterface*,
voice_media_send_channel,
(),
(override));
MOCK_METHOD(VideoMediaSendChannelInterface*,
video_media_send_channel,
(),
(override));
MOCK_METHOD(MediaReceiveChannelInterface*,
media_receive_channel,
(),
(override));
MOCK_METHOD(VoiceMediaReceiveChannelInterface*,
voice_media_receive_channel,
(),
(override));
MOCK_METHOD(VideoMediaReceiveChannelInterface*,
video_media_receive_channel,
(),
(override));
MOCK_METHOD(absl::string_view, transport_name, (), (const, override));
MOCK_METHOD(const std::string&, mid, (), (const, override));
MOCK_METHOD(void, Enable, (bool), (override));

190
pc/test/mock_voice_media_channel.h → pc/test/mock_voice_media_receive_channel_interface.h
View file

@ -1,5 +1,5 @@
/*
* Copyright 2021 The WebRTC project authors. All Rights Reserved.
* Copyright 2023 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
@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef PC_TEST_MOCK_VOICE_MEDIA_CHANNEL_H_
#define PC_TEST_MOCK_VOICE_MEDIA_CHANNEL_H_
#ifndef PC_TEST_MOCK_VOICE_MEDIA_RECEIVE_CHANNEL_INTERFACE_H_
#define PC_TEST_MOCK_VOICE_MEDIA_RECEIVE_CHANNEL_INTERFACE_H_
#include <memory>
#include <set>
@ -23,102 +23,16 @@
#include "test/gmock.h"
#include "test/gtest.h"
using ::testing::InvokeWithoutArgs;
using ::testing::Mock;
namespace cricket {
class MockVoiceMediaChannel : public VoiceMediaChannel {
class MockVoiceMediaReceiveChannelInterface
: public VoiceMediaReceiveChannelInterface {
public:
MockVoiceMediaChannel(MediaChannel::Role role,
webrtc::TaskQueueBase* network_thread)
: VoiceMediaChannel(role, network_thread) {}
MockVoiceMediaReceiveChannelInterface() {
ON_CALL(*this, AsVoiceReceiveChannel).WillByDefault(testing::Return(this));
}
MOCK_METHOD(void,
SetInterface,
(MediaChannelNetworkInterface * iface),
(override));
MOCK_METHOD(void,
OnPacketReceived,
(const webrtc::RtpPacketReceived& packet),
(override));
MOCK_METHOD(void,
OnPacketSent,
(const rtc::SentPacket& sent_packet),
(override));
MOCK_METHOD(void, OnReadyToSend, (bool ready), (override));
MOCK_METHOD(void,
OnNetworkRouteChanged,
(absl::string_view transport_name,
const rtc::NetworkRoute& network_route),
(override));
MOCK_METHOD(void, SetExtmapAllowMixed, (bool extmap_allow_mixed), (override));
MOCK_METHOD(bool, ExtmapAllowMixed, (), (const, override));
MOCK_METHOD(bool, HasNetworkInterface, (), (const, override));
MOCK_METHOD(bool, AddSendStream, (const StreamParams& sp), (override));
MOCK_METHOD(bool, RemoveSendStream, (uint32_t ssrc), (override));
MOCK_METHOD(bool, AddRecvStream, (const StreamParams& sp), (override));
MOCK_METHOD(bool, RemoveRecvStream, (uint32_t ssrc), (override));
MOCK_METHOD(void, ResetUnsignaledRecvStream, (), (override));
MOCK_METHOD(absl::optional<uint32_t>,
GetUnsignaledSsrc,
(),
(const, override));
MOCK_METHOD(void,
ChooseReceiverReportSsrc,
(const std::set<uint32_t>&),
(override));
MOCK_METHOD(absl::optional<Codec>, GetSendCodec, (), (const, override));
MOCK_METHOD(bool, SendCodecHasNack, (), (const, override));
MOCK_METHOD(void,
SetSsrcListChangedCallback,
(absl::AnyInvocable<void(const std::set<uint32_t>&)>),
(override));
MOCK_METHOD(void,
SetSendCodecChangedCallback,
(absl::AnyInvocable<void()>),
(override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override));
MOCK_METHOD(int, GetRtpSendTimeExtnId, (), (const, override));
MOCK_METHOD(
void,
SetFrameEncryptor,
(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor),
(override));
MOCK_METHOD(
void,
SetFrameDecryptor,
(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
(override));
MOCK_METHOD(webrtc::RtpParameters,
GetRtpSendParameters,
(uint32_t ssrc),
(const, override));
MOCK_METHOD(webrtc::RTCError,
SetRtpSendParameters,
(uint32_t ssrc,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback),
(override));
MOCK_METHOD(
void,
SetEncoderToPacketizerFrameTransformer,
(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
(override));
MOCK_METHOD(
void,
SetDepacketizerToDecoderFrameTransformer,
(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
(override));
MOCK_METHOD(bool,
SetSendParameters,
(const AudioSendParameters& params),
(override));
// VoiceMediaReceiveChannelInterface
MOCK_METHOD(bool,
SetRecvParameters,
(const AudioRecvParameters& params),
@ -127,34 +41,20 @@ class MockVoiceMediaChannel : public VoiceMediaChannel {
GetRtpReceiveParameters,
(uint32_t ssrc),
(const, override));
MOCK_METHOD(std::vector<webrtc::RtpSource>,
GetSources,
(uint32_t ssrc),
(const, override));
MOCK_METHOD(webrtc::RtpParameters,
GetDefaultRtpReceiveParameters,
(),
(const, override));
MOCK_METHOD(void, SetPlayout, (bool playout), (override));
MOCK_METHOD(void, SetSend, (bool send), (override));
MOCK_METHOD(bool,
SetAudioSend,
(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source),
(override));
MOCK_METHOD(bool,
SetOutputVolume,
(uint32_t ssrc, double volume),
(override));
MOCK_METHOD(bool, SetDefaultOutputVolume, (double volume), (override));
MOCK_METHOD(bool, CanInsertDtmf, (), (override));
MOCK_METHOD(bool,
InsertDtmf,
(uint32_t ssrc, int event, int duration),
(override));
MOCK_METHOD(bool, GetSendStats, (VoiceMediaSendInfo * info), (override));
MOCK_METHOD(bool,
GetReceiveStats,
(VoiceMediaReceiveInfo * info, bool get_and_clear_legacy_stats),
(override));
MOCK_METHOD(void,
SetRawAudioSink,
(uint32_t ssrc, std::unique_ptr<webrtc::AudioSinkInterface> sink),
@ -163,11 +63,56 @@ class MockVoiceMediaChannel : public VoiceMediaChannel {
SetDefaultRawAudioSink,
(std::unique_ptr<webrtc::AudioSinkInterface> sink),
(override));
MOCK_METHOD(std::vector<webrtc::RtpSource>,
GetSources,
(uint32_t ssrc),
(const, override));
MOCK_METHOD(bool,
GetStats,
(VoiceMediaReceiveInfo * stats, bool reset_legacy),
(override));
MOCK_METHOD(void, SetReceiveNackEnabled, (bool enabled), (override));
MOCK_METHOD(void, SetReceiveNonSenderRttEnabled, (bool enabled), (override));
// MediaReceiveChannelInterface
MOCK_METHOD(VideoMediaReceiveChannelInterface*,
AsVideoReceiveChannel,
(),
(override));
MOCK_METHOD(VoiceMediaReceiveChannelInterface*,
AsVoiceReceiveChannel,
(),
(override));
MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
MOCK_METHOD(bool, AddRecvStream, (const StreamParams& sp), (override));
MOCK_METHOD(bool, RemoveRecvStream, (uint32_t ssrc), (override));
MOCK_METHOD(void, ResetUnsignaledRecvStream, (), (override));
MOCK_METHOD(void,
SetInterface,
(MediaChannelNetworkInterface * iface),
(override));
MOCK_METHOD(void,
OnPacketReceived,
(const webrtc::RtpPacketReceived& packet),
(override));
MOCK_METHOD(absl::optional<uint32_t>,
GetUnsignaledSsrc,
(),
(const, override));
MOCK_METHOD(void,
ChooseReceiverReportSsrc,
(const std::set<uint32_t>& choices),
(override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override));
MOCK_METHOD(
void,
SetFrameDecryptor,
(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
(override));
MOCK_METHOD(
void,
SetDepacketizerToDecoderFrameTransformer,
(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
(override));
MOCK_METHOD(bool,
SetBaseMinimumPlayoutDelayMs,
(uint32_t ssrc, int delay_ms),
@ -176,11 +121,10 @@ class MockVoiceMediaChannel : public VoiceMediaChannel {
GetBaseMinimumPlayoutDelayMs,
(uint32_t ssrc),
(const, override));
MOCK_METHOD(bool, SenderNackEnabled, (), (const, override));
MOCK_METHOD(bool, SenderNonSenderRttEnabled, (), (const, override));
MOCK_METHOD(void, SetReceiveNackEnabled, (bool enabled), (override));
MOCK_METHOD(void, SetReceiveNonSenderRttEnabled, (bool enabled), (override));
};
static_assert(!std::is_abstract_v<MockVoiceMediaReceiveChannelInterface>, "");
} // namespace cricket
#endif // PC_TEST_MOCK_VOICE_MEDIA_CHANNEL_H_
#endif // PC_TEST_MOCK_VOICE_MEDIA_RECEIVE_CHANNEL_INTERFACE_H_