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Move remaining traces of VoiceEngine

Browse Source 
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
  utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.

NOPRESUBMIT=true

Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
This commit is contained in:
Fredrik Solenberg
2018-01-17 11:18:31 +01:00
committed by Commit Bot
parent 18bc3e19c4
commit a8b7c7f4c6
50 changed files with 79 additions and 282 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
.gn
View File

@ -42,7 +42,6 @@ check_targets = [
"//system_wrappers/*",
"//test/*",
"//video/*",
"//voice_engine/*",
"//third_party/libyuv/*",
]

2
BUILD.gn
View File

@ -63,7 +63,6 @@ if (!build_with_chromium) {
"video:screenshare_loopback",
"video:sv_loopback",
"video:video_loopback",
"voice_engine:voice_engine_unittests",
]
if (is_android) {
deps += [
@ -342,7 +341,6 @@ if (!build_with_chromium) {
"sdk",
"system_wrappers:system_wrappers_default",
"video",
"voice_engine",
]
if (build_with_mozilla) {

15
WATCHLISTS
View File

@ -23,8 +23,7 @@
'filepath': '^[^/]*$|^webrtc/[^/]*$|^webrtc/build/.*',
},
'documented_interfaces': {
'filepath': '^webrtc/[^/]*\.h$|'\
'webrtc/voice_engine/include/.*',
'filepath': '^webrtc/[^/]*\.h$',
},
'build_files': {
'filepath': '\.gyp$|\.gypi$|Android\.mk$',
@ -50,9 +49,6 @@
'video': {
'filepath': 'webrtc/video/.*',
},
'voice_engine': {
'filepath': 'webrtc/voice_engine/.*',
},
'common_audio': {
'filepath': 'webrtc/common_audio/.*',
},
@ -126,20 +122,11 @@
'call': ['mflodman@webrtc.org',
'solenberg@webrtc.org',
'stefan@webrtc.org'],
'media_engine': ['solenberg@webrtc.org'],
'video': ['mflodman@webrtc.org',
'stefan@webrtc.org',
'video-team@agora.io',
'yujie.mao@webrtc.org',
'zhengzhonghou@agora.io'],
'voice_engine': ['alessiob@webrtc.org',
'andrew@webrtc.org',
'audio-team@agora.io',
'henrika@webrtc.org',
'henrik.lundin@webrtc.org',
'minyue@webrtc.org',
'peah@webrtc.org',
'solenberg@webrtc.org'],
'video_capture': ['mflodman@webrtc.org',
'perkj@webrtc.org',
'sdk-team@agora.io',

4
api/DEPS
View File

@ -7,10 +7,6 @@ include_rules = [
]
specific_include_rules = {
"peerconnection_jni\.cc": [
"+voice_engine",
],
# TODO(ossu): Remove this exception when {builtin_,}audio_encoder_factory.h
# has moved to api/.
"peerconnectioninterface\.h": [

31
audio/BUILD.gn
View File

@ -14,6 +14,8 @@ if (is_android) {
rtc_static_library("audio") {
sources = [
"audio_level.cc",
"audio_level.h",
"audio_receive_stream.cc",
"audio_receive_stream.h",
"audio_send_stream.cc",
@ -22,11 +24,19 @@ rtc_static_library("audio") {
"audio_state.h",
"audio_transport_impl.cc",
"audio_transport_impl.h",
"channel.cc",
"channel.h",
"channel_proxy.cc",
"channel_proxy.h",
"conversion.h",
"null_audio_poller.cc",
"null_audio_poller.h",
"remix_resample.cc",
"remix_resample.h",
"time_interval.cc",
"time_interval.h",
"transport_feedback_packet_loss_tracker.cc",
"transport_feedback_packet_loss_tracker.h",
]
if (!build_with_chromium && is_clang) {
@ -36,15 +46,23 @@ rtc_static_library("audio") {
deps = [
"..:webrtc_common",
"../api:array_view",
"../api:audio_mixer_api",
"../api:call_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../call:bitrate_allocator",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/audio_coding",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:audio_network_adaptor_config",
"../modules/audio_coding:cng",
"../modules/audio_device",
"../modules/audio_processing",
@ -53,14 +71,16 @@ rtc_static_library("audio") {
"../modules/pacing:pacing",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../voice_engine",
"../voice_engine:audio_level",
"../system_wrappers:metrics_api",
"utility:audio_frame_operations",
]
}
@ -94,7 +114,10 @@ if (rtc_include_tests) {
"audio_send_stream_tests.cc",
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
"mock_voe_channel_proxy.h",
"remix_resample_unittest.cc",
"time_interval_unittest.cc",
"transport_feedback_packet_loss_tracker_unittest.cc",
]
deps = [
":audio",
@ -104,7 +127,9 @@ if (rtc_include_tests) {
"../call:mock_rtp_interfaces",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../common_audio",
"../logging:mocks",
"../modules:module_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing_statistics",
@ -116,6 +141,7 @@ if (rtc_include_tests) {
"../modules/pacing:pacing",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_task_queue",
@ -125,7 +151,6 @@ if (rtc_include_tests) {
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_support",
"../voice_engine",
"utility:utility_tests",
"//testing/gmock",
"//testing/gtest",

3
audio/DEPS
View File

@ -5,14 +5,15 @@ include_rules = [
"+modules/audio_coding",
"+modules/audio_device",
"+modules/audio_mixer",
"+modules/audio_processing",
"+modules/audio_processing/include",
"+modules/bitrate_controller",
"+modules/congestion_controller",
"+modules/pacing",
"+modules/remote_bitrate_estimator",
"+modules/rtp_rtcp",
"+modules/utility",
"+system_wrappers",
"+voice_engine",
]
specific_include_rules = {

2
voice_engine/audio_level.cc → audio/audio_level.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "voice_engine/audio_level.h"
#include "audio/audio_level.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/include/module_common_types.h"

7
voice_engine/audio_level.h → audio/audio_level.h
View File

@ -8,12 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef VOICE_ENGINE_AUDIO_LEVEL_H_
#define VOICE_ENGINE_AUDIO_LEVEL_H_
#ifndef AUDIO_AUDIO_LEVEL_H_
#define AUDIO_AUDIO_LEVEL_H_
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@ -57,4 +56,4 @@ class AudioLevel {
} // namespace voe
} // namespace webrtc
#endif // VOICE_ENGINE_AUDIO_LEVEL_H_
#endif // AUDIO_AUDIO_LEVEL_H_

2
audio/audio_receive_stream.cc
View File

@ -16,6 +16,7 @@
#include "api/call/audio_sink.h"
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/channel_proxy.h"
#include "audio/conversion.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
@ -24,7 +25,6 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/timeutils.h"
#include "voice_engine/channel_proxy.h"
namespace webrtc {

2
audio/audio_receive_stream_unittest.cc
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@ -15,6 +15,7 @@
#include "api/test/mock_audio_mixer.h"
#include "audio/audio_receive_stream.h"
#include "audio/conversion.h"
#include "audio/mock_voe_channel_proxy.h"
#include "call/rtp_stream_receiver_controller.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_device/include/mock_audio_device.h"
@ -24,7 +25,6 @@
#include "modules/rtp_rtcp/source/byte_io.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_voe_channel_proxy.h"
namespace webrtc {
namespace test {

2
audio/audio_send_stream.cc
View File

@ -15,6 +15,7 @@
#include <vector>
#include "audio/audio_state.h"
#include "audio/channel_proxy.h"
#include "audio/conversion.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
@ -28,7 +29,6 @@
#include "rtc_base/task_queue.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
#include "voice_engine/channel_proxy.h"
namespace webrtc {
namespace internal {

2
audio/audio_send_stream.h
View File

@ -15,6 +15,7 @@
#include <vector>
#include "audio/time_interval.h"
#include "audio/transport_feedback_packet_loss_tracker.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/bitrate_allocator.h"
@ -22,7 +23,6 @@
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/transport_feedback_packet_loss_tracker.h"
namespace webrtc {
class RtcEventLog;

2
audio/audio_send_stream_unittest.cc
View File

@ -15,6 +15,7 @@
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/conversion.h"
#include "audio/mock_voe_channel_proxy.h"
#include "call/fake_rtp_transport_controller_send.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
@ -32,7 +33,6 @@
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
#include "test/mock_audio_encoder_factory.h"
#include "test/mock_voe_channel_proxy.h"
namespace webrtc {
namespace test {

2
audio/audio_transport_impl.cc
View File

@ -14,10 +14,10 @@
#include <memory>
#include <utility>
#include "audio/remix_resample.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/audio_send_stream.h"
#include "rtc_base/logging.h"
#include "voice_engine/utility.h"
namespace webrtc {

2
audio/audio_transport_impl.h
View File

@ -14,6 +14,7 @@
#include <vector>
#include "api/audio/audio_mixer.h"
#include "audio/audio_level.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
@ -22,7 +23,6 @@
#include "rtc_base/criticalsection.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread_annotations.h"
#include "voice_engine/audio_level.h"
namespace webrtc {

12
voice_engine/channel.cc → audio/channel.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "voice_engine/channel.h"
#include "audio/channel.h"
#include <algorithm>
#include <map>
@ -365,10 +365,7 @@ bool Channel::SendRtp(const uint8_t* data,
return false;
}
uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len;
if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
if (!_transportPtr->SendRtp(data, len, options)) {
RTC_LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed";
return false;
}
@ -383,10 +380,7 @@ bool Channel::SendRtcp(const uint8_t* data, size_t len) {
return false;
}
uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len;
int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
int n = _transportPtr->SendRtcp(data, len);
if (n < 0) {
RTC_LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed";
return false;

11
voice_engine/channel.h → audio/channel.h
View File

@ -8,16 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef VOICE_ENGINE_CHANNEL_H_
#define VOICE_ENGINE_CHANNEL_H_
#ifndef AUDIO_CHANNEL_H_
#define AUDIO_CHANNEL_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/optional.h"
#include "audio/audio_level.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_processing/rms_level.h"
@ -29,7 +33,6 @@
#include "rtc_base/event.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/audio_level.h"
namespace rtc {
class TimestampWrapAroundHandler;
@ -410,4 +413,4 @@ class Channel
} // namespace voe
} // namespace webrtc
#endif // VOICE_ENGINE_CHANNEL_H_
#endif // AUDIO_CHANNEL_H_

2
voice_engine/channel_proxy.cc → audio/channel_proxy.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "voice_engine/channel_proxy.h"
#include "audio/channel_proxy.h"
#include <utility>

9
voice_engine/channel_proxy.h → audio/channel_proxy.h
View File

@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef VOICE_ENGINE_CHANNEL_PROXY_H_
#define VOICE_ENGINE_CHANNEL_PROXY_H_
#ifndef AUDIO_CHANNEL_PROXY_H_
#define AUDIO_CHANNEL_PROXY_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
@ -18,11 +19,11 @@
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/rtpreceiverinterface.h"
#include "audio/channel.h"
#include "call/rtp_packet_sink_interface.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/channel.h"
namespace webrtc {
@ -141,4 +142,4 @@ class ChannelProxy : public RtpPacketSinkInterface {
} // namespace voe
} // namespace webrtc
#endif // VOICE_ENGINE_CHANNEL_PROXY_H_
#endif // AUDIO_CHANNEL_PROXY_H_

11
test/mock_voe_channel_proxy.h → audio/mock_voe_channel_proxy.h
View File

@ -8,14 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef TEST_MOCK_VOE_CHANNEL_PROXY_H_
#define TEST_MOCK_VOE_CHANNEL_PROXY_H_
#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "audio/channel_proxy.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "test/gmock.h"
#include "voice_engine/channel_proxy.h"
namespace webrtc {
namespace test {
@ -101,4 +104,4 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
} // namespace test
} // namespace webrtc
#endif // TEST_MOCK_VOE_CHANNEL_PROXY_H_
#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_

2
voice_engine/utility.cc → audio/remix_resample.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "voice_engine/utility.h"
#include "audio/remix_resample.h"
#include "audio/utility/audio_frame_operations.h"
#include "common_audio/resampler/include/push_resampler.h"

11
voice_engine/utility.h → audio/remix_resample.h
View File

@ -8,15 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* Contains functions often used by different parts of VoiceEngine.
*/
#ifndef VOICE_ENGINE_UTILITY_H_
#define VOICE_ENGINE_UTILITY_H_
#ifndef AUDIO_REMIX_RESAMPLE_H_
#define AUDIO_REMIX_RESAMPLE_H_
#include "common_audio/resampler/include/push_resampler.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@ -48,4 +43,4 @@ void RemixAndResample(const int16_t* src_data,
} // namespace voe
} // namespace webrtc
#endif // VOICE_ENGINE_UTILITY_H_
#endif // AUDIO_REMIX_RESAMPLE_H_

2
voice_engine/utility_unittest.cc → audio/remix_resample_unittest.cc
View File

@ -10,12 +10,12 @@
#include <math.h>
#include "audio/remix_resample.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/format_macros.h"
#include "test/gtest.h"
#include "voice_engine/utility.h"
namespace webrtc {
namespace voe {

2
voice_engine/transport_feedback_packet_loss_tracker.cc → audio/transport_feedback_packet_loss_tracker.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "voice_engine/transport_feedback_packet_loss_tracker.h"
#include "audio/transport_feedback_packet_loss_tracker.h"
#include <limits>
#include <utility>

7
voice_engine/transport_feedback_packet_loss_tracker.h → audio/transport_feedback_packet_loss_tracker.h
View File

@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
#define VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
#ifndef AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
#define AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
#include <map>
#include <vector>
#include "api/optional.h"
#include "modules/include/module_common_types.h"
@ -138,4 +139,4 @@ class TransportFeedbackPacketLossTracker final {
} // namespace webrtc
#endif // VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
#endif // AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_

2
voice_engine/transport_feedback_packet_loss_tracker_unittest.cc → audio/transport_feedback_packet_loss_tracker_unittest.cc
View File

@ -13,12 +13,12 @@
#include <numeric>
#include <vector>
#include "audio/transport_feedback_packet_loss_tracker.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "rtc_base/checks.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "voice_engine/transport_feedback_packet_loss_tracker.h"
namespace webrtc {

1
call/BUILD.gn
View File

@ -271,7 +271,6 @@ if (rtc_include_tests) {
"../test:test_support",
"../test:video_test_common",
"../video",
"../voice_engine",
"//testing/gtest",
]
if (!build_with_chromium && is_clang) {

1
call/DEPS
View File

@ -11,7 +11,6 @@ include_rules = [
"+modules/rtp_rtcp",
"+modules/utility",
"+system_wrappers",
"+voice_engine",
"+video",
]

3
call/audio_receive_stream.h
View File

@ -98,9 +98,6 @@ class AudioReceiveStream {
Transport* rtcp_send_transport = nullptr;
// TODO(solenberg): Remove once clients don't use it anymore.
int voe_channel_id = -1;
// NetEq settings.
size_t jitter_buffer_max_packets = 50;
bool jitter_buffer_fast_accelerate = false;

3
call/audio_send_stream.h
View File

@ -87,9 +87,6 @@ class AudioSendStream {
// the entire life of the AudioSendStream and is owned by the API client.
Transport* send_transport = nullptr;
// TODO(solenberg): Remove once clients don't use it anymore.
int voe_channel_id = -1;
// Bitrate limits used for variable audio bitrate streams. Set both to -1 to
// disable audio bitrate adaptation.
// Note: This is still an experimental feature and not ready for real usage.

4
call/audio_state.h
View File

@ -19,16 +19,12 @@ namespace webrtc {
class AudioDeviceModule;
class AudioProcessing;
class AudioTransport;
class VoiceEngine;
// AudioState holds the state which must be shared between multiple instances of
// webrtc::Call for audio processing purposes.
class AudioState : public rtc::RefCountInterface {
public:
struct Config {
// TODO(solenberg): Remove once clients don't use it anymore.
VoiceEngine* voice_engine = nullptr;
// The audio mixer connected to active receive streams. One per
// AudioState.
rtc::scoped_refptr<AudioMixer> audio_mixer;

2
media/BUILD.gn
View File

@ -317,7 +317,6 @@ rtc_static_library("rtc_audio_video") {
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
"../voice_engine",
]
}
@ -616,7 +615,6 @@ if (rtc_include_tests) {
"../test:audio_codec_mocks",
"../test:test_support",
"../test:video_test_common",
"../voice_engine:voice_engine",
]
}
}

1
media/DEPS
View File

@ -14,7 +14,6 @@ include_rules = [
"+pc",
"+sound",
"+system_wrappers",
"+voice_engine",
"+usrsctplib",
"+third_party/libyuv",
]

1
modules/audio_mixer/DEPS
View File

@ -10,5 +10,4 @@ include_rules = [
"+modules/rtp_rtcp",
"+modules/utility",
"+system_wrappers",
"+voice_engine",
]

1
modules/remote_bitrate_estimator/BUILD.gn
View File

@ -155,7 +155,6 @@ if (rtc_include_tests) {
"../../system_wrappers:field_trial_api",
"../../test:perf_test",
"../../test:test_support",
"../../voice_engine",
"../bitrate_controller",
"../congestion_controller",
"../congestion_controller:delay_based_bwe",

6
modules/remote_bitrate_estimator/DEPS
View File

@ -2,9 +2,3 @@ include_rules = [
"+logging/rtc_event_log",
"+system_wrappers",
]
specific_include_rules = {
"nada\.h": [
"+voice_engine",
],
}

1
modules/remote_bitrate_estimator/test/estimators/nada.h
View File

@ -24,7 +24,6 @@
#include "modules/include/module_common_types.h"
#include "modules/remote_bitrate_estimator/test/bwe.h"
#include "rtc_base/constructormagic.h"
#include "voice_engine/channel.h"
namespace webrtc {

1
pc/DEPS
View File

@ -18,7 +18,6 @@ include_rules = [
specific_include_rules = {
"androidtestinitializer\.cc": [
"+base/android", # Allowed only for Android tests.
"+voice_engine",
],
"srtpfilter_unittest\.cc": [
"+crypto",

1
sdk/android/BUILD.gn
View File

@ -111,7 +111,6 @@ rtc_static_library("audio_jni") {
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../modules/audio_processing:audio_processing",
"../../rtc_base:rtc_base_approved",
"../../voice_engine:voice_engine",
]
}

1
sdk/android/src/jni/DEPS
View File

@ -12,5 +12,4 @@ include_rules = [
"+modules/video_coding",
"+pc",
"+system_wrappers/include",
"+voice_engine/include/voe_base.h",
]

2
test/BUILD.gn
View File

@ -567,7 +567,6 @@ rtc_source_set("test_common") {
"layer_filtering_transport.cc",
"layer_filtering_transport.h",
"mock_transport.h",
"mock_voe_channel_proxy.h",
"null_transport.cc",
"null_transport.h",
"rtp_rtcp_observer.h",
@ -628,7 +627,6 @@ rtc_source_set("test_common") {
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../video",
"../voice_engine",
"//testing/gmock",
"//testing/gtest",
]

1
test/DEPS
View File

@ -16,7 +16,6 @@ include_rules = [
"+modules/video_coding",
"+sdk",
"+system_wrappers",
"+voice_engine",
"+third_party/libyuv",
]

2
test/fuzzers/BUILD.gn
View File

@ -416,9 +416,9 @@ webrtc_fuzzer_test("transport_feedback_packet_loss_tracker_fuzzer") {
]
deps = [
"../../api:array_view",
"../../audio",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:rtc_base_approved",
"../../voice_engine",
]
}
rtc_static_library("audio_processing_fuzzer_helper") {

2
test/fuzzers/DEPS
View File

@ -1,3 +1,3 @@
include_rules = [
"+webrtc",
"+audio",
]

2
test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
View File

@ -11,10 +11,10 @@
#include <algorithm>
#include "api/array_view.h"
#include "audio/transport_feedback_packet_loss_tracker.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "voice_engine/transport_feedback_packet_loss_tracker.h"
namespace webrtc {

2
video/BUILD.gn
View File

@ -94,7 +94,6 @@ rtc_static_library("video") {
"../rtc_base:sequenced_task_checker",
"../rtc_base:weak_ptr",
"../system_wrappers",
"../voice_engine",
]
if (!build_with_mozilla) {
@ -130,7 +129,6 @@ if (rtc_include_tests) {
"../test:test_support_test_artifacts",
"../test:video_test_common",
"../test:video_test_support",
"../voice_engine",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).

1
video/DEPS
View File

@ -15,5 +15,4 @@ include_rules = [
"+modules/video_capture",
"+modules/video_processing",
"+system_wrappers",
"+voice_engine",
]

148
voice_engine/BUILD.gn
View File

@ -1,148 +0,0 @@
# Copyright (c) 2014 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.
import("../webrtc.gni")
visibility = [ ":*" ]
rtc_static_library("voice_engine") {
visibility += [ "*" ]
sources = [
"channel.cc",
"channel.h",
"channel_proxy.cc",
"channel_proxy.h",
"transport_feedback_packet_loss_tracker.cc",
"transport_feedback_packet_loss_tracker.h",
"utility.cc",
"utility.h",
]
if (is_win) {
cflags = [
# TODO(kjellander): Bug 261: fix this warning.
"/wd4373", # Virtual function override.
]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":audio_level",
"..:webrtc_common",
"../:typedefs",
"../api:array_view",
"../api:audio_mixer_api",
"../api:call_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:refcountedbase",
"../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
"../audio/utility:audio_frame_operations",
"../call:rtp_interfaces",
"../common_audio",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/audio_coding",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:audio_network_adaptor_config",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/bitrate_controller",
"../modules/media_file",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
]
}
rtc_static_library("audio_level") {
visibility += [
":voice_engine",
"../audio:audio",
]
sources = [
"audio_level.cc",
"audio_level.h",
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../common_audio",
"../modules:module_api",
"../rtc_base:rtc_base_approved",
]
}
if (rtc_include_tests) {
rtc_test("voice_engine_unittests") {
visibility += webrtc_default_visibility
deps = [
":voice_engine",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../common_audio",
"../modules:module_api",
"../modules/audio_coding",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/media_file",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../modules/video_capture:video_capture",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../system_wrappers",
"../test:test_common",
"../test:test_main",
"../test:video_test_common",
"//testing/gmock",
"//testing/gtest",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
shard_timeout = 900
}
sources = [
"transport_feedback_packet_loss_tracker_unittest.cc",
"utility_unittest.cc",
]
data = [
"../resources/utility/encapsulated_pcm16b_8khz.wav",
"../resources/utility/encapsulated_pcmu_8khz.wav",
]
if (is_win) {
cflags = [
# TODO(kjellander): Bug 261: fix this warning.
"/wd4373", # Virtual function override.
]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}

14
voice_engine/DEPS
View File

@ -1,14 +0,0 @@
include_rules = [
"+audio/utility/audio_frame_operations.h",
"+call",
"+common_audio",
"+logging/rtc_event_log",
"+modules/audio_coding",
"+modules/audio_device",
"+modules/audio_processing",
"+modules/media_file",
"+modules/pacing",
"+modules/rtp_rtcp",
"+modules/utility",
"+system_wrappers",
]

10
voice_engine/OWNERS
View File

@ -1,10 +0,0 @@
henrikg@webrtc.org
henrika@webrtc.org
niklas.enbom@webrtc.org
solenberg@webrtc.org
# These are for the common case of adding or renaming files. If you're doing
# structural changes, please get a review from a reviewer in this file.
per-file *.gn=*
per-file *.gni=*