Moving/renaming webrtc/common.h.
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Original-Commit-Position: refs/heads/master@{#14109}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 88499ecaca050be2dd21fff17140d6681c3bac95
diff --git a/BUILD.gn b/BUILD.gn
index f42cc24..f9845a3 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -325,9 +325,6 @@
rtc_source_set("webrtc_common") {
sources = [
- "audio_sink.h",
- "common.cc",
- "common.h",
"common_types.cc",
"common_types.h",
"config.cc",
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 81fbdb7..b980a5e 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -159,9 +159,9 @@
FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
audio_rtp_speed);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
- Config voe_config;
- voe_config.Set<VoicePacing>(new VoicePacing(true));
- int send_channel_id = voe_base->CreateChannel(voe_config);
+ VoEBase::ChannelConfig config;
+ config.enable_voice_pacing = true;
+ int send_channel_id = voe_base->CreateChannel(config);
int recv_channel_id = voe_base->CreateChannel();
AudioState::Config send_audio_state_config;
diff --git a/common.gyp b/common.gyp
index e5d41da..3b5fe90 100644
--- a/common.gyp
+++ b/common.gyp
@@ -12,8 +12,6 @@
'target_name': 'webrtc_common',
'type': 'static_library',
'sources': [
- 'common.cc',
- 'common.h',
'common_types.cc',
'common_types.h',
'config.h',
diff --git a/common_types.h b/common_types.h
index 1ef5f91..ea040fd 100644
--- a/common_types.h
+++ b/common_types.h
@@ -52,8 +52,6 @@
namespace webrtc {
-class Config;
-
class RewindableStream {
public:
virtual ~RewindableStream() {}
diff --git a/config.h b/config.h
index d932eda..9303f54 100644
--- a/config.h
+++ b/config.h
@@ -17,7 +17,6 @@
#include <vector>
#include "webrtc/base/optional.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/typedefs.h"
@@ -170,35 +169,6 @@
rtc::Optional<VideoDecoderH264Settings> h264_extra_settings;
};
-// Controls the capacity of the packet buffer in NetEq. The capacity is the
-// maximum number of packets that the buffer can contain. If the limit is
-// exceeded, the buffer will be flushed. The capacity does not affect the actual
-// audio delay in the general case, since this is governed by the target buffer
-// level (calculated from the jitter profile). It is only in the rare case of
-// severe network freezes that a higher capacity will lead to a (transient)
-// increase in audio delay.
-struct NetEqCapacityConfig {
- NetEqCapacityConfig() : enabled(false), capacity(0) {}
- explicit NetEqCapacityConfig(int value) : enabled(true), capacity(value) {}
- static const ConfigOptionID identifier = ConfigOptionID::kNetEqCapacityConfig;
- bool enabled;
- int capacity;
-};
-
-struct NetEqFastAccelerate {
- NetEqFastAccelerate() : enabled(false) {}
- explicit NetEqFastAccelerate(bool value) : enabled(value) {}
- static const ConfigOptionID identifier = ConfigOptionID::kNetEqFastAccelerate;
- bool enabled;
-};
-
-struct VoicePacing {
- VoicePacing() : enabled(false) {}
- explicit VoicePacing(bool value) : enabled(value) {}
- static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
- bool enabled;
-};
-
} // namespace webrtc
#endif // WEBRTC_CONFIG_H_
diff --git a/media/engine/fakewebrtcvoiceengine.h b/media/engine/fakewebrtcvoiceengine.h
index 9b5440c..33ea7ca 100644
--- a/media/engine/fakewebrtcvoiceengine.h
+++ b/media/engine/fakewebrtcvoiceengine.h
@@ -154,7 +154,7 @@
int associate_send_channel = -1;
std::vector<webrtc::CodecInst> recv_codecs;
webrtc::CodecInst send_codec;
- int neteq_capacity = -1;
+ size_t neteq_capacity = 0;
bool neteq_fast_accelerate = false;
};
@@ -190,21 +190,6 @@
void set_fail_create_channel(bool fail_create_channel) {
fail_create_channel_ = fail_create_channel;
}
- int AddChannel(const webrtc::Config& config) {
- if (fail_create_channel_) {
- return -1;
- }
- Channel* ch = new Channel();
- auto db = webrtc::acm2::RentACodec::Database();
- ch->recv_codecs.assign(db.begin(), db.end());
- if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
- ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
- }
- ch->neteq_fast_accelerate =
- config.Get<webrtc::NetEqFastAccelerate>().enabled;
- channels_[++last_channel_] = ch;
- return last_channel_;
- }
int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
@@ -237,11 +222,20 @@
return nullptr;
}
WEBRTC_FUNC(CreateChannel, ()) {
- webrtc::Config empty_config;
- return AddChannel(empty_config);
+ return CreateChannel(webrtc::VoEBase::ChannelConfig());
}
- WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
- return AddChannel(config);
+ WEBRTC_FUNC(CreateChannel, (const webrtc::VoEBase::ChannelConfig& config)) {
+ if (fail_create_channel_) {
+ return -1;
+ }
+ Channel* ch = new Channel();
+ auto db = webrtc::acm2::RentACodec::Database();
+ ch->recv_codecs.assign(db.begin(), db.end());
+ ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer;
+ ch->neteq_fast_accelerate =
+ config.acm_config.neteq_config.enable_fast_accelerate;
+ channels_[++last_channel_] = ch;
+ return last_channel_;
}
WEBRTC_FUNC(DeleteChannel, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
@@ -547,7 +541,7 @@
void EnableStereoChannelSwapping(bool enable) override {
stereo_swapping_enabled_ = enable;
}
- int GetNetEqCapacity() const {
+ size_t GetNetEqCapacity() const {
auto ch = channels_.find(last_channel_);
ASSERT(ch != channels_.end());
return ch->second->neteq_capacity;
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 0169d5a..2992411 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -29,7 +29,6 @@
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/common.h"
#include "webrtc/media/base/audiosource.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/streamparams.h"
@@ -538,7 +537,7 @@
LOG(LS_INFO) << ToString(codec);
}
- voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
+ channel_config_.enable_voice_pacing = true;
// Temporarily turn logging level up for the Init() call.
webrtc::Trace::SetTraceCallback(this);
@@ -802,17 +801,14 @@
if (options.audio_jitter_buffer_max_packets) {
LOG(LS_INFO) << "NetEq capacity is "
<< *options.audio_jitter_buffer_max_packets;
- voe_config_.Set<webrtc::NetEqCapacityConfig>(
- new webrtc::NetEqCapacityConfig(
- *options.audio_jitter_buffer_max_packets));
+ channel_config_.acm_config.neteq_config.max_packets_in_buffer =
+ std::max(20, *options.audio_jitter_buffer_max_packets);
}
-
if (options.audio_jitter_buffer_fast_accelerate) {
LOG(LS_INFO) << "NetEq fast mode? "
<< *options.audio_jitter_buffer_fast_accelerate;
- voe_config_.Set<webrtc::NetEqFastAccelerate>(
- new webrtc::NetEqFastAccelerate(
- *options.audio_jitter_buffer_fast_accelerate));
+ channel_config_.acm_config.neteq_config.enable_fast_accelerate =
+ *options.audio_jitter_buffer_fast_accelerate;
}
if (options.typing_detection) {
@@ -1076,7 +1072,7 @@
int WebRtcVoiceEngine::CreateVoEChannel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- return voe_wrapper_->base()->CreateChannel(voe_config_);
+ return voe_wrapper_->base()->CreateChannel(channel_config_);
}
webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h
index 6508e23..27ae593 100644
--- a/media/engine/webrtcvoiceengine.h
+++ b/media/engine/webrtcvoiceengine.h
@@ -24,7 +24,6 @@
#include "webrtc/base/stream.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/call.h"
-#include "webrtc/common.h"
#include "webrtc/config.h"
#include "webrtc/media/base/rtputils.h"
#include "webrtc/media/engine/webrtccommon.h"
@@ -137,7 +136,7 @@
std::vector<AudioCodec> send_codecs_;
std::vector<AudioCodec> recv_codecs_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
- webrtc::Config voe_config_;
+ webrtc::VoEBase::ChannelConfig channel_config_;
bool is_dumping_aec_ = false;
webrtc::AgcConfig default_agc_config_;
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 81241dc..2cf764f 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -322,6 +322,7 @@
"audio_processing/beamformer/matrix_unittest.cc",
"audio_processing/beamformer/mock_nonlinear_beamformer.h",
"audio_processing/beamformer/nonlinear_beamformer_unittest.cc",
+ "audio_processing/config_unittest.cc",
"audio_processing/echo_cancellation_impl_unittest.cc",
"audio_processing/splitting_filter_unittest.cc",
"audio_processing/transient/dyadic_decimator_unittest.cc",
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index 833398a..c308308 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -22,7 +22,6 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/base/timeutils.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
@@ -48,7 +47,7 @@
}
}
-APITest::APITest(const Config& config)
+APITest::APITest()
: _acmA(AudioCodingModule::Create(1)),
_acmB(AudioCodingModule::Create(2)),
_channel_A2B(NULL),
diff --git a/modules/audio_coding/test/APITest.h b/modules/audio_coding/test/APITest.h
index af2a3a1..99a7201 100644
--- a/modules/audio_coding/test/APITest.h
+++ b/modules/audio_coding/test/APITest.h
@@ -23,8 +23,6 @@
namespace webrtc {
-class Config;
-
enum APITESTAction {
TEST_CHANGE_CODEC_ONLY = 0,
DTX_TEST = 1
@@ -32,7 +30,7 @@
class APITest : public ACMTest {
public:
- explicit APITest(const Config& config);
+ APITest();
~APITest();
void Perform();
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 891471d..f6764ac 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -13,7 +13,6 @@
#include <memory>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index 6d6f380..7df139b 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -20,8 +20,6 @@
namespace webrtc {
-class Config;
-
class TestPack : public AudioPacketizationCallback {
public:
TestPack();
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 24cda11..dbb238f 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -12,7 +12,6 @@
#include <assert.h>
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
diff --git a/modules/audio_coding/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h
index e936f75..09d9259 100644
--- a/modules/audio_coding/test/TestRedFec.h
+++ b/modules/audio_coding/test/TestRedFec.h
@@ -20,8 +20,6 @@
namespace webrtc {
-class Config;
-
class TestRedFec : public ACMTest {
public:
explicit TestRedFec();
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index 50702f9..04428d5 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -16,7 +16,6 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
diff --git a/modules/audio_coding/test/utility.cc b/modules/audio_coding/test/utility.cc
index 89368bc..58b5a68 100644
--- a/modules/audio_coding/test/utility.cc
+++ b/modules/audio_coding/test/utility.cc
@@ -16,7 +16,6 @@
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
diff --git a/modules/audio_coding/test/utility.h b/modules/audio_coding/test/utility.h
index dbd398e..b6c3a7a 100644
--- a/modules/audio_coding/test/utility.h
+++ b/modules/audio_coding/test/utility.h
@@ -128,10 +128,6 @@
uint32_t _numFrameTypes[5];
};
-void UseLegacyAcm(webrtc::Config* config);
-
-void UseNewAcm(webrtc::Config* config);
-
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 71338ea..2387c2e 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -71,6 +71,8 @@
"high_pass_filter_impl.cc",
"high_pass_filter_impl.h",
"include/audio_processing.h",
+ "include/config.cc",
+ "include/config.h",
"level_controller/biquad_filter.cc",
"level_controller/biquad_filter.h",
"level_controller/down_sampler.cc",
diff --git a/modules/audio_processing/audio_processing.gypi b/modules/audio_processing/audio_processing.gypi
index cbd4fdf..78f133a 100644
--- a/modules/audio_processing/audio_processing.gypi
+++ b/modules/audio_processing/audio_processing.gypi
@@ -83,6 +83,8 @@
'high_pass_filter_impl.cc',
'high_pass_filter_impl.h',
'include/audio_processing.h',
+ 'include/config.cc',
+ 'include/config.h',
'level_controller/biquad_filter.cc',
'level_controller/biquad_filter.h',
'level_controller/down_sampler.cc',
diff --git a/test/common_unittest.cc b/modules/audio_processing/config_unittest.cc
similarity index 96%
rename from test/common_unittest.cc
rename to modules/audio_processing/config_unittest.cc
index a239dad..2dd9f0d 100644
--- a/test/common_unittest.cc
+++ b/modules/audio_processing/config_unittest.cc
@@ -7,7 +7,7 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common.h"
+#include "webrtc/modules/audio_processing/include/config.h"
#include "testing/gtest/include/gtest/gtest.h"
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 09e5d5b..44ff732 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -21,8 +21,8 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/platform_file.h"
-#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/beamformer/array_util.h"
+#include "webrtc/modules/audio_processing/include/config.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/common.cc b/modules/audio_processing/include/config.cc
similarity index 90%
rename from common.cc
rename to modules/audio_processing/include/config.cc
index bc24818..bbbc452 100644
--- a/common.cc
+++ b/modules/audio_processing/include/config.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common.h"
+#include "webrtc/modules/audio_processing/include/config.h"
namespace webrtc {
diff --git a/common.h b/modules/audio_processing/include/config.h
similarity index 90%
rename from common.h
rename to modules/audio_processing/include/config.h
index 79db50c..615e3fa 100644
--- a/common.h
+++ b/modules/audio_processing/include/config.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_COMMON_H_
-#define WEBRTC_COMMON_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_CONFIG_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_CONFIG_H_
#include <map>
@@ -23,10 +23,10 @@
enum class ConfigOptionID {
kMyExperimentForTest,
kAlgo1CostFunctionForTest,
- kTemporalLayersFactory,
- kNetEqCapacityConfig,
- kNetEqFastAccelerate,
- kVoicePacing,
+ kTemporalLayersFactory, // Deprecated
+ kNetEqCapacityConfig, // Deprecated
+ kNetEqFastAccelerate, // Deprecated
+ kVoicePacing, // Deprecated
kExtendedFilter,
kDelayAgnostic,
kExperimentalAgc,
@@ -128,7 +128,6 @@
delete it;
it = new Option<T>(value);
}
-
} // namespace webrtc
-#endif // WEBRTC_COMMON_H_
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_CONFIG_H_
diff --git a/modules/audio_processing/test/process_test.cc b/modules/audio_processing/test/process_test.cc
index 6d5b979..ad769ad 100644
--- a/modules/audio_processing/test/process_test.cc
+++ b/modules/audio_processing/test/process_test.cc
@@ -20,8 +20,8 @@
#include "webrtc/base/format_macros.h"
#include "webrtc/base/timeutils.h"
-#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/include/config.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/include/module_common_types.h"
diff --git a/modules/modules.gyp b/modules/modules.gyp
index a8da0b7..a345ba4 100644
--- a/modules/modules.gyp
+++ b/modules/modules.gyp
@@ -245,6 +245,7 @@
'audio_processing/beamformer/matrix_unittest.cc',
'audio_processing/beamformer/mock_nonlinear_beamformer.h',
'audio_processing/beamformer/nonlinear_beamformer_unittest.cc',
+ 'audio_processing/config_unittest.cc',
'audio_processing/echo_cancellation_impl_unittest.cc',
'audio_processing/splitting_filter_unittest.cc',
'audio_processing/transient/dyadic_decimator_unittest.cc',
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 6b6a2bb..6991449 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -282,7 +282,6 @@
"channel_transport/udp_socket_manager_unittest.cc",
"channel_transport/udp_socket_wrapper_unittest.cc",
"channel_transport/udp_transport_unittest.cc",
- "common_unittest.cc",
"fake_network_pipe_unittest.cc",
"frame_generator_unittest.cc",
"rtp_file_reader_unittest.cc",
diff --git a/test/call_test.cc b/test/call_test.cc
index a766f79..e81d947 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/base/checks.h"
-#include "webrtc/common.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/test/call_test.h"
@@ -312,9 +311,9 @@
voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
decoder_factory_));
- Config voe_config;
- voe_config.Set<VoicePacing>(new VoicePacing(true));
- voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
+ VoEBase::ChannelConfig config;
+ config.enable_voice_pacing = true;
+ voe_send_.channel_id = voe_send_.base->CreateChannel(config);
EXPECT_GE(voe_send_.channel_id, 0);
voe_recv_.voice_engine = VoiceEngine::Create();
diff --git a/test/mock_voice_engine.h b/test/mock_voice_engine.h
index 469028d..abd493b 100644
--- a/test/mock_voice_engine.h
+++ b/test/mock_voice_engine.h
@@ -30,8 +30,7 @@
// http://crbug.com/428099.
MockVoiceEngine(
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr)
- : VoiceEngineImpl(new Config(), true),
- decoder_factory_(decoder_factory) {
+ : decoder_factory_(decoder_factory) {
// Increase ref count so this object isn't automatically deleted whenever
// interfaces are Release():d.
++_ref_count;
@@ -123,7 +122,7 @@
MOCK_METHOD0(audio_processing, AudioProcessing*());
MOCK_METHOD0(Terminate, int());
MOCK_METHOD0(CreateChannel, int());
- MOCK_METHOD1(CreateChannel, int(const Config& config));
+ MOCK_METHOD1(CreateChannel, int(const ChannelConfig& config));
MOCK_METHOD1(DeleteChannel, int(int channel));
MOCK_METHOD1(StartReceive, int(int channel));
MOCK_METHOD1(StopReceive, int(int channel));
diff --git a/test/test.gyp b/test/test.gyp
index fc9a2ad..7f64b77 100644
--- a/test/test.gyp
+++ b/test/test.gyp
@@ -210,7 +210,6 @@
'<(DEPTH)/testing/gtest.gyp:gtest',
],
'sources': [
- 'common_unittest.cc',
'fake_network_pipe_unittest.cc',
'frame_generator_unittest.cc',
'rtp_file_reader_unittest.cc',
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index baccc71..aa1c9b1 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -70,9 +70,9 @@
voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
voe->codec = webrtc::VoECodec::GetInterface(voe->voice_engine);
EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory));
- webrtc::Config voe_config;
- voe_config.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
- voe->send_channel_id = voe->base->CreateChannel(voe_config);
+ webrtc::VoEBase::ChannelConfig config;
+ config.enable_voice_pacing = true;
+ voe->send_channel_id = voe->base->CreateChannel(config);
EXPECT_GE(voe->send_channel_id, 0);
voe->receive_channel_id = voe->base->CreateChannel();
EXPECT_GE(voe->receive_channel_id, 0);
diff --git a/video/video_stream_decoder.h b/video/video_stream_decoder.h
index 465c8c4..ad35557 100644
--- a/video/video_stream_decoder.h
+++ b/video/video_stream_decoder.h
@@ -28,7 +28,6 @@
class CallStatsObserver;
class ChannelStatsObserver;
-class Config;
class EncodedImageCallback;
class I420FrameCallback;
class ReceiveStatisticsProxy;
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 29ee301..301b5a6 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -21,7 +21,6 @@
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/call/rtc_event_log.h"
-#include "webrtc/common.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ -34,7 +33,6 @@
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_external_media.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/output_mixer.h"
@@ -754,13 +752,12 @@
Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- const Config& config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ const VoEBase::ChannelConfig& config) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
instanceId);
- channel = new Channel(channelId, instanceId, config, decoder_factory);
+ channel = new Channel(channelId, instanceId, config);
if (channel == NULL) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel() unable to allocate memory for"
@@ -819,8 +816,7 @@
Channel::Channel(int32_t channelId,
uint32_t instanceId,
- const Config& config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
+ const VoEBase::ChannelConfig& config)
: _instanceId(instanceId),
_channelId(channelId),
event_log_proxy_(new RtcEventLogProxy()),
@@ -886,27 +882,18 @@
rtcp_observer_(new VoERtcpObserver(this)),
network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
associate_send_channel_(ChannelOwner(nullptr)),
- pacing_enabled_(config.Get<VoicePacing>().enabled),
+ pacing_enabled_(config.enable_voice_pacing),
feedback_observer_proxy_(new TransportFeedbackProxy()),
seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
kMaxRetransmissionWindowMs)),
- decoder_factory_(decoder_factory) {
+ decoder_factory_(config.acm_config.decoder_factory) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Channel() - ctor");
- AudioCodingModule::Config acm_config;
+ AudioCodingModule::Config acm_config(config.acm_config);
acm_config.id = VoEModuleId(instanceId, channelId);
- if (config.Get<NetEqCapacityConfig>().enabled) {
- // Clamping the buffer capacity at 20 packets. While going lower will
- // probably work, it makes little sense.
- acm_config.neteq_config.max_packets_in_buffer =
- std::max(20, config.Get<NetEqCapacityConfig>().capacity);
- }
- acm_config.neteq_config.enable_fast_accelerate =
- config.Get<NetEqFastAccelerate>().enabled;
acm_config.neteq_config.enable_muted_state = true;
- acm_config.decoder_factory = decoder_factory;
audio_coding_.reset(AudioCodingModule::Create(acm_config));
_outputAudioLevel.Clear();
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 1cf49d9..307b61e 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -29,6 +29,7 @@
#include "webrtc/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
+#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/network_predictor.h"
@@ -42,7 +43,6 @@
namespace webrtc {
class AudioDeviceModule;
-class Config;
class FileWrapper;
class PacketRouter;
class ProcessThread;
@@ -175,12 +175,10 @@
Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- const Config& config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
+ const VoEBase::ChannelConfig& config);
Channel(int32_t channelId,
uint32_t instanceId,
- const Config& config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
+ const VoEBase::ChannelConfig& config);
int32_t Init();
int32_t SetEngineInformation(Statistics& engineStatistics,
OutputMixer& outputMixer,
diff --git a/voice_engine/channel_manager.cc b/voice_engine/channel_manager.cc
index bcc22b0..b8ffe6e 100644
--- a/voice_engine/channel_manager.cc
+++ b/voice_engine/channel_manager.cc
@@ -10,7 +10,6 @@
#include "webrtc/voice_engine/channel_manager.h"
-#include "webrtc/common.h"
#include "webrtc/voice_engine/channel.h"
namespace webrtc {
@@ -45,26 +44,13 @@
ChannelOwner::ChannelRef::ChannelRef(class Channel* channel)
: channel(channel), ref_count(1) {}
-ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
- : instance_id_(instance_id), last_channel_id_(-1), config_(config) {}
+ChannelManager::ChannelManager(uint32_t instance_id)
+ : instance_id_(instance_id), last_channel_id_(-1) {}
ChannelOwner ChannelManager::CreateChannel(
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
- return CreateChannelInternal(config_, decoder_factory);
-}
-
-ChannelOwner ChannelManager::CreateChannel(
- const Config& external_config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
- return CreateChannelInternal(external_config, decoder_factory);
-}
-
-ChannelOwner ChannelManager::CreateChannelInternal(
- const Config& config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ const VoEBase::ChannelConfig& config) {
Channel* channel;
- Channel::CreateChannel(channel, ++last_channel_id_, instance_id_, config,
- decoder_factory);
+ Channel::CreateChannel(channel, ++last_channel_id_, instance_id_, config);
ChannelOwner channel_owner(channel);
rtc::CritScope crit(&lock_);
diff --git a/voice_engine/channel_manager.h b/voice_engine/channel_manager.h
index adc9d90..ae16aca 100644
--- a/voice_engine/channel_manager.h
+++ b/voice_engine/channel_manager.h
@@ -19,10 +19,10 @@
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/typedefs.h"
+#include "webrtc/voice_engine/include/voe_base.h"
namespace webrtc {
-class Config;
class AudioDecoderFactory;
namespace voe {
@@ -72,7 +72,7 @@
class ChannelManager {
public:
- ChannelManager(uint32_t instance_id, const Config& config);
+ ChannelManager(uint32_t instance_id);
// Upon construction of an Iterator it will grab a copy of the channel list of
// the ChannelManager. The iteration will then occur over this state, not the
@@ -95,16 +95,8 @@
RTC_DISALLOW_COPY_AND_ASSIGN(Iterator);
};
- // CreateChannel will always return a valid ChannelOwner instance. The channel
- // is created either based on internal configuration, i.e. |config_|, by
- // calling CreateChannel(...), or using and external configuration
- // |external_config| if the overloaded method
- // CreateChannel(const Config& external_config, ...) is called.
- ChannelOwner CreateChannel(
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
- ChannelOwner CreateChannel(
- const Config& external_config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
+ // CreateChannel will always return a valid ChannelOwner instance.
+ ChannelOwner CreateChannel(const VoEBase::ChannelConfig& config);
// ChannelOwner.channel() will be NULL if channel_id is invalid or no longer
// exists. This should be checked with ChannelOwner::IsValid().
@@ -117,11 +109,6 @@
size_t NumOfChannels() const;
private:
- // Create a channel given a configuration, |config|.
- ChannelOwner CreateChannelInternal(
- const Config& config,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
-
uint32_t instance_id_;
Atomic32 last_channel_id_;
@@ -129,8 +116,6 @@
rtc::CriticalSection lock_;
std::vector<ChannelOwner> channels_;
- const Config& config_;
-
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelManager);
};
} // namespace voe
diff --git a/voice_engine/include/voe_base.h b/voice_engine/include/voe_base.h
index 91df7d3..751394e 100644
--- a/voice_engine/include/voe_base.h
+++ b/voice_engine/include/voe_base.h
@@ -36,6 +36,7 @@
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/common_types.h"
namespace webrtc {
@@ -43,9 +44,6 @@
class AudioDeviceModule;
class AudioProcessing;
class AudioTransport;
-class Config;
-
-const int kVoEDefault = -1;
// VoiceEngineObserver
class WEBRTC_DLLEXPORT VoiceEngineObserver {
@@ -65,7 +63,6 @@
// Creates a VoiceEngine object, which can then be used to acquire
// sub-APIs. Returns NULL on failure.
static VoiceEngine* Create();
- static VoiceEngine* Create(const Config& config);
// Deletes a created VoiceEngine object and releases the utilized resources.
// Note that if there are outstanding references held via other interfaces,
@@ -99,6 +96,11 @@
// VoEBase
class WEBRTC_DLLEXPORT VoEBase {
public:
+ struct ChannelConfig {
+ AudioCodingModule::Config acm_config;
+ bool enable_voice_pacing = false;
+ };
+
// Factory for the VoEBase sub-API. Increases an internal reference
// counter if successful. Returns NULL if the API is not supported or if
// construction fails.
@@ -146,11 +148,13 @@
virtual int Terminate() = 0;
// Creates a new channel and allocates the required resources for it.
- // One can use |config| to configure the channel. Currently that is used for
- // choosing between ACM1 and ACM2, when creating Audio Coding Module.
+ // The second version accepts a |config| struct which includes an Audio Coding
+ // Module config and an option to enable voice pacing. Note that the
+ // decoder_factory member of the ACM config will be ignored (the decoder
+ // factory set through Init() will always be used).
// Returns channel ID or -1 in case of an error.
virtual int CreateChannel() = 0;
- virtual int CreateChannel(const Config& config) = 0;
+ virtual int CreateChannel(const ChannelConfig& config) = 0;
// Deletes an existing channel and releases the utilized resources.
// Returns -1 in case of an error, 0 otherwise.
diff --git a/voice_engine/shared_data.cc b/voice_engine/shared_data.cc
index 7a67561..41e261b 100644
--- a/voice_engine/shared_data.cc
+++ b/voice_engine/shared_data.cc
@@ -22,9 +22,9 @@
static int32_t _gInstanceCounter = 0;
-SharedData::SharedData(const Config& config)
+SharedData::SharedData()
: _instanceId(++_gInstanceCounter),
- _channelManager(_gInstanceCounter, config),
+ _channelManager(_gInstanceCounter),
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
_moduleProcessThreadPtr(
diff --git a/voice_engine/shared_data.h b/voice_engine/shared_data.h
index ea84eff..0e57f9d 100644
--- a/voice_engine/shared_data.h
+++ b/voice_engine/shared_data.h
@@ -25,8 +25,6 @@
class ProcessThread;
namespace webrtc {
-class Config;
-
namespace voe {
class TransmitMixer;
@@ -77,11 +75,10 @@
AudioDeviceModule::AudioLayer _audioDeviceLayer;
- SharedData(const Config& config);
+ SharedData();
virtual ~SharedData();
};
} // namespace voe
-
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_SHARED_DATA_H
diff --git a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
index b9cd9e9..1b226ff 100644
--- a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
+++ b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
@@ -11,9 +11,6 @@
#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
-#include <assert.h>
-
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
@@ -65,7 +62,6 @@
webrtc::VoEHardware* voe_hardware_;
webrtc::VoEExternalMedia* voe_xmedia_;
webrtc::VoENetEqStats* voe_neteq_stats_;
- webrtc::Config config_;
};
#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index 9c917b5..f74bdb1 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -12,7 +12,6 @@
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
-#include "webrtc/common.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
@@ -355,25 +354,20 @@
}
int VoEBaseImpl::CreateChannel() {
- rtc::CritScope cs(shared_->crit_sec());
- if (!shared_->statistics().Initialized()) {
- shared_->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ChannelOwner channel_owner =
- shared_->channel_manager().CreateChannel(decoder_factory_);
- return InitializeChannel(&channel_owner);
+ return CreateChannel(ChannelConfig());
}
-int VoEBaseImpl::CreateChannel(const Config& config) {
+int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
rtc::CritScope cs(shared_->crit_sec());
if (!shared_->statistics().Initialized()) {
shared_->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
+
+ ChannelConfig config_copy(config);
+ config_copy.acm_config.decoder_factory = decoder_factory_;
voe::ChannelOwner channel_owner =
- shared_->channel_manager().CreateChannel(config, decoder_factory_);
+ shared_->channel_manager().CreateChannel(config_copy);
return InitializeChannel(&channel_owner);
}
diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h
index 93e1cf5..050e9bb 100644
--- a/voice_engine/voe_base_impl.h
+++ b/voice_engine/voe_base_impl.h
@@ -41,7 +41,7 @@
int Terminate() override;
int CreateChannel() override;
- int CreateChannel(const Config& config) override;
+ int CreateChannel(const ChannelConfig& config) override;
int DeleteChannel(int channel) override;
int StartReceive(int channel) override;
diff --git a/voice_engine/voice_engine_impl.cc b/voice_engine/voice_engine_impl.cc
index 919d0e8..5b92d27 100644
--- a/voice_engine/voice_engine_impl.cc
+++ b/voice_engine/voice_engine_impl.cc
@@ -29,8 +29,8 @@
// improvement here.
static int32_t gVoiceEngineInstanceCounter = 0;
-VoiceEngine* GetVoiceEngine(const Config* config, bool owns_config) {
- VoiceEngineImpl* self = new VoiceEngineImpl(config, owns_config);
+VoiceEngine* GetVoiceEngine() {
+ VoiceEngineImpl* self = new VoiceEngineImpl();
if (self != NULL) {
self->AddRef(); // First reference. Released in VoiceEngine::Delete.
gVoiceEngineInstanceCounter++;
@@ -72,12 +72,7 @@
}
VoiceEngine* VoiceEngine::Create() {
- Config* config = new Config();
- return GetVoiceEngine(config, true);
-}
-
-VoiceEngine* VoiceEngine::Create(const Config& config) {
- return GetVoiceEngine(&config, false);
+ return GetVoiceEngine();
}
int VoiceEngine::SetTraceFilter(unsigned int filter) {
diff --git a/voice_engine/voice_engine_impl.h b/voice_engine/voice_engine_impl.h
index aa58007..3237001 100644
--- a/voice_engine/voice_engine_impl.h
+++ b/voice_engine/voice_engine_impl.h
@@ -83,8 +83,8 @@
#endif
public VoEBaseImpl {
public:
- VoiceEngineImpl(const Config* config, bool owns_config)
- : SharedData(*config),
+ VoiceEngineImpl()
+ : SharedData(),
#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
VoEAudioProcessingImpl(this),
#endif
@@ -114,9 +114,7 @@
VoEVolumeControlImpl(this),
#endif
VoEBaseImpl(this),
- _ref_count(0),
- own_config_(owns_config ? config : NULL) {
- }
+ _ref_count(0) {}
~VoiceEngineImpl() override { assert(_ref_count.Value() == 0); }
int AddRef();
@@ -132,8 +130,6 @@
// manipulate the reference count. See: fake_voice_engine.h.
protected:
Atomic32 _ref_count;
- private:
- std::unique_ptr<const Config> own_config_;
};
} // namespace webrtc
