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Moving/renaming webrtc/common.h.

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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-Commit-Position: refs/heads/master@{#14109}
This commit is contained in:
solenberg 2016-09-07 07:34:41 -07:00 committed by Commit bot
parent 4016a0b2e8
commit 88499ecaca
44 changed files with 96 additions and 221 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/BUILD.gn
View File

@ -325,9 +325,6 @@ if (!build_with_chromium) {
rtc_source_set("webrtc_common") {
sources = [
"audio_sink.h",
"common.cc",
"common.h",
"common_types.cc",
"common_types.h",
"config.cc",

6
webrtc/call/call_perf_tests.cc
View File

@ -159,9 +159,9 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
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;

2
webrtc/common.gyp
View File

@ -12,8 +12,6 @@
'target_name': 'webrtc_common',
'type': 'static_library',
'sources': [
'common.cc',
'common.h',
'common_types.cc',
'common_types.h',
'config.h',

2
webrtc/common_types.h
View File

@ -52,8 +52,6 @@
namespace webrtc {
class Config;
class RewindableStream {
public:
virtual ~RewindableStream() {}

30
webrtc/config.h
View File

@ -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 @@ class DecoderSpecificSettings {
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_

36
webrtc/media/engine/fakewebrtcvoiceengine.h
View File

@ -154,7 +154,7 @@ class FakeWebRtcVoiceEngine
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 @@ class FakeWebRtcVoiceEngine
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 @@ class FakeWebRtcVoiceEngine
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 @@ class FakeWebRtcVoiceEngine
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;

16
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -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 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
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 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
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 @@ void WebRtcVoiceEngine::StopAecDump() {
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() {

3
webrtc/media/engine/webrtcvoiceengine.h
View File

@ -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 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
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_;

1
webrtc/modules/BUILD.gn
View File

@ -322,6 +322,7 @@ if (rtc_include_tests) {
"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",

3
webrtc/modules/audio_coding/test/APITest.cc
View File

@ -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 @@ void APITest::Wait(uint32_t waitLengthMs) {
}
}
APITest::APITest(const Config& config)
APITest::APITest()
: _acmA(AudioCodingModule::Create(1)),
_acmB(AudioCodingModule::Create(2)),
_channel_A2B(NULL),

4
webrtc/modules/audio_coding/test/APITest.h
View File

@ -23,8 +23,6 @@
namespace webrtc {
class Config;
enum APITESTAction {
TEST_CHANGE_CODEC_ONLY = 0,
DTX_TEST = 1
@ -32,7 +30,7 @@ enum APITESTAction {
class APITest : public ACMTest {
public:
explicit APITest(const Config& config);
APITest();
~APITest();
void Perform();

1
webrtc/modules/audio_coding/test/PacketLossTest.cc
View File

@ -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 {

2
webrtc/modules/audio_coding/test/TestAllCodecs.h
View File

@ -20,8 +20,6 @@
namespace webrtc {
class Config;
class TestPack : public AudioPacketizationCallback {
public:
TestPack();

1
webrtc/modules/audio_coding/test/TestRedFec.cc
View File

@ -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"

2
webrtc/modules/audio_coding/test/TestRedFec.h
View File

@ -20,8 +20,6 @@
namespace webrtc {
class Config;
class TestRedFec : public ACMTest {
public:
explicit TestRedFec();

1
webrtc/modules/audio_coding/test/delay_test.cc
View File

@ -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"

1
webrtc/modules/audio_coding/test/utility.cc
View File

@ -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"

4
webrtc/modules/audio_coding/test/utility.h
View File

@ -128,10 +128,6 @@ class VADCallback : public ACMVADCallback {
uint32_t _numFrameTypes[5];
};
void UseLegacyAcm(webrtc::Config* config);
void UseNewAcm(webrtc::Config* config);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_

2
webrtc/modules/audio_processing/BUILD.gn
View File

@ -71,6 +71,8 @@ rtc_source_set("audio_processing") {
"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",

2
webrtc/modules/audio_processing/audio_processing.gypi
View File

@ -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',

2
webrtc/test/common_unittest.cc → webrtc/modules/audio_processing/config_unittest.cc
View File

@ -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"

2
webrtc/modules/audio_processing/include/audio_processing.h
View File

@ -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 {

2
webrtc/common.cc → webrtc/modules/audio_processing/include/config.cc
View File

@ -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 {

15
webrtc/common.h → webrtc/modules/audio_processing/include/config.h
View File

@ -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 @@ namespace webrtc {
enum class ConfigOptionID {
kMyExperimentForTest,
kAlgo1CostFunctionForTest,
kTemporalLayersFactory,
kNetEqCapacityConfig,
kNetEqFastAccelerate,
kVoicePacing,
kTemporalLayersFactory, // Deprecated
kNetEqCapacityConfig, // Deprecated
kNetEqFastAccelerate, // Deprecated
kVoicePacing, // Deprecated
kExtendedFilter,
kDelayAgnostic,
kExperimentalAgc,
@ -128,7 +128,6 @@ void Config::Set(T* value) {
delete it;
it = new Option<T>(value);
}
} // namespace webrtc
#endif // WEBRTC_COMMON_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_CONFIG_H_

2
webrtc/modules/audio_processing/test/process_test.cc
View File

@ -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"

1
webrtc/modules/modules.gyp
View File

@ -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',

1
webrtc/test/BUILD.gn
View File

@ -282,7 +282,6 @@ rtc_test("test_support_unittests") {
"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",

7
webrtc/test/call_test.cc
View File

@ -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 @@ void CallTest::CreateVoiceEngines() {
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();

5
webrtc/test/mock_voice_engine.h
View File

@ -30,8 +30,7 @@ class MockVoiceEngine : public VoiceEngineImpl {
// 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 @@ class MockVoiceEngine : public VoiceEngineImpl {
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));

1
webrtc/test/test.gyp
View File

@ -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',

6
webrtc/video/video_quality_test.cc
View File

@ -70,9 +70,9 @@ void CreateVoiceEngine(VoiceEngineState* voe,
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);

1
webrtc/video/video_stream_decoder.h
View File

@ -28,7 +28,6 @@ namespace webrtc {
class CallStatsObserver;
class ChannelStatsObserver;
class Config;
class EncodedImageCallback;
class I420FrameCallback;
class ReceiveStatisticsProxy;

25
webrtc/voice_engine/channel.cc
View File

@ -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 @@ int32_t Channel::CreateChannel(
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 @@ void Channel::RecordFileEnded(int32_t id) {
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 @@ Channel::Channel(int32_t channelId,
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();

8
webrtc/voice_engine/channel.h
View File

@ -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 @@ class TimestampWrapAroundHandler;
namespace webrtc {
class AudioDeviceModule;
class Config;
class FileWrapper;
class PacketRouter;
class ProcessThread;
@ -175,12 +175,10 @@ class Channel
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,

22
webrtc/voice_engine/channel_manager.cc
View File

@ -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& ChannelOwner::operator=(const ChannelOwner& other) {
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_);

23
webrtc/voice_engine/channel_manager.h
View File

@ -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 ChannelOwner {
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 @@ class ChannelManager {
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 @@ class ChannelManager {
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 @@ class ChannelManager {
rtc::CriticalSection lock_;
std::vector<ChannelOwner> channels_;
const Config& config_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelManager);
};
} // namespace voe

18
webrtc/voice_engine/include/voe_base.h
View File

@ -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 @@ namespace webrtc {
class AudioDeviceModule;
class AudioProcessing;
class AudioTransport;
class Config;
const int kVoEDefault = -1;
// VoiceEngineObserver
class WEBRTC_DLLEXPORT VoiceEngineObserver {
@ -65,7 +63,6 @@ class WEBRTC_DLLEXPORT VoiceEngine {
// 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 @@ class WEBRTC_DLLEXPORT VoiceEngine {
// 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 @@ class WEBRTC_DLLEXPORT VoEBase {
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.

4
webrtc/voice_engine/shared_data.cc
View File

@ -22,9 +22,9 @@ namespace voe {
static int32_t _gInstanceCounter = 0;
SharedData::SharedData(const Config& config)
SharedData::SharedData()
: _instanceId(++_gInstanceCounter),
_channelManager(_gInstanceCounter, config),
_channelManager(_gInstanceCounter),
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
_moduleProcessThreadPtr(

5
webrtc/voice_engine/shared_data.h
View File

@ -25,8 +25,6 @@
class ProcessThread;
namespace webrtc {
class Config;
namespace voe {
class TransmitMixer;
@ -77,11 +75,10 @@ protected:
AudioDeviceModule::AudioLayer _audioDeviceLayer;
SharedData(const Config& config);
SharedData();
virtual ~SharedData();
};
} // namespace voe
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_SHARED_DATA_H

4
webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
View File

@ -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 @@ class BeforeInitializationFixture : public testing::Test {
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_

18
webrtc/voice_engine/voe_base_impl.cc
View File

@ -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::Terminate() {
}
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);
}

2
webrtc/voice_engine/voe_base_impl.h
View File

@ -41,7 +41,7 @@ class VoEBaseImpl : public VoEBase,
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;

11
webrtc/voice_engine/voice_engine_impl.cc
View File

@ -29,8 +29,8 @@ namespace webrtc {
// 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 @@ std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
}
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) {

10
webrtc/voice_engine/voice_engine_impl.h
View File

@ -83,8 +83,8 @@ class VoiceEngineImpl : public voe::SharedData, // Must be the first base class
#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 @@ class VoiceEngineImpl : public voe::SharedData, // Must be the first base class
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 @@ class VoiceEngineImpl : public voe::SharedData, // Must be the first base class
// manipulate the reference count. See: fake_voice_engine.h.
protected:
Atomic32 _ref_count;
private:
std::unique_ptr<const Config> own_config_;
};
} // namespace webrtc