Format almost everything.
This CL was generated by running
git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format
Most of these changes are clang-format grouping and reordering includes
differently.
Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index 9d004c6..043092c 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/modules/audio_coding/acm2/acm_receive_test.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
#include <stddef.h> // for size_t
+
#include <memory>
#include <string>
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 5ac71dd..6de45e7 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -12,6 +12,7 @@
#include <stdlib.h>
#include <string.h>
+
#include <cstdint>
#include <vector>
@@ -95,8 +96,7 @@
format = neteq_->GetDecoderFormat(payload_type);
}
if (!format) {
- RTC_LOG_F(LS_ERROR) << "Payload-type "
- << payload_type
+ RTC_LOG_F(LS_ERROR) << "Payload-type " << payload_type
<< " is not registered.";
return -1;
}
@@ -218,8 +218,8 @@
return neteq_->TargetDelayMs();
}
-absl::optional<std::pair<int, SdpAudioFormat>>
- AcmReceiver::LastDecoder() const {
+absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
+ const {
rtc::CritScope lock(&crit_sect_);
if (!last_decoder_) {
return absl::nullopt;
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 1f449a3..8d62312 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
#include <stdint.h>
+
#include <map>
#include <memory>
#include <string>
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 780026d..5d40fc1 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -314,14 +314,13 @@
{1, {"PCMA", 8000, 1}},
{2, {"ISAC", 32000, 1}},
{3, {"L16", 32000, 1}}};
- const std::map<int, int> cng_payload_types = {{8000, 100},
- {16000, 101},
- {32000, 102}};
+ const std::map<int, int> cng_payload_types = {
+ {8000, 100}, {16000, 101}, {32000, 102}};
{
std::map<int, SdpAudioFormat> receive_codecs = codecs;
for (const auto& cng_type : cng_payload_types) {
- receive_codecs.emplace(
- std::make_pair(cng_type.second, SdpAudioFormat("CN", cng_type.first, 1)));
+ receive_codecs.emplace(std::make_pair(
+ cng_type.second, SdpAudioFormat("CN", cng_type.first, 1)));
}
receiver_->SetCodecs(receive_codecs);
}
@@ -333,7 +332,7 @@
packet_sent_ = false;
InsertOnePacketOfSilence(
SetEncoder(0, codecs.at(0), cng_payload_types)); // Enough to test
- // with one codec.
+ // with one codec.
ASSERT_TRUE(packet_sent_);
EXPECT_EQ(AudioFrameType::kAudioFrameCN, last_frame_type_);
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 741cef9..e500c78 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include <assert.h>
+
#include <algorithm>
#include <cstdint>
@@ -654,7 +655,7 @@
}
absl::optional<std::pair<int, SdpAudioFormat>>
- AudioCodingModuleImpl::ReceiveCodec() const {
+AudioCodingModuleImpl::ReceiveCodec() const {
rtc::CritScope lock(&acm_crit_sect_);
return receiver_.LastDecoder();
}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index f30deed..ffa2bdc 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -8,8 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/include/audio_coding_module.h"
+
#include <stdio.h>
#include <string.h>
+
#include <atomic>
#include <memory>
#include <vector>
@@ -28,7 +31,6 @@
#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
@@ -54,9 +56,9 @@
#include "test/mock_audio_encoder.h"
#include "test/testsupport/file_utils.h"
+using ::testing::_;
using ::testing::AtLeast;
using ::testing::Invoke;
-using ::testing::_;
namespace webrtc {
diff --git a/modules/audio_coding/acm2/call_statistics_unittest.cc b/modules/audio_coding/acm2/call_statistics_unittest.cc
index 528708f..d7ac953 100644
--- a/modules/audio_coding/acm2/call_statistics_unittest.cc
+++ b/modules/audio_coding/acm2/call_statistics_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_coding/acm2/call_statistics.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index ff6ac01..11f93e6 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include <stdint.h>
+
#include <utility>
#include <vector>
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index fc2358b..e7cd056 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
#include <stdio.h>
+
#include <memory>
#include "absl/types/optional.h"
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index 76531d0..9612996 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -8,13 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+
#include <utility>
#include <vector>
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
-#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index f077357..76f52ad 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+
#include "rtc_base/numerics/safe_conversions.h"
#include "test/field_trial.h"
#include "test/gtest.h"
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/modules/audio_coding/audio_network_adaptor/channel_controller.cc
index a1c30db..2f5af67 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+
#include <algorithm>
-#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
index bfa6f01..21504bc 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+
#include <memory>
-#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index ce47699..7fa4096 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+
#include <string>
#include <utility>
-#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
#include "rtc_base/fake_clock.h"
@@ -129,8 +130,7 @@
TEST(ControllerManagerTest, ControllersWithoutCharPointAtEndAndInDefaultOrder) {
auto states = CreateControllerManager();
- CheckControllersOrder(&states, 0,
- 0.0,
+ CheckControllersOrder(&states, 0, 0.0,
{kNumControllers - 2, kNumControllers - 1, -1, -1});
}
diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
index 805df0a..6daefa5 100644
--- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -33,9 +33,9 @@
#if WEBRTC_ENABLE_PROTOBUF
namespace {
+using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
using audio_network_adaptor::debug_dump::Event;
using audio_network_adaptor::debug_dump::NetworkMetrics;
-using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
RTC_CHECK(dump_file->is_open());
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
index cbfea95..48384c9 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+
#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
index 67bf9e5..567df6f 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+
#include <memory>
-#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
index 7925b89..7aa522e 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -8,7 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+
#include <math.h>
+
#include <algorithm>
#include <cstdlib>
#include <utility>
@@ -18,7 +21,6 @@
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/rtc_event_log.h"
-#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
index 5d5e5df..2c34453 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
@@ -8,11 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+
#include <memory>
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
-#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index d3f54ee..d95cbce 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -8,17 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+
#include <utility>
#include "common_audio/mocks/mock_smoothing_filter.h"
-#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
#include "test/gtest.h"
namespace webrtc {
+using ::testing::_;
using ::testing::NiceMock;
using ::testing::Return;
-using ::testing::_;
namespace {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
index 4438a23..c51d561 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
@@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
+
#include <random>
#include <utility>
-#include "modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
index e182247..0268ddc 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
#include <stddef.h>
+
#include <map>
#include <set>
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index 8d6d815..6709336 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+
#include <memory>
#include <utility>
-#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
#include "test/gtest.h"
namespace webrtc {
@@ -209,8 +210,7 @@
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20}, 20);
// Use a low uplink bandwidth and a low uplink packet loss fraction that would
// cause frame length to increase if receiver frame length included 60ms.
- UpdateNetworkMetrics(controller.get(),
- kFl20msTo60msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
@@ -219,8 +219,7 @@
TEST(FrameLengthControllerTest, Maintain20MsOnMediumUplinkBandwidth) {
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
- UpdateNetworkMetrics(controller.get(),
- kMediumBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
@@ -231,18 +230,15 @@
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
// Use a low uplink bandwidth that would cause frame length to increase if
// uplink packet loss fraction was low.
- UpdateNetworkMetrics(controller.get(),
- kFl20msTo60msBandwidthBps,
- kMediumPacketLossFraction,
- kOverheadBytesPerPacket);
+ UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+ kMediumPacketLossFraction, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
TEST(FrameLengthControllerTest, Maintain60MsWhenNo120msCriteriaIsSet) {
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60, 120}, 60);
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo120msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
@@ -307,13 +303,11 @@
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
{20, 60, 120}, 20);
// It takes two steps for frame length to go from 20ms to 120ms.
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo120msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo120msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 120);
@@ -322,13 +316,11 @@
TEST(FrameLengthControllerTest, Stall60MsIf120MsNotInReceiverFrameLengthRange) {
auto controller =
CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(), {20, 60}, 20);
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo120msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo120msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
@@ -337,38 +329,31 @@
TEST(FrameLengthControllerTest, CheckBehaviorOnChangingNetworkMetrics) {
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
{20, 60, 120}, 20);
- UpdateNetworkMetrics(controller.get(),
- kMediumBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
- UpdateNetworkMetrics(controller.get(),
- kFl20msTo60msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo120msBandwidthBps,
- kMediumPacketLossFraction,
- kOverheadBytesPerPacket);
+ UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+ kMediumPacketLossFraction, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo120msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 120);
- UpdateNetworkMetrics(controller.get(),
- kFl120msTo60msBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kFl120msTo60msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(),
- kMediumBandwidthBps,
+ UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
diff --git a/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc b/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc
index 0375e76..9984049 100644
--- a/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+
#include <memory>
-#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
#include "test/gtest.h"
// A threshold curve divides 2D space into three domains - below, on and above
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
index 9b36dfd..968c118 100644
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+
#include <memory>
-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
index a548be8..108b1c1 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -8,11 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+
#include <limits>
#include <memory>
#include <vector>
-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index 2ef3236..8a11834 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.h
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
#include <stddef.h>
+
#include <memory>
#include "api/audio_codecs/audio_encoder.h"
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 3ecefd4..6dda862 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -8,21 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+
#include <memory>
#include <vector>
#include "common_audio/vad/mock/mock_vad.h"
-#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
-using ::testing::Return;
using ::testing::_;
-using ::testing::SetArgPointee;
using ::testing::InSequence;
using ::testing::Invoke;
+using ::testing::Return;
+using ::testing::SetArgPointee;
namespace webrtc {
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.h b/modules/audio_coding/codecs/cng/webrtc_cng.h
index 6ff7529..563f676 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.h
+++ b/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
#include <stdint.h>
+
#include <cstddef>
#include "api/array_view.h"
diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 8fae71c..6185918 100644
--- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
+++ b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
index 4de55a0..f02ca7f 100644
--- a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
+++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#include <string.h>
+
#include <utility>
#include "modules/audio_coding/codecs/g722/g722_interface.h"
diff --git a/modules/audio_coding/codecs/isac/fix/include/isacfix.h b/modules/audio_coding/codecs/isac/fix/include/isacfix.h
index 8fcfebb..4c95bfd 100644
--- a/modules/audio_coding/codecs/isac/fix/include/isacfix.h
+++ b/modules/audio_coding/codecs/isac/fix/include/isacfix.h
@@ -15,7 +15,9 @@
#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
-typedef struct { void* dummy; } ISACFIX_MainStruct;
+typedef struct {
+ void* dummy;
+} ISACFIX_MainStruct;
#if defined(__cplusplus)
extern "C" {
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
index 554ec0c..82793f1 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
+
#include "system_wrappers/include/cpu_features_wrapper.h"
#include "test/gtest.h"
diff --git a/modules/audio_coding/codecs/isac/fix/source/structs.h b/modules/audio_coding/codecs/isac/fix/source/structs.h
index 59226ac..89375fb 100644
--- a/modules/audio_coding/codecs/isac/fix/source/structs.h
+++ b/modules/audio_coding/codecs/isac/fix/source/structs.h
@@ -167,12 +167,10 @@
uint32_t prevSendTime; /* Send time for previous packet, from RTP header */
uint32_t prevArrivalTime; /* Arrival time for previous packet (in ms using
timeGetTime()) */
- uint16_t
- prevRtpRate; /* rate of previous packet, derived from RTP timestamps (in
- bits/s) */
- uint32_t
- lastUpdate; /* Time since the last update of the Bottle Neck estimate (in
- samples) */
+ uint16_t prevRtpRate; /* rate of previous packet, derived from RTP timestamps
+ (in bits/s) */
+ uint32_t lastUpdate; /* Time since the last update of the Bottle Neck estimate
+ (in samples) */
uint32_t lastReduction; /* Time sinse the last reduction (in samples) */
int32_t countUpdates; /* How many times the estimate was update in the
beginning */
@@ -197,9 +195,8 @@
uint32_t sendBwAvg; /* The estimated bottle neck rate from here to there (in
bits/s) */
- int32_t
- sendMaxDelayAvg; /* The estimated mean absolute jitter value, as seen on
- the other siee (in ms) */
+ int32_t sendMaxDelayAvg; /* The estimated mean absolute jitter value, as seen
+ on the other siee (in ms) */
int16_t countRecPkts; /* number of packets received since last update */
int16_t highSpeedRec; /* flag for marking that a high speed network has been
@@ -308,8 +305,8 @@
packet */
int16_t payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec
packet */
- int16_t maxPayloadBytes; /* Maximum allowed number of bits for both 30 and 60
- msec packet */
+ int16_t maxPayloadBytes; /* Maximum allowed number of bits for both 30 and 60
+ msec packet */
int16_t maxRateInBytes; /* Maximum allowed rate in bytes per 30 msec packet */
int16_t enforceFrameSize; /* If set iSAC will never change packet size */
@@ -339,8 +336,8 @@
} ISACFIX_SubStruct;
typedef struct {
- int32_t lpcGains
- [12]; /* 6 lower-band & 6 upper-band we may need to double it for 60*/
+ int32_t lpcGains[12]; /* 6 lower-band & 6 upper-band we may need to double it
+ for 60*/
/* */
uint32_t W_upper; /* Upper boundary of interval W */
uint32_t streamval;
diff --git a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
index 87ae0e0..07bab05 100644
--- a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
+++ b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+
#include <limits>
-#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
index 6d9b013..c98b21d 100644
--- a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
+++ b/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
@@ -7,9 +7,10 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
+
#include <string>
-#include "modules/audio_coding/codecs/isac/main/include/isac.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
index 2d92dfa..56ff22c 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
@@ -18,9 +18,8 @@
#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
-#include "modules/audio_coding/codecs/isac/main/source/structs.h"
-
#include "modules/audio_coding/codecs/isac/main/source/settings.h"
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
#define KLT_STEPSIZE 1.00000000
#define KLT_NUM_AVG_GAIN 0
diff --git a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 188e105..934794d 100644
--- a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -16,6 +16,7 @@
#include <stdlib.h>
#include <string.h>
#include <time.h>
+
#include <iostream>
/* include API */
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
index 41b08f7..21da136 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <vector>
#include "absl/types/optional.h"
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
index 5e5e6d4..efc3f0d 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
+++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
#include <stddef.h>
+
#include <memory>
#include <vector>
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index 3a2bb71..c792722 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index ab6e7db..8ae9ee7 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -8,12 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+
#include <array>
#include <memory>
#include <utility>
#include "absl/memory/memory.h"
-#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "common_audio/mocks/mock_smoothing_filter.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc
index 287213c..7f13380 100644
--- a/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -15,9 +15,9 @@
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
+using std::get;
using std::string;
using std::tuple;
-using std::get;
using ::testing::TestWithParam;
namespace webrtc {
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index 8a5bb6a..f0f2ef0 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -101,9 +101,9 @@
} // namespace
using test::AudioLoop;
+using ::testing::Combine;
using ::testing::TestWithParam;
using ::testing::Values;
-using ::testing::Combine;
// Maximum number of bytes in output bitstream.
const size_t kMaxBytes = 2000;
diff --git a/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
index 5c44bc5..6dbc8ee 100644
--- a/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
+++ b/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <memory>
-
#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+#include <memory>
+
#include "common_audio/channel_buffer.h"
#include "test/gtest.h"
@@ -29,7 +29,7 @@
const size_t num_channels = input.num_channels();
const size_t total_frames = input.num_frames();
AudioRingBuffer buf(num_channels, buffer_frames);
- std::unique_ptr<float* []> slice(new float*[num_channels]);
+ std::unique_ptr<float*[]> slice(new float*[num_channels]);
size_t input_pos = 0;
size_t output_pos = 0;
diff --git a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
index bfdb2e6..9c8e789 100644
--- a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
+++ b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <memory>
-
#include "modules/audio_coding/codecs/opus/test/blocker.h"
+#include <memory>
+
#include "rtc_base/arraysize.h"
#include "test/gtest.h"
diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
index 0334104..f08c4a6 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc b/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
index 8f8bba5..ecf91b4 100644
--- a/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
+++ b/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include <stdint.h>
+
#include <initializer_list>
namespace webrtc {
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 124e811..a0db045 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include <string.h>
+
#include <utility>
#include <vector>
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index f05de19..448df27 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <memory>
#include "absl/types/optional.h"
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 0f5a811..22e9a7f 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -8,21 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+
#include <memory>
#include <vector>
-#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
-using ::testing::Return;
using ::testing::_;
-using ::testing::SetArgPointee;
using ::testing::InSequence;
using ::testing::Invoke;
using ::testing::MockFunction;
+using ::testing::Return;
+using ::testing::SetArgPointee;
namespace webrtc {
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 61fa541..d711cca 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -225,8 +225,8 @@
// the last received payload.
// An empty Optional if no payload has yet been received.
//
- virtual absl::optional<std::pair<int, SdpAudioFormat>>
- ReceiveCodec() const = 0;
+ virtual absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec()
+ const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t IncomingPacket()
diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h
index a21bf57..0bb0b28 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.h
+++ b/modules/audio_coding/neteq/audio_multi_vector.h
@@ -13,6 +13,7 @@
#include <stdint.h>
#include <string.h>
+
#include <vector>
#include "api/array_view.h"
diff --git a/modules/audio_coding/neteq/audio_vector.h b/modules/audio_coding/neteq/audio_vector.h
index c8279da..a257586 100644
--- a/modules/audio_coding/neteq/audio_vector.h
+++ b/modules/audio_coding/neteq/audio_vector.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
#include <string.h>
+
#include <cstdint>
#include <memory>
diff --git a/modules/audio_coding/neteq/background_noise.h b/modules/audio_coding/neteq/background_noise.h
index 5191179..631db0d 100644
--- a/modules/audio_coding/neteq/background_noise.h
+++ b/modules/audio_coding/neteq/background_noise.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
#include <string.h> // size_t
+
#include <memory>
#include "api/array_view.h"
diff --git a/modules/audio_coding/neteq/buffer_level_filter.cc b/modules/audio_coding/neteq/buffer_level_filter.cc
index 144da63..d238665 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.cc
+++ b/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include <stdint.h>
+
#include <algorithm>
#include "rtc_base/numerics/safe_conversions.h"
diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc
index cb2b74d..a21cdda 100644
--- a/modules/audio_coding/neteq/comfort_noise.cc
+++ b/modules/audio_coding/neteq/comfort_noise.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/neteq/comfort_noise.h"
#include <assert.h>
+
#include <cstdint>
#include <memory>
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index fc255e5..e4a32fb 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -12,6 +12,7 @@
#include <assert.h>
#include <stdio.h>
+
#include <string>
#include "absl/types/optional.h"
@@ -201,9 +202,9 @@
decoder_frame_length, sample_rate_, estimate_dtx_delay_);
if ((prev_mode == kModeExpand || prev_mode == kModeCodecPlc) &&
expand.MuteFactor(0) < 16384 / 2 &&
- current_span < static_cast<size_t>(delay_manager_->TargetLevel() *
- packet_length_samples_ *
- kPostponeDecodingLevel / 100)>> 8 &&
+ current_span<static_cast<size_t>(delay_manager_->TargetLevel() *
+ packet_length_samples_ *
+ kPostponeDecodingLevel / 100)>> 8 &&
!packet_buffer_.ContainsDtxOrCngPacket(decoder_database_)) {
return kExpand;
}
diff --git a/modules/audio_coding/neteq/decision_logic_unittest.cc b/modules/audio_coding/neteq/decision_logic_unittest.cc
index 5c3d489..82f86c0 100644
--- a/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -11,6 +11,7 @@
// Unit tests for DecisionLogic class and derived classes.
#include "modules/audio_coding/neteq/decision_logic.h"
+
#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/delay_manager.h"
diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index 2049569..e755e7b 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/neteq/decoder_database.h"
#include <stddef.h>
+
#include <cstdint>
#include <list>
#include <type_traits>
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index 3a74896..bdaa28c 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -13,6 +13,7 @@
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
+
#include <algorithm>
#include <numeric>
#include <string>
@@ -30,7 +31,7 @@
namespace {
-constexpr int kLimitProbability = 1020054733; // 19/20 in Q30.
+constexpr int kLimitProbability = 1020054733; // 19/20 in Q30.
constexpr int kMinBaseMinimumDelayMs = 0;
constexpr int kMaxBaseMinimumDelayMs = 10000;
constexpr int kIatFactor = 32745; // 0.9993 in Q15.
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index 3075bfb..adefea1 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -191,13 +191,13 @@
// detection and streaming mode (Q0).
// TODO(turajs) change the comment according to the implementation of
// minimum-delay.
- int target_level_; // Currently preferred buffer level in (fractions)
- // of packets (Q8), before adding any extra delay.
- int packet_len_ms_; // Length of audio in each incoming packet [ms].
- uint16_t last_seq_no_; // Sequence number for last received packet.
- uint32_t last_timestamp_; // Timestamp for the last received packet.
- int minimum_delay_ms_; // Externally set minimum delay.
- int maximum_delay_ms_; // Externally set maximum allowed delay.
+ int target_level_; // Currently preferred buffer level in (fractions)
+ // of packets (Q8), before adding any extra delay.
+ int packet_len_ms_; // Length of audio in each incoming packet [ms].
+ uint16_t last_seq_no_; // Sequence number for last received packet.
+ uint32_t last_timestamp_; // Timestamp for the last received packet.
+ int minimum_delay_ms_; // Externally set minimum delay.
+ int maximum_delay_ms_; // Externally set maximum allowed delay.
DelayPeakDetector& peak_detector_;
int last_pack_cng_or_dtmf_;
const bool frame_length_change_experiment_;
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index 1004261..a8e2b3d 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -39,8 +39,8 @@
constexpr int kForgetFactor = 32745;
} // namespace
-using ::testing::Return;
using ::testing::_;
+using ::testing::Return;
class DelayManagerTest : public ::testing::Test {
protected:
@@ -683,7 +683,7 @@
EXPECT_EQ(DelayManager::HistogramMode::RELATIVE_ARRIVAL_DELAY,
dm_->histogram_mode());
EXPECT_EQ(kDefaultHistogramQuantile,
- dm_->histogram_quantile()); // 0.95 in Q30.
+ dm_->histogram_quantile()); // 0.95 in Q30.
EXPECT_EQ(
kForgetFactor,
dm_->histogram()->base_forget_factor_for_testing()); // 0.9993 in Q15.
@@ -696,7 +696,7 @@
EXPECT_EQ(DelayManager::HistogramMode::INTER_ARRIVAL_TIME,
dm_->histogram_mode());
EXPECT_EQ(kDefaultHistogramQuantile,
- dm_->histogram_quantile()); // 0.95 in Q30.
+ dm_->histogram_quantile()); // 0.95 in Q30.
EXPECT_EQ(
kForgetFactor,
dm_->histogram()->base_forget_factor_for_testing()); // 0.9993 in Q15.
diff --git a/modules/audio_coding/neteq/delay_peak_detector.h b/modules/audio_coding/neteq/delay_peak_detector.h
index 8cd198d..15db189 100644
--- a/modules/audio_coding/neteq/delay_peak_detector.h
+++ b/modules/audio_coding/neteq/delay_peak_detector.h
@@ -13,6 +13,7 @@
#include <stdint.h>
#include <string.h>
+
#include <list>
#include <memory>
diff --git a/modules/audio_coding/neteq/dtmf_buffer.h b/modules/audio_coding/neteq/dtmf_buffer.h
index a994e3a..6bf75e1 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.h
+++ b/modules/audio_coding/neteq/dtmf_buffer.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <list>
#include "rtc_base/constructor_magic.h"
diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h
index 9fc11eb..45d78d0 100644
--- a/modules/audio_coding/neteq/expand.h
+++ b/modules/audio_coding/neteq/expand.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
#include <assert.h>
+
#include <memory>
#include "modules/audio_coding/neteq/audio_vector.h"
diff --git a/modules/audio_coding/neteq/expand_uma_logger.cc b/modules/audio_coding/neteq/expand_uma_logger.cc
index 01c2dab..5db6d21 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.cc
+++ b/modules/audio_coding/neteq/expand_uma_logger.cc
@@ -8,6 +8,7 @@
*/
#include "modules/audio_coding/neteq/expand_uma_logger.h"
+
#include "rtc_base/checks.h"
#include "system_wrappers/include/metrics.h"
diff --git a/modules/audio_coding/neteq/expand_uma_logger.h b/modules/audio_coding/neteq/expand_uma_logger.h
index 7cb11b1..1139bb6 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.h
+++ b/modules/audio_coding/neteq/expand_uma_logger.h
@@ -11,6 +11,7 @@
#define MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
#include <stdint.h>
+
#include <memory>
#include <string>
diff --git a/modules/audio_coding/neteq/histogram.cc b/modules/audio_coding/neteq/histogram.cc
index fc0801e..99ea9aa 100644
--- a/modules/audio_coding/neteq/histogram.cc
+++ b/modules/audio_coding/neteq/histogram.cc
@@ -8,12 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/neteq/histogram.h"
+
#include <algorithm>
#include <cstdlib>
#include <numeric>
#include "absl/types/optional.h"
-#include "modules/audio_coding/neteq/histogram.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
diff --git a/modules/audio_coding/neteq/histogram_unittest.cc b/modules/audio_coding/neteq/histogram_unittest.cc
index 6255a0c..4df8b48 100644
--- a/modules/audio_coding/neteq/histogram_unittest.cc
+++ b/modules/audio_coding/neteq/histogram_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/neteq/histogram.h"
+
#include <cmath>
-#include "modules/audio_coding/neteq/histogram.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
index 031195c..d76afa4 100644
--- a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
+++ b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
#include "modules/audio_coding/neteq/buffer_level_filter.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_decoder_database.h b/modules/audio_coding/neteq/mock/mock_decoder_database.h
index d1db213..d83dc7f 100644
--- a/modules/audio_coding/neteq/mock/mock_decoder_database.h
+++ b/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -14,7 +14,6 @@
#include <string>
#include "modules/audio_coding/neteq/decoder_database.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h b/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
index f7f0465..82706f8 100644
--- a/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
+++ b/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
#include "modules/audio_coding/neteq/delay_peak_detector.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h b/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
index 11b571f..d9fe5d4 100644
--- a/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
+++ b/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
#include "modules/audio_coding/neteq/dtmf_buffer.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h b/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
index be4b7b5..eea8bee 100644
--- a/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
+++ b/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_expand.h b/modules/audio_coding/neteq/mock/mock_expand.h
index aed0164..286325b 100644
--- a/modules/audio_coding/neteq/mock/mock_expand.h
+++ b/modules/audio_coding/neteq/mock/mock_expand.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
#include "modules/audio_coding/neteq/expand.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_histogram.h b/modules/audio_coding/neteq/mock/mock_histogram.h
index 09b1b89..91ae18f 100644
--- a/modules/audio_coding/neteq/mock/mock_histogram.h
+++ b/modules/audio_coding/neteq/mock/mock_histogram.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_
#include "modules/audio_coding/neteq/histogram.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
index b477b1a..7efeb15 100644
--- a/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
#include "modules/audio_coding/neteq/packet_buffer.h"
-
#include "test/gmock.h"
namespace webrtc {
@@ -23,10 +22,8 @@
: PacketBuffer(max_number_of_packets, tick_timer) {}
virtual ~MockPacketBuffer() { Die(); }
MOCK_METHOD0(Die, void());
- MOCK_METHOD0(Flush,
- void());
- MOCK_CONST_METHOD0(Empty,
- bool());
+ MOCK_METHOD0(Flush, void());
+ MOCK_CONST_METHOD0(Empty, bool());
int InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
return InsertPacketWrapped(&packet, stats);
}
@@ -41,12 +38,10 @@
absl::optional<uint8_t>* current_rtp_payload_type,
absl::optional<uint8_t>* current_cng_rtp_payload_type,
StatisticsCalculator* stats));
- MOCK_CONST_METHOD1(NextTimestamp,
- int(uint32_t* next_timestamp));
+ MOCK_CONST_METHOD1(NextTimestamp, int(uint32_t* next_timestamp));
MOCK_CONST_METHOD2(NextHigherTimestamp,
- int(uint32_t timestamp, uint32_t* next_timestamp));
- MOCK_CONST_METHOD0(PeekNextPacket,
- const Packet*());
+ int(uint32_t timestamp, uint32_t* next_timestamp));
+ MOCK_CONST_METHOD0(PeekNextPacket, const Packet*());
MOCK_METHOD0(GetNextPacket, absl::optional<Packet>());
MOCK_METHOD1(DiscardNextPacket, int(StatisticsCalculator* stats));
MOCK_METHOD3(DiscardOldPackets,
@@ -55,12 +50,9 @@
StatisticsCalculator* stats));
MOCK_METHOD2(DiscardAllOldPackets,
void(uint32_t timestamp_limit, StatisticsCalculator* stats));
- MOCK_CONST_METHOD0(NumPacketsInBuffer,
- size_t());
- MOCK_METHOD1(IncrementWaitingTimes,
- void(int));
- MOCK_CONST_METHOD0(current_memory_bytes,
- int());
+ MOCK_CONST_METHOD0(NumPacketsInBuffer, size_t());
+ MOCK_METHOD1(IncrementWaitingTimes, void(int));
+ MOCK_CONST_METHOD0(current_memory_bytes, int());
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h b/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
index 426c467..68fd356 100644
--- a/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
+++ b/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
#include "modules/audio_coding/neteq/red_payload_splitter.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
index aedb1df..086c7c5 100644
--- a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
+++ b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
#include "modules/audio_coding/neteq/statistics_calculator.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/nack_tracker.cc b/modules/audio_coding/neteq/nack_tracker.cc
index e3ecfea..8358769 100644
--- a/modules/audio_coding/neteq/nack_tracker.cc
+++ b/modules/audio_coding/neteq/nack_tracker.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/neteq/nack_tracker.h"
#include <assert.h>
+
#include <cstdint>
#include <utility>
diff --git a/modules/audio_coding/neteq/nack_tracker.h b/modules/audio_coding/neteq/nack_tracker.h
index d7c6b08..5a56734 100644
--- a/modules/audio_coding/neteq/nack_tracker.h
+++ b/modules/audio_coding/neteq/nack_tracker.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <map>
#include <vector>
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 62184b0..fc47d91 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/neteq/neteq_impl.h"
#include <assert.h>
+
#include <algorithm>
#include <cstdint>
#include <cstring>
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index d529c9e..9e1af10 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -64,13 +64,7 @@
class NetEqImpl : public webrtc::NetEq {
public:
- enum class OutputType {
- kNormalSpeech,
- kPLC,
- kCNG,
- kPLCCNG,
- kVadPassive
- };
+ enum class OutputType { kNormalSpeech, kPLC, kCNG, kPLCCNG, kVadPassive };
enum ErrorCodes {
kNoError = 0,
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 0c7c090..ded54bf 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/neteq/neteq_impl.h"
+
#include <memory>
#include <utility>
#include <vector>
@@ -26,7 +28,6 @@
#include "modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
#include "modules/audio_coding/neteq/mock/mock_packet_buffer.h"
#include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
-#include "modules/audio_coding/neteq/neteq_impl.h"
#include "modules/audio_coding/neteq/preemptive_expand.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
@@ -198,7 +199,7 @@
UseNoMocks();
CreateInstance();
// Event: 2, E bit, Volume: 17, Length: 4336.
- uint8_t payload[kPayloadLength] = { 0x02, 0x80 + 0x11, 0x10, 0xF0 };
+ uint8_t payload[kPayloadLength] = {0x02, 0x80 + 0x11, 0x10, 0xF0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
@@ -228,13 +229,14 @@
EXPECT_THAT(output.packet_infos_, IsEmpty());
// Verify first 64 samples of actual output.
- const std::vector<int16_t> kOutput({
- 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, -1578, -2816, -3460, -3403, -2709, -1594,
- -363, 671, 1269, 1328, 908, 202, -513, -964, -955, -431, 504, 1617,
- 2602, 3164, 3101, 2364, 1073, -511, -2047, -3198, -3721, -3525, -2688,
- -1440, -99, 1015, 1663, 1744, 1319, 588, -171, -680, -747, -315, 515,
- 1512, 2378, 2828, 2674, 1877, 568, -986, -2446, -3482, -3864, -3516,
- -2534, -1163 });
+ const std::vector<int16_t> kOutput(
+ {0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ -1578, -2816, -3460, -3403, -2709, -1594, -363, 671, 1269, 1328,
+ 908, 202, -513, -964, -955, -431, 504, 1617, 2602, 3164,
+ 3101, 2364, 1073, -511, -2047, -3198, -3721, -3525, -2688, -1440,
+ -99, 1015, 1663, 1744, 1319, 588, -171, -680, -747, -315,
+ 515, 1512, 2378, 2828, 2674, 1877, 568, -986, -2446, -3482,
+ -3864, -3516, -2534, -1163});
ASSERT_GE(kMaxOutputSize, kOutput.size());
EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data()));
}
@@ -269,7 +271,6 @@
bool use_mock_payload_splitter_ = true;
};
-
// This tests the interface class NetEq.
// TODO(hlundin): Move to separate file?
TEST(NetEq, CreateAndDestroy) {
@@ -358,8 +359,7 @@
// Expectations for packet buffer.
EXPECT_CALL(*mock_packet_buffer_, Empty())
.WillOnce(Return(false)); // Called once after first packet is inserted.
- EXPECT_CALL(*mock_packet_buffer_, Flush())
- .Times(1);
+ EXPECT_CALL(*mock_packet_buffer_, Flush()).Times(1);
EXPECT_CALL(*mock_packet_buffer_, InsertPacketList(_, _, _, _, _))
.Times(2)
.WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType),
@@ -373,8 +373,7 @@
.WillOnce(Return(&fake_packet));
// Expectations for DTMF buffer.
- EXPECT_CALL(*mock_dtmf_buffer_, Flush())
- .Times(1);
+ EXPECT_CALL(*mock_dtmf_buffer_, Flush()).Times(1);
// Expectations for delay manager.
{
@@ -384,8 +383,7 @@
EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf())
.Times(2)
.WillRepeatedly(Return(-1));
- EXPECT_CALL(*mock_delay_manager_, set_last_pack_cng_or_dtmf(0))
- .Times(1);
+ EXPECT_CALL(*mock_delay_manager_, set_last_pack_cng_or_dtmf(0)).Times(1);
EXPECT_CALL(*mock_delay_manager_, ResetPacketIatCount()).Times(1);
// Expectations when the second packet is inserted. Slightly different.
EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf())
@@ -409,7 +407,7 @@
const int kPayloadLengthSamples = 80;
const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit.
- const uint8_t kPayloadType = 17; // Just an arbitrary number.
+ const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
@@ -896,18 +894,18 @@
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
AudioFrame output;
AudioFrame::SpeechType expected_type[8] = {
- AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech,
- AudioFrame::kCNG, AudioFrame::kCNG,
- AudioFrame::kCNG, AudioFrame::kCNG,
- AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech
- };
+ AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech, AudioFrame::kCNG,
+ AudioFrame::kCNG, AudioFrame::kCNG, AudioFrame::kCNG,
+ AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech};
int expected_timestamp_increment[8] = {
-1, // will not be used.
10 * kSampleRateKhz,
- -1, -1, // timestamp will be empty during CNG mode; indicated by -1 here.
- -1, -1,
- 50 * kSampleRateKhz, 10 * kSampleRateKhz
- };
+ -1,
+ -1, // timestamp will be empty during CNG mode; indicated by -1 here.
+ -1,
+ -1,
+ 50 * kSampleRateKhz,
+ 10 * kSampleRateKhz};
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
@@ -1008,11 +1006,9 @@
.Times(AtLeast(1))
.WillRepeatedly(Return(rtc::checked_cast<int>(kNetEqMaxFrameSize)));
- EXPECT_CALL(decoder, SampleRateHz())
- .WillRepeatedly(Return(kSampleRateHz));
+ EXPECT_CALL(decoder, SampleRateHz()).WillRepeatedly(Return(kSampleRateHz));
- EXPECT_CALL(decoder, Channels())
- .WillRepeatedly(Return(kChannels));
+ EXPECT_CALL(decoder, Channels()).WillRepeatedly(Return(kChannels));
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("L16", 8000, 1)));
@@ -1190,10 +1186,8 @@
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, _))
.WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
- EXPECT_CALL(mock_decoder, ErrorCode())
- .WillOnce(Return(kDecoderErrorCode));
- EXPECT_CALL(mock_decoder, HasDecodePlc())
- .WillOnce(Return(false));
+ EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));
+ EXPECT_CALL(mock_decoder, HasDecodePlc()).WillOnce(Return(false));
int16_t dummy_output[kFrameLengthSamples] = {0};
{
@@ -1308,8 +1302,7 @@
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, _))
.WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
- EXPECT_CALL(mock_decoder, ErrorCode())
- .WillOnce(Return(kDecoderErrorCode));
+ EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));
int16_t dummy_output[kFrameLengthSamples] = {0};
{
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index e05a790..20e5a5a 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -13,7 +13,6 @@
#include "absl/memory/memory.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
-
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/ref_counted_object.h"
@@ -25,8 +24,8 @@
namespace test {
using ::testing::_;
-using ::testing::SetArgPointee;
using ::testing::Return;
+using ::testing::SetArgPointee;
class MockAudioDecoder final : public AudioDecoder {
public:
diff --git a/modules/audio_coding/neteq/packet.h b/modules/audio_coding/neteq/packet.h
index 4f50e4d..238e769 100644
--- a/modules/audio_coding/neteq/packet.h
+++ b/modules/audio_coding/neteq/packet.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_NETEQ_PACKET_H_
#include <stdint.h>
+
#include <list>
#include <memory>
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index ca42222..688ce8d 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -11,6 +11,7 @@
// Unit tests for PacketBuffer class.
#include "modules/audio_coding/neteq/packet_buffer.h"
+
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
@@ -20,11 +21,11 @@
#include "test/gmock.h"
#include "test/gtest.h"
-using ::testing::Return;
-using ::testing::StrictMock;
using ::testing::_;
using ::testing::InSequence;
using ::testing::MockFunction;
+using ::testing::Return;
+using ::testing::StrictMock;
namespace {
class MockEncodedAudioFrame : public webrtc::AudioDecoder::EncodedAudioFrame {
@@ -54,12 +55,16 @@
int frame_size_;
};
-PacketGenerator::PacketGenerator(uint16_t seq_no, uint32_t ts, uint8_t pt,
+PacketGenerator::PacketGenerator(uint16_t seq_no,
+ uint32_t ts,
+ uint8_t pt,
int frame_size) {
Reset(seq_no, ts, pt, frame_size);
}
-void PacketGenerator::Reset(uint16_t seq_no, uint32_t ts, uint8_t pt,
+void PacketGenerator::Reset(uint16_t seq_no,
+ uint32_t ts,
+ uint8_t pt,
int frame_size) {
seq_no_ = seq_no;
ts_ = ts;
@@ -211,7 +216,7 @@
¤t_cng_pt, &mock_stats));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
- EXPECT_EQ(0, current_pt); // Current payload type changed to 0.
+ EXPECT_EQ(0, current_pt); // Current payload type changed to 0.
EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type not changed.
buffer.Flush(); // Clean up.
@@ -260,7 +265,7 @@
¤t_cng_pt, &mock_stats));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); // Only the last packet.
- EXPECT_EQ(1, current_pt); // Current payload type changed to 1.
+ EXPECT_EQ(1, current_pt); // Current payload type changed to 1.
EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type not changed.
buffer.Flush(); // Clean up.
@@ -276,24 +281,15 @@
const int kPayloadLength = 10;
PacketsToInsert packet_facts[kPackets] = {
- {0xFFFD, 0xFFFFFFD7, 0, true, 0},
- {0xFFFE, 0xFFFFFFE1, 0, true, 1},
- {0xFFFE, 0xFFFFFFD7, 1, false, -1},
- {0xFFFF, 0xFFFFFFEB, 0, true, 2},
- {0xFFFF, 0xFFFFFFE1, 1, false, -1},
- {0x0000, 0xFFFFFFF5, 0, true, 3},
- {0x0000, 0xFFFFFFEB, 1, false, -1},
- {0x0001, 0xFFFFFFFF, 0, true, 4},
- {0x0001, 0xFFFFFFF5, 1, false, -1},
- {0x0002, 0x0000000A, 0, true, 5},
- {0x0002, 0xFFFFFFFF, 1, false, -1},
- {0x0003, 0x0000000A, 1, false, -1},
- {0x0004, 0x0000001E, 0, true, 7},
- {0x0004, 0x00000014, 1, false, 6},
- {0x0005, 0x0000001E, 0, true, -1},
- {0x0005, 0x00000014, 1, false, -1},
- {0x0006, 0x00000028, 0, true, 8},
- {0x0006, 0x0000001E, 1, false, -1},
+ {0xFFFD, 0xFFFFFFD7, 0, true, 0}, {0xFFFE, 0xFFFFFFE1, 0, true, 1},
+ {0xFFFE, 0xFFFFFFD7, 1, false, -1}, {0xFFFF, 0xFFFFFFEB, 0, true, 2},
+ {0xFFFF, 0xFFFFFFE1, 1, false, -1}, {0x0000, 0xFFFFFFF5, 0, true, 3},
+ {0x0000, 0xFFFFFFEB, 1, false, -1}, {0x0001, 0xFFFFFFFF, 0, true, 4},
+ {0x0001, 0xFFFFFFF5, 1, false, -1}, {0x0002, 0x0000000A, 0, true, 5},
+ {0x0002, 0xFFFFFFFF, 1, false, -1}, {0x0003, 0x0000000A, 1, false, -1},
+ {0x0004, 0x0000001E, 0, true, 7}, {0x0004, 0x00000014, 1, false, 6},
+ {0x0005, 0x0000001E, 0, true, -1}, {0x0005, 0x00000014, 1, false, -1},
+ {0x0006, 0x00000028, 0, true, 8}, {0x0006, 0x0000001E, 1, false, -1},
};
const size_t kExpectPacketsInBuffer = 9;
@@ -310,10 +306,8 @@
InSequence s;
MockFunction<void(int check_point_id)> check;
for (int i = 0; i < kPackets; ++i) {
- gen.Reset(packet_facts[i].sequence_number,
- packet_facts[i].timestamp,
- packet_facts[i].payload_type,
- kFrameSize);
+ gen.Reset(packet_facts[i].sequence_number, packet_facts[i].timestamp,
+ packet_facts[i].payload_type, kFrameSize);
Packet packet = gen.NextPacket(kPayloadLength, nullptr);
packet.priority.codec_level = packet_facts[i].primary ? 0 : 1;
if (packet_facts[i].extract_order < 0) {
@@ -495,7 +489,7 @@
ASSERT_TRUE(buffer.PeekNextPacket());
EXPECT_EQ(kCngPt, buffer.PeekNextPacket()->payload_type);
EXPECT_EQ(current_pt, absl::nullopt); // Current payload type not set.
- EXPECT_EQ(kCngPt, current_cng_pt); // CNG payload type set.
+ EXPECT_EQ(kCngPt, current_cng_pt); // CNG payload type set.
// Insert second packet, which is wide-band speech.
{
@@ -513,7 +507,7 @@
ASSERT_TRUE(buffer.PeekNextPacket());
EXPECT_EQ(kSpeechPt, buffer.PeekNextPacket()->payload_type);
- EXPECT_EQ(kSpeechPt, current_pt); // Current payload type set.
+ EXPECT_EQ(kSpeechPt, current_pt); // Current payload type set.
EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type reset.
buffer.Flush(); // Clean up.
@@ -752,11 +746,11 @@
EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
limit_timestamp, limit_timestamp, kZeroHorizon));
// 1 sample behind is old.
- EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp - 1, limit_timestamp, kZeroHorizon));
+ EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 1,
+ limit_timestamp, kZeroHorizon));
// 2^31 - 1 samples behind is old.
- EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp - k2Pow31Minus1, limit_timestamp, kZeroHorizon));
+ EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - k2Pow31Minus1,
+ limit_timestamp, kZeroHorizon));
// 1 sample ahead is not old.
EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
limit_timestamp + 1, limit_timestamp, kZeroHorizon));
@@ -772,26 +766,26 @@
// Fixed horizon at 10 samples.
static const uint32_t kHorizon = 10;
// Timestamp on the limit is not old.
- EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp, limit_timestamp, kHorizon));
+ EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp,
+ limit_timestamp, kHorizon));
// 1 sample behind is old.
- EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp - 1, limit_timestamp, kHorizon));
+ EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 1,
+ limit_timestamp, kHorizon));
// 9 samples behind is old.
- EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp - 9, limit_timestamp, kHorizon));
+ EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 9,
+ limit_timestamp, kHorizon));
// 10 samples behind is not old.
- EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp - 10, limit_timestamp, kHorizon));
+ EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 10,
+ limit_timestamp, kHorizon));
// 2^31 - 1 samples behind is not old.
EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
limit_timestamp - k2Pow31Minus1, limit_timestamp, kHorizon));
// 1 sample ahead is not old.
- EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp + 1, limit_timestamp, kHorizon));
+ EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp + 1,
+ limit_timestamp, kHorizon));
// 2^31 samples ahead is not old.
- EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
- limit_timestamp + (1 << 31), limit_timestamp, kHorizon));
+ EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp + (1 << 31),
+ limit_timestamp, kHorizon));
}
} // namespace
diff --git a/modules/audio_coding/neteq/red_payload_splitter.cc b/modules/audio_coding/neteq/red_payload_splitter.cc
index 2a9befa..7ff5679 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -12,6 +12,7 @@
#include <assert.h>
#include <stddef.h>
+
#include <cstdint>
#include <list>
#include <utility>
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 4ef239a..d0fd26e 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -12,6 +12,7 @@
#include <assert.h>
#include <string.h> // memset
+
#include <algorithm>
#include "modules/audio_coding/neteq/delay_manager.h"
diff --git a/modules/audio_coding/neteq/sync_buffer.cc b/modules/audio_coding/neteq/sync_buffer.cc
index fee18cc..4949bb2 100644
--- a/modules/audio_coding/neteq/sync_buffer.cc
+++ b/modules/audio_coding/neteq/sync_buffer.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/neteq/sync_buffer.h"
+
#include <algorithm> // Access to min.
-#include "modules/audio_coding/neteq/sync_buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/sync_buffer.h b/modules/audio_coding/neteq/sync_buffer.h
index 7f6c111..754716b 100644
--- a/modules/audio_coding/neteq/sync_buffer.h
+++ b/modules/audio_coding/neteq/sync_buffer.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <vector>
#include "api/audio/audio_frame.h"
diff --git a/modules/audio_coding/neteq/sync_buffer_unittest.cc b/modules/audio_coding/neteq/sync_buffer_unittest.cc
index 29c3bca..860dbae 100644
--- a/modules/audio_coding/neteq/sync_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/sync_buffer_unittest.cc
@@ -9,8 +9,8 @@
*/
#include "modules/audio_coding/neteq/sync_buffer.h"
-#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
index a43c26a..9ec9d44 100644
--- a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
@@ -11,7 +11,6 @@
#include <memory>
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
-
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
diff --git a/modules/audio_coding/neteq/tick_timer.h b/modules/audio_coding/neteq/tick_timer.h
index 724dd12..2504ce3 100644
--- a/modules/audio_coding/neteq/tick_timer.h
+++ b/modules/audio_coding/neteq/tick_timer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
#include <stdint.h>
+
#include <memory>
#include "rtc_base/checks.h"
diff --git a/modules/audio_coding/neteq/tick_timer_unittest.cc b/modules/audio_coding/neteq/tick_timer_unittest.cc
index 875f04d..c501329 100644
--- a/modules/audio_coding/neteq/tick_timer_unittest.cc
+++ b/modules/audio_coding/neteq/tick_timer_unittest.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <memory>
-
#include "modules/audio_coding/neteq/tick_timer.h"
+#include <memory>
+
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/modules/audio_coding/neteq/time_stretch_unittest.cc b/modules/audio_coding/neteq/time_stretch_unittest.cc
index 6f875f0..da3a982 100644
--- a/modules/audio_coding/neteq/time_stretch_unittest.cc
+++ b/modules/audio_coding/neteq/time_stretch_unittest.cc
@@ -10,14 +10,13 @@
// Unit tests for Accelerate and PreemptiveExpand classes.
-#include "modules/audio_coding/neteq/accelerate.h"
-#include "modules/audio_coding/neteq/preemptive_expand.h"
-
#include <map>
#include <memory>
#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/accelerate.h"
#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"
@@ -64,8 +63,7 @@
sample_rate_hz_(32000),
block_size_(30 * sample_rate_hz_ / 1000), // 30 ms
audio_(new int16_t[block_size_]),
- background_noise_(kNumChannels) {
- }
+ background_noise_(kNumChannels) {}
const int16_t* Next30Ms() {
RTC_CHECK(input_file_->Read(block_size_, audio_.get()));
diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
index 5b14189..9ba63e3 100644
--- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
+++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -9,15 +9,16 @@
*/
#include "modules/audio_coding/neteq/timestamp_scaler.h"
+
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "modules/audio_coding/neteq/packet.h"
#include "test/gmock.h"
#include "test/gtest.h"
+using ::testing::_;
using ::testing::Return;
using ::testing::ReturnNull;
-using ::testing::_;
namespace webrtc {
diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
index 7adb15b..6a79ce4 100644
--- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
+++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
#include <stdio.h>
+
#include <string>
#include "modules/audio_coding/neteq/tools/packet_source.h"
diff --git a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
index bf016a1..52f7ea8 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
+++ b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
@@ -11,8 +11,8 @@
// Unit tests for test InputAudioFile class.
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index ad6aaa5..a990a81 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -8,10 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+
#include <stdio.h>
+
#include <cmath>
-#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
diff --git a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
index 6933fc0..3f06b1c 100644
--- a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
+++ b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
@@ -12,6 +12,7 @@
#include <inttypes.h>
#include <stdio.h>
+
#include <utility>
namespace webrtc {
diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
index 9b68c5c..a7061eb 100644
--- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -14,6 +14,7 @@
#include <limits.h> // For ULONG_MAX returned by strtoul.
#include <stdio.h>
#include <stdlib.h> // For strtoul.
+
#include <fstream>
#include <iostream>
#include <memory>
diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h
index c923a1e..d729c9c 100644
--- a/modules/audio_coding/neteq/tools/output_audio_file.h
+++ b/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -13,6 +13,7 @@
#include <assert.h>
#include <stdio.h>
+
#include <string>
#include "modules/audio_coding/neteq/tools/audio_sink.h"
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index f0cc0a3..f864aa1 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
#include <string.h>
+
#include <iostream>
#include <limits>
#include <set>
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.cc b/modules/audio_coding/neteq/tools/rtp_generator.cc
index ab7acdc..accd163 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.cc
+++ b/modules/audio_coding/neteq/tools/rtp_generator.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <assert.h>
-
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include <assert.h>
+
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/rtp_jitter.cc b/modules/audio_coding/neteq/tools/rtp_jitter.cc
index 3521145..cccaa9a 100644
--- a/modules/audio_coding/neteq/tools/rtp_jitter.cc
+++ b/modules/audio_coding/neteq/tools/rtp_jitter.cc
@@ -9,6 +9,7 @@
*/
#include <stdio.h>
+
#include <algorithm>
#include <fstream>
#include <iostream>
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 8ad0e00..e76bacb 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/test/Channel.h"
#include <assert.h>
+
#include <iostream>
#include "rtc_base/format_macros.h"
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 25e273a..20e415d 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -12,6 +12,7 @@
#include <stdio.h>
#include <stdlib.h>
+
#include <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -23,14 +24,10 @@
namespace webrtc {
-TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
- : _rtpStream(rtpStream),
- _frequency(frequency),
- _seqNo(0) {
-}
+TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency)
+ : _rtpStream(rtpStream), _frequency(frequency), _seqNo(0) {}
-TestPacketization::~TestPacketization() {
-}
+TestPacketization::~TestPacketization() {}
int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
const uint8_t payloadType,
@@ -43,15 +40,14 @@
}
Sender::Sender()
- : _acm(NULL),
- _pcmFile(),
- _audioFrame(),
- _packetization(NULL) {
-}
+ : _acm(NULL), _pcmFile(), _audioFrame(), _packetization(NULL) {}
-void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int in_sample_rate,
- int payload_type, SdpAudioFormat format) {
+void Sender::Setup(AudioCodingModule* acm,
+ RTPStream* rtpStream,
+ std::string in_file_name,
+ int in_sample_rate,
+ int payload_type,
+ SdpAudioFormat format) {
// Open input file
const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
_pcmFile.Open(file_name, in_sample_rate, "rb");
@@ -96,11 +92,13 @@
Receiver::Receiver()
: _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
- _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
-}
+ _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
-void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, size_t channels, int file_num) {
+void Receiver::Setup(AudioCodingModule* acm,
+ RTPStream* rtpStream,
+ std::string out_file_name,
+ size_t channels,
+ int file_num) {
EXPECT_EQ(0, acm->InitializeReceiver());
if (channels == 1) {
@@ -187,14 +185,14 @@
return false;
}
EXPECT_EQ(0, ok);
- if (ok < 0){
+ if (ok < 0) {
return false;
}
if (_playoutLengthSmpls == 0) {
return false;
}
- _pcmFile.Write10MsData(audioFrame.data(),
- audioFrame.samples_per_channel_ * audioFrame.num_channels_);
+ _pcmFile.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_ *
+ audioFrame.num_channels_);
return true;
}
@@ -225,17 +223,15 @@
EncodeDecodeTest::EncodeDecodeTest() = default;
void EncodeDecodeTest::Perform() {
- const std::map<int, SdpAudioFormat> send_codecs = {{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {0, {"PCMU", 8000, 1}},
- {8, {"PCMA", 8000, 1}},
+ const std::map<int, SdpAudioFormat> send_codecs = {
+ {103, {"ISAC", 16000, 1}}, {104, {"ISAC", 32000, 1}},
+ {107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}},
+ {8, {"PCMA", 8000, 1}},
#ifdef WEBRTC_CODEC_ILBC
- {102, {"ILBC", 8000, 1}},
+ {102, {"ILBC", 8000, 1}},
#endif
- {9, {"G722", 8000, 1}}};
+ {9, {"G722", 8000, 1}}};
int file_num = 0;
for (const auto& send_codec : send_codecs) {
RTPFile rtpFile;
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index ec95766..a3d1a26 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -26,7 +26,7 @@
// TestPacketization callback which writes the encoded payloads to file
class TestPacketization : public AudioPacketizationCallback {
public:
- TestPacketization(RTPStream *rtpStream, uint16_t frequency);
+ TestPacketization(RTPStream* rtpStream, uint16_t frequency);
~TestPacketization();
int32_t SendData(const AudioFrameType frameType,
const uint8_t payloadType,
@@ -35,8 +35,11 @@
const size_t payloadSize) override;
private:
- static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
- int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
+ static void MakeRTPheader(uint8_t* rtpHeader,
+ uint8_t payloadType,
+ int16_t seqNo,
+ uint32_t timeStamp,
+ uint32_t ssrc);
RTPStream* _rtpStream;
int32_t _frequency;
int16_t _seqNo;
@@ -45,9 +48,12 @@
class Sender {
public:
Sender();
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int in_sample_rate,
- int payload_type, SdpAudioFormat format);
+ void Setup(AudioCodingModule* acm,
+ RTPStream* rtpStream,
+ std::string in_file_name,
+ int in_sample_rate,
+ int payload_type,
+ SdpAudioFormat format);
void Teardown();
void Run();
bool Add10MsData();
@@ -65,8 +71,11 @@
public:
Receiver();
virtual ~Receiver() {}
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, size_t channels, int file_num);
+ void Setup(AudioCodingModule* acm,
+ RTPStream* rtpStream,
+ std::string out_file_name,
+ size_t channels,
+ int file_num);
void Teardown();
void Run();
virtual bool IncomingPacket();
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index cbe066f..727f692 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -147,7 +147,7 @@
rtpFile.WriteHeader();
SenderWithFEC sender;
sender.Setup(acm.get(), &rtpFile, in_file_name_, 120, send_format,
- expected_loss_rate_);
+ expected_loss_rate_);
sender.Run();
sender.Teardown();
rtpFile.Close();
@@ -156,7 +156,7 @@
rtpFile.ReadHeader();
ReceiverWithPacketLoss receiver;
receiver.Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 15,
- actual_loss_rate_, burst_length_);
+ actual_loss_rate_, burst_length_);
receiver.Run();
receiver.Teardown();
rtpFile.Close();
diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index b26f6ec..4c0dfd8 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
#include <string>
+
#include "modules/audio_coding/test/EncodeDecodeTest.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index 1273fa8..db4e0f3 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -11,6 +11,7 @@
#include "RTPFile.h"
#include <stdlib.h>
+
#include <limits>
#ifdef WIN32
diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index 1c555ed..eda576d 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_CODING_TEST_RTPFILE_H_
#include <stdio.h>
+
#include <queue>
#include "modules/audio_coding/include/audio_coding_module.h"
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index a3f0964..be4460e 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -112,8 +112,7 @@
channel_a_to_b_(NULL),
test_count_(0),
packet_size_samples_(0),
- packet_size_bytes_(0) {
-}
+ packet_size_bytes_(0) {}
TestAllCodecs::~TestAllCodecs() {
if (channel_a_to_b_ != NULL) {
@@ -360,13 +359,15 @@
my_acm = acm_b_.get();
break;
}
- default: { break; }
+ default: {
+ break;
+ }
}
ASSERT_TRUE(my_acm != NULL);
auto factory = CreateBuiltinAudioEncoderFactory();
constexpr int payload_type = 17;
- SdpAudioFormat format = { codec_name, clockrate_hz, num_channels };
+ SdpAudioFormat format = {codec_name, clockrate_hz, num_channels};
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
my_acm->SetEncoder(
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 0cb2415..5155958 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -61,8 +61,8 @@
}
void TestRedFec::Perform() {
- const std::string file_name = webrtc::test::ResourcePath(
- "audio_coding/testfile32kHz", "pcm");
+ const std::string file_name =
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
_inFileA.Open(file_name, 32000, "rb");
ASSERT_EQ(0, _acmA->InitializeReceiver());
@@ -183,9 +183,8 @@
config.payload_type = cn_payload_type;
config.vad_mode = vad_mode.value();
encoder = CreateComfortNoiseEncoder(std::move(config));
- receive_codecs.emplace(
- std::make_pair(cn_payload_type,
- SdpAudioFormat("CN", codec_format.clockrate_hz, 1)));
+ receive_codecs.emplace(std::make_pair(
+ cn_payload_type, SdpAudioFormat("CN", codec_format.clockrate_hz, 1)));
}
if (use_red) {
AudioEncoderCopyRed::Config config;
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index ea8735b..42bdbd8 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -104,8 +104,7 @@
test_cntr_(0),
pack_size_samp_(0),
pack_size_bytes_(0),
- counter_(0) {
-}
+ counter_(0) {}
TestStereo::~TestStereo() {
if (channel_a2b_ != NULL) {
diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 56e2c46..113dbe0 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -9,6 +9,7 @@
*/
#include <stdio.h>
+
#include <string>
#include <vector>
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index 585c1db..91dbfd6 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -59,7 +59,6 @@
const int payload_type1,
const SdpAudioFormat& format2,
const int payload_type2) {
-
//--- Set A codecs
_acmA->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index 7776a60..ae6c2b7 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -34,8 +34,8 @@
constexpr int kISAC16kPayloadType = 103;
constexpr int kISAC32kPayloadType = 104;
-const SdpAudioFormat kISAC16kFormat = { "ISAC", 16000, 1 };
-const SdpAudioFormat kISAC32kFormat = { "ISAC", 32000, 1 };
+const SdpAudioFormat kISAC16kFormat = {"ISAC", 16000, 1};
+const SdpAudioFormat kISAC32kFormat = {"ISAC", 32000, 1};
AudioEncoderIsacFloat::Config TweakConfig(
AudioEncoderIsacFloat::Config config,
@@ -107,9 +107,9 @@
}
void ISACTest::ACMTestTimer::CurrentTime(unsigned long& h,
- unsigned char& m,
- unsigned char& s,
- unsigned short& ms) {
+ unsigned char& m,
+ unsigned char& s,
+ unsigned short& ms) {
h = _hour;
m = _min;
s = _sec;
@@ -146,9 +146,9 @@
void ISACTest::Setup() {
// Register both iSAC-wb & iSAC-swb in both sides as receiver codecs.
- std::map<int, SdpAudioFormat> receive_codecs =
- {{kISAC16kPayloadType, kISAC16kFormat},
- {kISAC32kPayloadType, kISAC32kFormat}};
+ std::map<int, SdpAudioFormat> receive_codecs = {
+ {kISAC16kPayloadType, kISAC16kFormat},
+ {kISAC32kPayloadType, kISAC32kFormat}};
_acmA->SetReceiveCodecs(receive_codecs);
_acmB->SetReceiveCodecs(receive_codecs);
@@ -165,14 +165,12 @@
file_name_swb_ =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
_inFileA.Open(file_name_swb_, 32000, "rb");
// Set test length to 500 ms (50 blocks of 10 ms each).
@@ -254,16 +252,14 @@
_outFileB.Open(file_name_out, 32000, "wb");
// Side A is sending super-wideband, and side B is sending wideband.
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- swbISACConfig),
- kISAC32kPayloadType));
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- wbISACConfig),
- kISAC16kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ swbISACConfig),
+ kISAC32kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ wbISACConfig),
+ kISAC16kPayloadType));
bool adaptiveMode = false;
if ((swbISACConfig.currentRateBitPerSec == -1) ||
@@ -309,14 +305,12 @@
// Start with side A sending super-wideband and side B seding wideband.
// Toggle sending wideband/super-wideband in this test.
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
int numSendCodecChanged = 0;
_myTimer.Reset();
@@ -330,18 +324,16 @@
// Switch side A to send super-wideband.
_inFileA.Close();
_inFileA.Open(file_name_swb_, 32000, "rb");
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
} else {
// Switch side A to send wideband.
_inFileA.Close();
_inFileA.Open(file_name_swb_, 32000, "rb");
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
}
numSendCodecChanged++;
}
@@ -351,18 +343,16 @@
// Switch side B to send super-wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
} else {
// Switch side B to send wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
}
numSendCodecChanged++;
}
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 6f7c6cf..2fb59d1 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -33,8 +33,8 @@
ASSERT_EQ(0, acm_->InitializeReceiver());
constexpr int pltype = 108;
- std::map<int, SdpAudioFormat> receive_codecs =
- {{pltype, {"L16", kSampleRateHz, 1}}};
+ std::map<int, SdpAudioFormat> receive_codecs = {
+ {pltype, {"L16", kSampleRateHz, 1}}};
acm_->SetReceiveCodecs(receive_codecs);
rtp_header_.payloadType = pltype;