Revert "Reland "Remove WEBRTC_TRACE.""

This reverts commit 68007e97ec9399125e4be9964af8b0338766cd91.

Reason for revert: More downstream breakages.

Original change's description:
> Reland "Remove WEBRTC_TRACE."
> 
> This is a reland of 2209b90449473e1df3e0797b6271c7624b41907d
> Original change's description:
> > Remove WEBRTC_TRACE.
> > 
> > Bug: webrtc:5118
> > Change-Id: I0af0f8845ee016fa61d7cecc526e2a672ec8732d
> > Reviewed-on: https://webrtc-review.googlesource.com/5382
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20114}
> 
> Bug: webrtc:5118
> Change-Id: I2d93fd40fcaa251c363bdcfb1c04b834a3a7f0e9
> Reviewed-on: https://webrtc-review.googlesource.com/6000
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20132}

TBR=solenberg@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

Change-Id: I093ee8c5c997c0dd46b3a3ca0e6271e3ea083d82
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5118
Reviewed-on: https://webrtc-review.googlesource.com/6320
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20133}
This commit is contained in:
Fredrik Solenberg
2017-10-04 08:49:37 +00:00
committed by Commit Bot
parent 68007e97ec
commit 39cefdb3c5
36 changed files with 1589 additions and 15 deletions

View File

@ -13,8 +13,10 @@
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_processing/agc/mock_agc.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "system_wrappers/include/trace.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/trace_to_stderr.h"
using ::testing::_;
using ::testing::DoAll;
@ -92,6 +94,7 @@ class AgcManagerDirectTest : public ::testing::Test {
test::MockGainControl gctrl_;
TestVolumeCallbacks volume_;
AgcManagerDirect manager_;
test::TraceToStderr trace_to_stderr;
};
TEST_F(AgcManagerDirectTest, StartupMinVolumeConfigurationIsRespected) {

View File

@ -38,6 +38,7 @@
#include "rtc_base/task_queue.h"
#include "rtc_base/thread.h"
#include "system_wrappers/include/event_wrapper.h"
#include "system_wrappers/include/trace.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@ -350,9 +351,11 @@ class ApmTest : public ::testing::Test {
virtual void TearDown();
static void SetUpTestCase() {
Trace::CreateTrace();
}
static void TearDownTestCase() {
Trace::ReturnTrace();
ClearTempFiles();
}

View File

@ -14,6 +14,7 @@
#include "modules/audio_processing/test/protobuf_utils.h"
#include "rtc_base/checks.h"
#include "test/testsupport/trace_to_stderr.h"
namespace webrtc {
namespace test {
@ -209,6 +210,11 @@ void AecDumpBasedSimulator::PrepareReverseProcessStreamCall(
}
void AecDumpBasedSimulator::Process() {
std::unique_ptr<test::TraceToStderr> trace_to_stderr;
if (settings_.use_verbose_logging) {
trace_to_stderr.reset(new test::TraceToStderr(true));
}
CreateAudioProcessor();
dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
@ -229,6 +235,8 @@ void AecDumpBasedSimulator::Process() {
webrtc::audioproc::Event event_msg;
int num_forward_chunks_processed = 0;
const float kOneBykChunksPerSecond =
1.f / AudioProcessingSimulator::kChunksPerSecond;
while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
switch (event_msg.type()) {
case webrtc::audioproc::Event::INIT:
@ -251,6 +259,10 @@ void AecDumpBasedSimulator::Process() {
default:
RTC_CHECK(false);
}
if (trace_to_stderr) {
trace_to_stderr->SetTimeSeconds(num_forward_chunks_processed *
kOneBykChunksPerSecond);
}
}
fclose(dump_input_file_);

View File

@ -15,6 +15,7 @@
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/checks.h"
#include "test/testsupport/trace_to_stderr.h"
namespace webrtc {
namespace test {
@ -87,6 +88,11 @@ void WavBasedSimulator::PrepareReverseProcessStreamCall() {
}
void WavBasedSimulator::Process() {
std::unique_ptr<test::TraceToStderr> trace_to_stderr;
if (settings_.use_verbose_logging) {
trace_to_stderr.reset(new test::TraceToStderr(true));
}
if (settings_.custom_call_order_filename) {
call_chain_ = WavBasedSimulator::GetCustomEventChain(
*settings_.custom_call_order_filename);
@ -100,6 +106,8 @@ void WavBasedSimulator::Process() {
bool samples_left_to_process = true;
int call_chain_index = 0;
int num_forward_chunks_processed = 0;
const int kOneBykChunksPerSecond =
1.f / AudioProcessingSimulator::kChunksPerSecond;
while (samples_left_to_process) {
switch (call_chain_[call_chain_index]) {
case SimulationEventType::kProcessStream:
@ -116,6 +124,11 @@ void WavBasedSimulator::Process() {
}
call_chain_index = (call_chain_index + 1) % call_chain_.size();
if (trace_to_stderr) {
trace_to_stderr->SetTimeSeconds(num_forward_chunks_processed *
kOneBykChunksPerSecond);
}
}
DestroyAudioProcessor();