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}
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@ -14,7 +14,6 @@
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#include "modules/audio_processing/test/protobuf_utils.h"
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#include "rtc_base/checks.h"
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#include "test/testsupport/trace_to_stderr.h"
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namespace webrtc {
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namespace test {
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@ -210,11 +209,6 @@ void AecDumpBasedSimulator::PrepareReverseProcessStreamCall(
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}
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void AecDumpBasedSimulator::Process() {
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std::unique_ptr<test::TraceToStderr> trace_to_stderr;
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if (settings_.use_verbose_logging) {
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trace_to_stderr.reset(new test::TraceToStderr(true));
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}
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CreateAudioProcessor();
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dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
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@ -235,8 +229,6 @@ void AecDumpBasedSimulator::Process() {
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webrtc::audioproc::Event event_msg;
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int num_forward_chunks_processed = 0;
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const float kOneBykChunksPerSecond =
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1.f / AudioProcessingSimulator::kChunksPerSecond;
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while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
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switch (event_msg.type()) {
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case webrtc::audioproc::Event::INIT:
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@ -259,10 +251,6 @@ void AecDumpBasedSimulator::Process() {
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default:
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RTC_CHECK(false);
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}
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if (trace_to_stderr) {
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trace_to_stderr->SetTimeSeconds(num_forward_chunks_processed *
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kOneBykChunksPerSecond);
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}
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}
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fclose(dump_input_file_);
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@ -15,7 +15,6 @@
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#include "modules/audio_processing/test/test_utils.h"
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#include "rtc_base/checks.h"
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#include "test/testsupport/trace_to_stderr.h"
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namespace webrtc {
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namespace test {
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@ -88,11 +87,6 @@ void WavBasedSimulator::PrepareReverseProcessStreamCall() {
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}
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void WavBasedSimulator::Process() {
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std::unique_ptr<test::TraceToStderr> trace_to_stderr;
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if (settings_.use_verbose_logging) {
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trace_to_stderr.reset(new test::TraceToStderr(true));
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}
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if (settings_.custom_call_order_filename) {
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call_chain_ = WavBasedSimulator::GetCustomEventChain(
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*settings_.custom_call_order_filename);
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@ -106,8 +100,6 @@ void WavBasedSimulator::Process() {
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bool samples_left_to_process = true;
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int call_chain_index = 0;
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int num_forward_chunks_processed = 0;
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const int kOneBykChunksPerSecond =
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1.f / AudioProcessingSimulator::kChunksPerSecond;
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while (samples_left_to_process) {
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switch (call_chain_[call_chain_index]) {
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case SimulationEventType::kProcessStream:
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@ -124,11 +116,6 @@ void WavBasedSimulator::Process() {
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}
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call_chain_index = (call_chain_index + 1) % call_chain_.size();
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if (trace_to_stderr) {
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trace_to_stderr->SetTimeSeconds(num_forward_chunks_processed *
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kOneBykChunksPerSecond);
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}
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}
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DestroyAudioProcessor();
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