Index: webrtc/modules/audio_processing/audio_processing_impl.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc |
index 2143fe18785d8cd1f0b313b2879780e8f29c9b42..a33294534307e1e593c1c605029eeb885d58f853 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -15,6 +15,7 @@ |
#include "webrtc/base/checks.h" |
#include "webrtc/base/platform_file.h" |
+#include "webrtc/base/trace_event.h" |
#include "webrtc/common_audio/audio_converter.h" |
#include "webrtc/common_audio/channel_buffer.h" |
#include "webrtc/common_audio/include/audio_util.h" |
@@ -547,6 +548,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src, |
int output_sample_rate_hz, |
ChannelLayout output_layout, |
float* const* dest) { |
+ TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_ChannelLayout"); |
StreamConfig input_stream; |
StreamConfig output_stream; |
{ |
@@ -575,6 +577,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src, |
const StreamConfig& input_config, |
const StreamConfig& output_config, |
float* const* dest) { |
+ TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig"); |
ProcessingConfig processing_config; |
{ |
// Acquire the capture lock in order to safely call the function |
@@ -637,6 +640,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src, |
} |
int AudioProcessingImpl::ProcessStream(AudioFrame* frame) { |
+ TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame"); |
{ |
// Acquire the capture lock in order to safely call the function |
// that retrieves the render side data. This function accesses apm |
@@ -816,6 +820,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data, |
size_t samples_per_channel, |
int rev_sample_rate_hz, |
ChannelLayout layout) { |
+ TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_ChannelLayout"); |
rtc::CritScope cs(&crit_render_); |
const StreamConfig reverse_config = { |
rev_sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout), |
@@ -831,6 +836,7 @@ int AudioProcessingImpl::ProcessReverseStream( |
const StreamConfig& reverse_input_config, |
const StreamConfig& reverse_output_config, |
float* const* dest) { |
+ TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig"); |
rtc::CritScope cs(&crit_render_); |
RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, reverse_input_config, |
reverse_output_config)); |
@@ -890,6 +896,7 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked( |
} |
int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) { |
+ TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame"); |
RETURN_ON_ERR(AnalyzeReverseStream(frame)); |
rtc::CritScope cs(&crit_render_); |
if (is_rev_processed()) { |
@@ -900,6 +907,7 @@ int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) { |
} |
int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) { |
+ TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_AudioFrame"); |
rtc::CritScope cs(&crit_render_); |
if (frame == nullptr) { |
return kNullPointerError; |