Introduced the new locking scheme
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1424663003
Cr-Commit-Position: refs/heads/master@{#10836}
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 0eacd28..c8175dc 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -14,7 +14,6 @@
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
@@ -44,10 +43,12 @@
} // namespace
GainControlImpl::GainControlImpl(const AudioProcessing* apm,
- CriticalSectionWrapper* crit)
+ rtc::CriticalSection* crit_render,
+ rtc::CriticalSection* crit_capture)
: ProcessingComponent(),
apm_(apm),
- crit_(crit),
+ crit_render_(crit_render),
+ crit_capture_(crit_capture),
mode_(kAdaptiveAnalog),
minimum_capture_level_(0),
maximum_capture_level_(255),
@@ -57,13 +58,18 @@
analog_capture_level_(0),
was_analog_level_set_(false),
stream_is_saturated_(false),
- render_queue_element_max_size_(0) {}
+ render_queue_element_max_size_(0) {
+ RTC_DCHECK(apm);
+ RTC_DCHECK(crit_render);
+ RTC_DCHECK(crit_capture);
+}
GainControlImpl::~GainControlImpl() {}
int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
+ rtc::CritScope cs(crit_render_);
if (!is_component_enabled()) {
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
assert(audio->num_frames_per_band() <= 160);
@@ -74,7 +80,7 @@
int err =
WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());
- if (err != apm_->kNoError)
+ if (err != AudioProcessing::kNoError)
return GetHandleError(my_handle);
// Buffer the samples in the render queue.
@@ -85,18 +91,21 @@
// Insert the samples into the queue.
if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+ // The data queue is full and needs to be emptied.
ReadQueuedRenderData();
// Retry the insert (should always work).
RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
}
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
// Read chunks of data that were received and queued on the render side from
// a queue. All the data chunks are buffered into the farend signal of the AGC.
void GainControlImpl::ReadQueuedRenderData() {
+ rtc::CritScope cs(crit_capture_);
+
if (!is_component_enabled()) {
return;
}
@@ -116,14 +125,16 @@
}
int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
+ rtc::CritScope cs(crit_capture_);
+
if (!is_component_enabled()) {
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
assert(audio->num_frames_per_band() <= 160);
assert(audio->num_channels() == num_handles());
- int err = apm_->kNoError;
+ int err = AudioProcessing::kNoError;
if (mode_ == kAdaptiveAnalog) {
capture_levels_.assign(num_handles(), analog_capture_level_);
@@ -135,7 +146,7 @@
audio->num_bands(),
audio->num_frames_per_band());
- if (err != apm_->kNoError) {
+ if (err != AudioProcessing::kNoError) {
return GetHandleError(my_handle);
}
}
@@ -155,23 +166,25 @@
capture_levels_[i] = capture_level_out;
- if (err != apm_->kNoError) {
+ if (err != AudioProcessing::kNoError) {
return GetHandleError(my_handle);
}
}
}
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+ rtc::CritScope cs(crit_capture_);
+
if (!is_component_enabled()) {
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
- return apm_->kStreamParameterNotSetError;
+ return AudioProcessing::kStreamParameterNotSetError;
}
assert(audio->num_frames_per_band() <= 160);
@@ -183,6 +196,8 @@
int32_t capture_level_out = 0;
uint8_t saturation_warning = 0;
+ // The call to stream_has_echo() is ok from a deadlock perspective
+ // as the capture lock is allready held.
int err = WebRtcAgc_Process(
my_handle,
audio->split_bands_const(i),
@@ -194,7 +209,7 @@
apm_->echo_cancellation()->stream_has_echo(),
&saturation_warning);
- if (err != apm_->kNoError) {
+ if (err != AudioProcessing::kNoError) {
return GetHandleError(my_handle);
}
@@ -215,22 +230,24 @@
}
was_analog_level_set_ = false;
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
// TODO(ajm): ensure this is called under kAdaptiveAnalog.
int GainControlImpl::set_stream_analog_level(int level) {
- CriticalSectionScoped crit_scoped(crit_);
+ rtc::CritScope cs(crit_capture_);
+
was_analog_level_set_ = true;
if (level < minimum_capture_level_ || level > maximum_capture_level_) {
- return apm_->kBadParameterError;
+ return AudioProcessing::kBadParameterError;
}
analog_capture_level_ = level;
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
int GainControlImpl::stream_analog_level() {
+ rtc::CritScope cs(crit_capture_);
// TODO(ajm): enable this assertion?
//assert(mode_ == kAdaptiveAnalog);
@@ -238,18 +255,21 @@
}
int GainControlImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(crit_);
+ rtc::CritScope cs_render(crit_render_);
+ rtc::CritScope cs_capture(crit_capture_);
return EnableComponent(enable);
}
bool GainControlImpl::is_enabled() const {
+ rtc::CritScope cs(crit_capture_);
return is_component_enabled();
}
int GainControlImpl::set_mode(Mode mode) {
- CriticalSectionScoped crit_scoped(crit_);
+ rtc::CritScope cs_render(crit_render_);
+ rtc::CritScope cs_capture(crit_capture_);
if (MapSetting(mode) == -1) {
- return apm_->kBadParameterError;
+ return AudioProcessing::kBadParameterError;
}
mode_ = mode;
@@ -257,22 +277,23 @@
}
GainControl::Mode GainControlImpl::mode() const {
+ rtc::CritScope cs(crit_capture_);
return mode_;
}
int GainControlImpl::set_analog_level_limits(int minimum,
int maximum) {
- CriticalSectionScoped crit_scoped(crit_);
+ rtc::CritScope cs(crit_capture_);
if (minimum < 0) {
- return apm_->kBadParameterError;
+ return AudioProcessing::kBadParameterError;
}
if (maximum > 65535) {
- return apm_->kBadParameterError;
+ return AudioProcessing::kBadParameterError;
}
if (maximum < minimum) {
- return apm_->kBadParameterError;
+ return AudioProcessing::kBadParameterError;
}
minimum_capture_level_ = minimum;
@@ -282,21 +303,24 @@
}
int GainControlImpl::analog_level_minimum() const {
+ rtc::CritScope cs(crit_capture_);
return minimum_capture_level_;
}
int GainControlImpl::analog_level_maximum() const {
+ rtc::CritScope cs(crit_capture_);
return maximum_capture_level_;
}
bool GainControlImpl::stream_is_saturated() const {
+ rtc::CritScope cs(crit_capture_);
return stream_is_saturated_;
}
int GainControlImpl::set_target_level_dbfs(int level) {
- CriticalSectionScoped crit_scoped(crit_);
+ rtc::CritScope cs(crit_capture_);
if (level > 31 || level < 0) {
- return apm_->kBadParameterError;
+ return AudioProcessing::kBadParameterError;
}
target_level_dbfs_ = level;
@@ -304,13 +328,14 @@
}
int GainControlImpl::target_level_dbfs() const {
+ rtc::CritScope cs(crit_capture_);
return target_level_dbfs_;
}
int GainControlImpl::set_compression_gain_db(int gain) {
- CriticalSectionScoped crit_scoped(crit_);
+ rtc::CritScope cs(crit_capture_);
if (gain < 0 || gain > 90) {
- return apm_->kBadParameterError;
+ return AudioProcessing::kBadParameterError;
}
compression_gain_db_ = gain;
@@ -318,31 +343,35 @@
}
int GainControlImpl::compression_gain_db() const {
+ rtc::CritScope cs(crit_capture_);
return compression_gain_db_;
}
int GainControlImpl::enable_limiter(bool enable) {
- CriticalSectionScoped crit_scoped(crit_);
+ rtc::CritScope cs(crit_capture_);
limiter_enabled_ = enable;
return Configure();
}
bool GainControlImpl::is_limiter_enabled() const {
+ rtc::CritScope cs(crit_capture_);
return limiter_enabled_;
}
int GainControlImpl::Initialize() {
int err = ProcessingComponent::Initialize();
- if (err != apm_->kNoError || !is_component_enabled()) {
+ if (err != AudioProcessing::kNoError || !is_component_enabled()) {
return err;
}
AllocateRenderQueue();
+ rtc::CritScope cs_capture(crit_capture_);
const int n = num_handles();
RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;
+
capture_levels_.assign(n, analog_capture_level_);
- return apm_->kNoError;
+ return AudioProcessing::kNoError;
}
void GainControlImpl::AllocateRenderQueue() {
@@ -350,6 +379,9 @@
std::max<size_t>(static_cast<size_t>(1),
kMaxAllowedValuesOfSamplesPerFrame * num_handles());
+ rtc::CritScope cs_render(crit_render_);
+ rtc::CritScope cs_capture(crit_capture_);
+
if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
render_queue_element_max_size_ = new_render_queue_element_max_size;
std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
@@ -375,6 +407,9 @@
}
int GainControlImpl::InitializeHandle(void* handle) const {
+ rtc::CritScope cs_render(crit_render_);
+ rtc::CritScope cs_capture(crit_capture_);
+
return WebRtcAgc_Init(static_cast<Handle*>(handle),
minimum_capture_level_,
maximum_capture_level_,
@@ -383,6 +418,8 @@
}
int GainControlImpl::ConfigureHandle(void* handle) const {
+ rtc::CritScope cs_render(crit_render_);
+ rtc::CritScope cs_capture(crit_capture_);
WebRtcAgcConfig config;
// TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
// change the interface.
@@ -397,6 +434,7 @@
}
int GainControlImpl::num_handles_required() const {
+ // Not locked as it only relies on APM public API which is threadsafe.
return apm_->num_output_channels();
}
@@ -404,6 +442,6 @@
// The AGC has no get_error() function.
// (Despite listing errors in its interface...)
assert(handle != NULL);
- return apm_->kUnspecifiedError;
+ return AudioProcessing::kUnspecifiedError;
}
} // namespace webrtc