aboutsummaryrefslogtreecommitdiff
path: root/webrtc/modules/audio_processing/gain_control_impl.h
diff options
context:
space:
mode:
Diffstat (limited to 'webrtc/modules/audio_processing/gain_control_impl.h')
-rw-r--r--webrtc/modules/audio_processing/gain_control_impl.h51
1 files changed, 37 insertions, 14 deletions
diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h
index f24d200cf2..72789ba5e1 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/webrtc/modules/audio_processing/gain_control_impl.h
@@ -13,19 +13,23 @@
#include <vector>
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_audio/swap_queue.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioBuffer;
-class CriticalSectionWrapper;
class GainControlImpl : public GainControl,
public ProcessingComponent {
public:
GainControlImpl(const AudioProcessing* apm,
- CriticalSectionWrapper* crit);
+ rtc::CriticalSection* crit_render,
+ rtc::CriticalSection* crit_capture);
virtual ~GainControlImpl();
int ProcessRenderAudio(AudioBuffer* audio);
@@ -41,6 +45,9 @@ class GainControlImpl : public GainControl,
bool is_limiter_enabled() const override;
Mode mode() const override;
+ // Reads render side data that has been queued on the render call.
+ void ReadQueuedRenderData();
+
private:
// GainControl implementation.
int Enable(bool enable) override;
@@ -61,21 +68,37 @@ class GainControlImpl : public GainControl,
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
- int num_handles_required() const override;
+ size_t num_handles_required() const override;
int GetHandleError(void* handle) const override;
+ void AllocateRenderQueue();
+
+ // Not guarded as its public API is thread safe.
const AudioProcessing* apm_;
- CriticalSectionWrapper* crit_;
- Mode mode_;
- int minimum_capture_level_;
- int maximum_capture_level_;
- bool limiter_enabled_;
- int target_level_dbfs_;
- int compression_gain_db_;
- std::vector<int> capture_levels_;
- int analog_capture_level_;
- bool was_analog_level_set_;
- bool stream_is_saturated_;
+
+ rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
+ rtc::CriticalSection* const crit_capture_;
+
+ Mode mode_ GUARDED_BY(crit_capture_);
+ int minimum_capture_level_ GUARDED_BY(crit_capture_);
+ int maximum_capture_level_ GUARDED_BY(crit_capture_);
+ bool limiter_enabled_ GUARDED_BY(crit_capture_);
+ int target_level_dbfs_ GUARDED_BY(crit_capture_);
+ int compression_gain_db_ GUARDED_BY(crit_capture_);
+ std::vector<int> capture_levels_ GUARDED_BY(crit_capture_);
+ int analog_capture_level_ GUARDED_BY(crit_capture_);
+ bool was_analog_level_set_ GUARDED_BY(crit_capture_);
+ bool stream_is_saturated_ GUARDED_BY(crit_capture_);
+
+ size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+ GUARDED_BY(crit_capture_);
+ std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
+ std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+ // Lock protection not needed.
+ rtc::scoped_ptr<
+ SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+ render_signal_queue_;
};
} // namespace webrtc