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-rw-r--r--chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h132
1 files changed, 100 insertions, 32 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h
index e48a2c18a4f..d34f305a96b 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -19,9 +19,10 @@
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
+
class AudioBuffer;
class CriticalSectionWrapper;
-class EchoCancellationImplWrapper;
+class EchoCancellationImpl;
class EchoControlMobileImpl;
class FileWrapper;
class GainControlImpl;
@@ -39,44 +40,92 @@ class Event;
} // namespace audioproc
#endif
-class AudioProcessingImpl : public AudioProcessing {
+class AudioRate {
public:
- enum {
- kSampleRate8kHz = 8000,
- kSampleRate16kHz = 16000,
- kSampleRate32kHz = 32000
- };
+ explicit AudioRate(int sample_rate_hz)
+ : rate_(sample_rate_hz),
+ samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {}
+ virtual ~AudioRate() {}
+
+ void set(int rate) {
+ rate_ = rate;
+ samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
+ }
- explicit AudioProcessingImpl(int id);
- virtual ~AudioProcessingImpl();
+ int rate() const { return rate_; }
+ int samples_per_channel() const { return samples_per_channel_; }
- CriticalSectionWrapper* crit() const;
+ private:
+ int rate_;
+ int samples_per_channel_;
+};
- int split_sample_rate_hz() const;
- bool was_stream_delay_set() const;
+class AudioFormat : public AudioRate {
+ public:
+ AudioFormat(int sample_rate_hz, int num_channels)
+ : AudioRate(sample_rate_hz),
+ num_channels_(num_channels) {}
+ virtual ~AudioFormat() {}
+
+ void set(int rate, int num_channels) {
+ AudioRate::set(rate);
+ num_channels_ = num_channels;
+ }
+
+ int num_channels() const { return num_channels_; }
+
+ private:
+ int num_channels_;
+};
+
+class AudioProcessingImpl : public AudioProcessing {
+ public:
+ explicit AudioProcessingImpl(const Config& config);
+ virtual ~AudioProcessingImpl();
// AudioProcessing methods.
virtual int Initialize() OVERRIDE;
- virtual int InitializeLocked();
+ virtual int Initialize(int input_sample_rate_hz,
+ int output_sample_rate_hz,
+ int reverse_sample_rate_hz,
+ ChannelLayout input_layout,
+ ChannelLayout output_layout,
+ ChannelLayout reverse_layout) OVERRIDE;
virtual void SetExtraOptions(const Config& config) OVERRIDE;
virtual int EnableExperimentalNs(bool enable) OVERRIDE;
virtual bool experimental_ns_enabled() const OVERRIDE {
return false;
}
virtual int set_sample_rate_hz(int rate) OVERRIDE;
+ virtual int input_sample_rate_hz() const OVERRIDE;
virtual int sample_rate_hz() const OVERRIDE;
- virtual int set_num_channels(int input_channels,
- int output_channels) OVERRIDE;
+ virtual int proc_sample_rate_hz() const OVERRIDE;
+ virtual int proc_split_sample_rate_hz() const OVERRIDE;
virtual int num_input_channels() const OVERRIDE;
virtual int num_output_channels() const OVERRIDE;
- virtual int set_num_reverse_channels(int channels) OVERRIDE;
virtual int num_reverse_channels() const OVERRIDE;
+ virtual void set_output_will_be_muted(bool muted) OVERRIDE;
+ virtual bool output_will_be_muted() const OVERRIDE;
virtual int ProcessStream(AudioFrame* frame) OVERRIDE;
+ virtual int ProcessStream(const float* const* src,
+ int samples_per_channel,
+ int input_sample_rate_hz,
+ ChannelLayout input_layout,
+ int output_sample_rate_hz,
+ ChannelLayout output_layout,
+ float* const* dest) OVERRIDE;
virtual int AnalyzeReverseStream(AudioFrame* frame) OVERRIDE;
+ virtual int AnalyzeReverseStream(const float* const* data,
+ int samples_per_channel,
+ int sample_rate_hz,
+ ChannelLayout layout) OVERRIDE;
virtual int set_stream_delay_ms(int delay) OVERRIDE;
virtual int stream_delay_ms() const OVERRIDE;
+ virtual bool was_stream_delay_set() const OVERRIDE;
virtual void set_delay_offset_ms(int offset) OVERRIDE;
virtual int delay_offset_ms() const OVERRIDE;
+ virtual void set_stream_key_pressed(bool key_pressed) OVERRIDE;
+ virtual bool stream_key_pressed() const OVERRIDE;
virtual int StartDebugRecording(
const char filename[kMaxFilenameSize]) OVERRIDE;
virtual int StartDebugRecording(FILE* handle) OVERRIDE;
@@ -89,18 +138,32 @@ class AudioProcessingImpl : public AudioProcessing {
virtual NoiseSuppression* noise_suppression() const OVERRIDE;
virtual VoiceDetection* voice_detection() const OVERRIDE;
- // Module methods.
- virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
+ protected:
+ // Overridden in a mock.
+ virtual int InitializeLocked();
private:
+ int InitializeLocked(int input_sample_rate_hz,
+ int output_sample_rate_hz,
+ int reverse_sample_rate_hz,
+ int num_input_channels,
+ int num_output_channels,
+ int num_reverse_channels);
+ int MaybeInitializeLocked(int input_sample_rate_hz,
+ int output_sample_rate_hz,
+ int reverse_sample_rate_hz,
+ int num_input_channels,
+ int num_output_channels,
+ int num_reverse_channels);
+ int ProcessStreamLocked();
+ int AnalyzeReverseStreamLocked();
+
bool is_data_processed() const;
- bool interleave_needed(bool is_data_processed) const;
+ bool output_copy_needed(bool is_data_processed) const;
bool synthesis_needed(bool is_data_processed) const;
bool analysis_needed(bool is_data_processed) const;
- int id_;
-
- EchoCancellationImplWrapper* echo_cancellation_;
+ EchoCancellationImpl* echo_cancellation_;
EchoControlMobileImpl* echo_control_mobile_;
GainControlImpl* gain_control_;
HighPassFilterImpl* high_pass_filter_;
@@ -110,29 +173,34 @@ class AudioProcessingImpl : public AudioProcessing {
std::list<ProcessingComponent*> component_list_;
CriticalSectionWrapper* crit_;
- AudioBuffer* render_audio_;
- AudioBuffer* capture_audio_;
+ scoped_ptr<AudioBuffer> render_audio_;
+ scoped_ptr<AudioBuffer> capture_audio_;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// TODO(andrew): make this more graceful. Ideally we would split this stuff
// out into a separate class with an "enabled" and "disabled" implementation.
int WriteMessageToDebugFile();
int WriteInitMessage();
scoped_ptr<FileWrapper> debug_file_;
- scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
- std::string event_str_; // Memory for protobuf serialization.
+ scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
+ std::string event_str_; // Memory for protobuf serialization.
#endif
- int sample_rate_hz_;
- int split_sample_rate_hz_;
- int samples_per_channel_;
+ AudioFormat fwd_in_format_;
+ AudioFormat fwd_proc_format_;
+ AudioRate fwd_out_format_;
+ AudioFormat rev_in_format_;
+ AudioFormat rev_proc_format_;
+ int split_rate_;
+
int stream_delay_ms_;
int delay_offset_ms_;
bool was_stream_delay_set_;
- int num_reverse_channels_;
- int num_input_channels_;
- int num_output_channels_;
+ bool output_will_be_muted_;
+
+ bool key_pressed_;
};
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-07 20:32:12 +0000