diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h | 117 |
1 files changed, 83 insertions, 34 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h b/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h index 2638bef6058..67e4f485043 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h @@ -8,75 +8,124 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_ +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ +#include <vector> + +#include "webrtc/modules/audio_processing/common.h" +#include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/scoped_vector.h" #include "webrtc/typedefs.h" namespace webrtc { -struct AudioChannel; -struct SplitAudioChannel; +class PushSincResampler; +class SplitChannelBuffer; +class IFChannelBuffer; + +struct SplitFilterStates { + SplitFilterStates() { + memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1)); + memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2)); + memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1)); + memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2)); + } + + static const int kStateSize = 6; + int analysis_filter_state1[kStateSize]; + int analysis_filter_state2[kStateSize]; + int synthesis_filter_state1[kStateSize]; + int synthesis_filter_state2[kStateSize]; +}; class AudioBuffer { public: - AudioBuffer(int max_num_channels, int samples_per_channel); + // TODO(ajm): Switch to take ChannelLayouts. + AudioBuffer(int input_samples_per_channel, + int num_input_channels, + int process_samples_per_channel, + int num_process_channels, + int output_samples_per_channel); virtual ~AudioBuffer(); int num_channels() const; int samples_per_channel() const; int samples_per_split_channel() const; + int samples_per_keyboard_channel() const; - int16_t* data(int channel) const; - int16_t* low_pass_split_data(int channel) const; - int16_t* high_pass_split_data(int channel) const; - int16_t* mixed_data(int channel) const; - int16_t* mixed_low_pass_data(int channel) const; - int16_t* low_pass_reference(int channel) const; + int16_t* data(int channel); + const int16_t* data(int channel) const; + int16_t* low_pass_split_data(int channel); + const int16_t* low_pass_split_data(int channel) const; + int16_t* high_pass_split_data(int channel); + const int16_t* high_pass_split_data(int channel) const; + const int16_t* mixed_data(int channel) const; + const int16_t* mixed_low_pass_data(int channel) const; + const int16_t* low_pass_reference(int channel) const; - int32_t* analysis_filter_state1(int channel) const; - int32_t* analysis_filter_state2(int channel) const; - int32_t* synthesis_filter_state1(int channel) const; - int32_t* synthesis_filter_state2(int channel) const; + // Float versions of the accessors, with automatic conversion back and forth + // as necessary. The range of the numbers are the same as for int16_t. + float* data_f(int channel); + float* low_pass_split_data_f(int channel); + float* high_pass_split_data_f(int channel); + + const float* keyboard_data() const; + + SplitFilterStates* filter_states(int channel); void set_activity(AudioFrame::VADActivity activity); AudioFrame::VADActivity activity() const; - bool is_muted() const; - + // Use for int16 interleaved data. void DeinterleaveFrom(AudioFrame* audioFrame); void InterleaveTo(AudioFrame* audioFrame) const; // If |data_changed| is false, only the non-audio data members will be copied // to |frame|. void InterleaveTo(AudioFrame* frame, bool data_changed) const; - void Mix(int num_mixed_channels); + + // Use for float deinterleaved data. + void CopyFrom(const float* const* data, + int samples_per_channel, + AudioProcessing::ChannelLayout layout); + void CopyTo(int samples_per_channel, + AudioProcessing::ChannelLayout layout, + float* const* data); + void CopyAndMix(int num_mixed_channels); void CopyAndMixLowPass(int num_mixed_channels); void CopyLowPassToReference(); private: - const int max_num_channels_; - int num_channels_; + // Called from DeinterleaveFrom() and CopyFrom(). + void InitForNewData(); + + const int input_samples_per_channel_; + const int num_input_channels_; + const int proc_samples_per_channel_; + const int num_proc_channels_; + const int output_samples_per_channel_; + int samples_per_split_channel_; int num_mixed_channels_; int num_mixed_low_pass_channels_; - // Whether the original data was replaced with mixed data. - bool data_was_mixed_; - const int samples_per_channel_; - int samples_per_split_channel_; bool reference_copied_; AudioFrame::VADActivity activity_; - bool is_muted_; - - int16_t* data_; - scoped_array<AudioChannel> channels_; - scoped_array<SplitAudioChannel> split_channels_; - scoped_array<AudioChannel> mixed_channels_; - // TODO(andrew): improve this, we don't need the full 32 kHz space here. - scoped_array<AudioChannel> mixed_low_pass_channels_; - scoped_array<AudioChannel> low_pass_reference_channels_; + + const float* keyboard_data_; + scoped_ptr<IFChannelBuffer> channels_; + scoped_ptr<SplitChannelBuffer> split_channels_; + scoped_ptr<SplitFilterStates[]> filter_states_; + scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_; + scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_; + scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_; + scoped_ptr<ChannelBuffer<float> > input_buffer_; + scoped_ptr<ChannelBuffer<float> > process_buffer_; + ScopedVector<PushSincResampler> input_resamplers_; + ScopedVector<PushSincResampler> output_resamplers_; }; + } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_ +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ |