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Diffstat (limited to 'src/modules/audio_processing/main/source/audio_buffer.h')
-rw-r--r-- | src/modules/audio_processing/main/source/audio_buffer.h | 68 |
1 files changed, 0 insertions, 68 deletions
diff --git a/src/modules/audio_processing/main/source/audio_buffer.h b/src/modules/audio_processing/main/source/audio_buffer.h deleted file mode 100644 index 15f850b67b..0000000000 --- a/src/modules/audio_processing/main/source/audio_buffer.h +++ /dev/null @@ -1,68 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * 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_ - -#include "typedefs.h" - - -namespace webrtc { - -struct AudioChannel; -struct SplitAudioChannel; -class AudioFrame; - -class AudioBuffer { - public: - AudioBuffer(WebRtc_Word32 max_num_channels, WebRtc_Word32 samples_per_channel); - virtual ~AudioBuffer(); - - WebRtc_Word32 num_channels() const; - WebRtc_Word32 samples_per_channel() const; - WebRtc_Word32 samples_per_split_channel() const; - - WebRtc_Word16* data(WebRtc_Word32 channel) const; - WebRtc_Word16* low_pass_split_data(WebRtc_Word32 channel) const; - WebRtc_Word16* high_pass_split_data(WebRtc_Word32 channel) const; - WebRtc_Word16* mixed_low_pass_data(WebRtc_Word32 channel) const; - WebRtc_Word16* low_pass_reference(WebRtc_Word32 channel) const; - - WebRtc_Word32* analysis_filter_state1(WebRtc_Word32 channel) const; - WebRtc_Word32* analysis_filter_state2(WebRtc_Word32 channel) const; - WebRtc_Word32* synthesis_filter_state1(WebRtc_Word32 channel) const; - WebRtc_Word32* synthesis_filter_state2(WebRtc_Word32 channel) const; - - void DeinterleaveFrom(AudioFrame* audioFrame); - void InterleaveTo(AudioFrame* audioFrame) const; - void Mix(WebRtc_Word32 num_mixed_channels); - void CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels); - void CopyLowPassToReference(); - - private: - const WebRtc_Word32 max_num_channels_; - WebRtc_Word32 num_channels_; - WebRtc_Word32 num_mixed_channels_; - WebRtc_Word32 num_mixed_low_pass_channels_; - const WebRtc_Word32 samples_per_channel_; - WebRtc_Word32 samples_per_split_channel_; - bool reference_copied_; - - WebRtc_Word16* data_; - // TODO(ajm): Prefer to make these vectors if permitted... - AudioChannel* channels_; - SplitAudioChannel* split_channels_; - // TODO(ajm): improve this, we don't need the full 32 kHz space here. - AudioChannel* mixed_low_pass_channels_; - AudioChannel* low_pass_reference_channels_; -}; -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_ |