diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc index 7eac7ecfe4..1ceba383bd 100644 --- a/webrtc/modules/audio_processing/audio_buffer.cc +++ b/webrtc/modules/audio_processing/audio_buffer.cc @@ -125,24 +125,6 @@ class IFChannelBuffer { ChannelBuffer fbuf_; }; -class SplitChannelBuffer { - public: - SplitChannelBuffer(int samples_per_split_channel, int num_channels) - : low_(samples_per_split_channel, num_channels), - high_(samples_per_split_channel, num_channels) { - } - ~SplitChannelBuffer() {} - - int16_t* low_channel(int i) { return low_.ibuf()->channel(i); } - int16_t* high_channel(int i) { return high_.ibuf()->channel(i); } - float* low_channel_f(int i) { return low_.fbuf()->channel(i); } - float* high_channel_f(int i) { return high_.fbuf()->channel(i); } - - private: - IFChannelBuffer low_; - IFChannelBuffer high_; -}; - AudioBuffer::AudioBuffer(int input_samples_per_channel, int num_input_channels, int process_samples_per_channel, @@ -198,8 +180,10 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel, if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) { samples_per_split_channel_ = kSamplesPer16kHzChannel; - split_channels_.reset(new SplitChannelBuffer(samples_per_split_channel_, - num_proc_channels_)); + split_channels_low_.reset(new IFChannelBuffer(samples_per_split_channel_, + num_proc_channels_)); + split_channels_high_.reset(new IFChannelBuffer(samples_per_split_channel_, + num_proc_channels_)); filter_states_.reset(new SplitFilterStates[num_proc_channels_]); } } @@ -302,8 +286,9 @@ float* AudioBuffer::data_f(int channel) { } const int16_t* AudioBuffer::low_pass_split_data(int channel) const { - return split_channels_.get() ? split_channels_->low_channel(channel) - : data(channel); + return split_channels_low_.get() + ? split_channels_low_->ibuf()->channel(channel) + : data(channel); } int16_t* AudioBuffer::low_pass_split_data(int channel) { @@ -313,8 +298,9 @@ int16_t* AudioBuffer::low_pass_split_data(int channel) { } const float* AudioBuffer::low_pass_split_data_f(int channel) const { - return split_channels_.get() ? split_channels_->low_channel_f(channel) - : data_f(channel); + return split_channels_low_.get() + ? split_channels_low_->fbuf()->channel(channel) + : data_f(channel); } float* AudioBuffer::low_pass_split_data_f(int channel) { @@ -324,7 +310,9 @@ float* AudioBuffer::low_pass_split_data_f(int channel) { } const int16_t* AudioBuffer::high_pass_split_data(int channel) const { - return split_channels_.get() ? split_channels_->high_channel(channel) : NULL; + return split_channels_high_.get() + ? split_channels_high_->ibuf()->channel(channel) + : NULL; } int16_t* AudioBuffer::high_pass_split_data(int channel) { @@ -333,8 +321,9 @@ int16_t* AudioBuffer::high_pass_split_data(int channel) { } const float* AudioBuffer::high_pass_split_data_f(int channel) const { - return split_channels_.get() ? split_channels_->high_channel_f(channel) - : NULL; + return split_channels_high_.get() + ? split_channels_high_->fbuf()->channel(channel) + : NULL; } float* AudioBuffer::high_pass_split_data_f(int channel) { diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h index 6b1a46f957..5c26ae29d0 100644 --- a/webrtc/modules/audio_processing/audio_buffer.h +++ b/webrtc/modules/audio_processing/audio_buffer.h @@ -23,7 +23,6 @@ namespace webrtc { class PushSincResampler; -class SplitChannelBuffer; class IFChannelBuffer; struct SplitFilterStates { @@ -115,7 +114,8 @@ class AudioBuffer { const float* keyboard_data_; scoped_ptr channels_; - scoped_ptr split_channels_; + scoped_ptr split_channels_low_; + scoped_ptr split_channels_high_; scoped_ptr filter_states_; scoped_ptr > mixed_low_pass_channels_; scoped_ptr > low_pass_reference_channels_;