Simplification and refactoring of the AudioBuffer code
This CL performs a major refactoring and simplification
of the AudioBuffer code that.
-Removes 7 of the 9 internal buffers of the AudioBuffer.
-Avoids the implicit copying required to keep the
internal buffers in sync.
-Removes all code relating to handling of fixed-point
sample data in the AudioBuffer.
-Changes the naming of the class methods to reflect
that only floating point is handled.
-Corrects some bugs in the code.
-Extends the handling of internal downmixing to be
more generic.
Bug: webrtc:10882
Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28928}
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index 16d5616..dd9b768 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -23,114 +23,142 @@
namespace webrtc {
-class IFChannelBuffer;
class PushSincResampler;
class SplittingFilter;
enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
+// Stores any audio data in a way that allows the audio processing module to
+// operate on it in a controlled manner.
class AudioBuffer {
public:
- // TODO(ajm): Switch to take ChannelLayouts.
- AudioBuffer(size_t input_num_frames,
- size_t num_input_channels,
- size_t process_num_frames,
- size_t num_process_channels,
- size_t output_num_frames);
+ AudioBuffer(size_t input_rate,
+ size_t input_num_channels,
+ size_t buffer_rate,
+ size_t buffer_num_channels,
+ size_t output_rate);
virtual ~AudioBuffer();
- size_t num_channels() const;
- size_t num_proc_channels() const { return num_proc_channels_; }
- void set_num_channels(size_t num_channels);
- size_t num_frames() const;
- size_t num_frames_per_band() const;
- size_t num_bands() const;
+ AudioBuffer(const AudioBuffer&) = delete;
+ AudioBuffer& operator=(const AudioBuffer&) = delete;
- // Returns a pointer array to the full-band channels.
+ // Specify that downmixing should be done by selecting a single channel.
+ void set_downmixing_to_specific_channel(size_t channel);
+
+ // Specify that downmixing should be done by averaging all channels,.
+ void set_downmixing_by_averaging();
+
+ // Set the number of channels in the buffer. The specified number of channels
+ // cannot be larger than the specified buffer_num_channels. The number is also
+ // reset at each call to CopyFrom or InterleaveFrom.
+ void set_num_channels(size_t num_channels);
+
+ size_t num_channels() const { return num_channels_; }
+ size_t num_frames() const { return buffer_num_frames_; }
+ size_t num_frames_per_band() const { return num_split_frames_; }
+ size_t num_bands() const { return num_bands_; }
+
+ // Returns pointer arrays to the full-band channels.
// Usage:
// channels()[channel][sample].
// Where:
- // 0 <= channel < |num_proc_channels_|
- // 0 <= sample < |proc_num_frames_|
- float* const* channels_f();
- const float* const* channels_const_f() const;
+ // 0 <= channel < |buffer_num_channels_|
+ // 0 <= sample < |buffer_num_frames_|
+ float* const* channels() { return data_->channels(); }
+ const float* const* channels_const() const { return data_->channels(); }
- // Returns a pointer array to the bands for a specific channel.
+ // Returns pointer arrays to the bands for a specific channel.
// Usage:
// split_bands(channel)[band][sample].
// Where:
- // 0 <= channel < |num_proc_channels_|
+ // 0 <= channel < |buffer_num_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_split_frames_|
- float* const* split_bands_f(size_t channel);
- const float* const* split_bands_const_f(size_t channel) const;
+ const float* const* split_bands_const(size_t channel) const {
+ return split_data_.get() ? split_data_->bands(channel)
+ : data_->bands(channel);
+ }
+ float* const* split_bands(size_t channel) {
+ return split_data_.get() ? split_data_->bands(channel)
+ : data_->bands(channel);
+ }
// Returns a pointer array to the channels for a specific band.
// Usage:
// split_channels(band)[channel][sample].
// Where:
// 0 <= band < |num_bands_|
- // 0 <= channel < |num_proc_channels_|
+ // 0 <= channel < |buffer_num_channels_|
// 0 <= sample < |num_split_frames_|
- const float* const* split_channels_const_f(Band band) const;
+ const float* const* split_channels_const(Band band) const {
+ if (split_data_.get()) {
+ return split_data_->channels(band);
+ } else {
+ return band == kBand0To8kHz ? data_->channels() : nullptr;
+ }
+ }
- // Use for int16 interleaved data.
- void DeinterleaveFrom(const AudioFrame* audioFrame);
- // If |data_changed| is false, only the non-audio data members will be copied
- // to |frame|.
- void InterleaveTo(AudioFrame* frame) const;
-
- // Use for float deinterleaved data.
+ // Copies data into the buffer.
+ void CopyFrom(const AudioFrame* frame);
void CopyFrom(const float* const* data, const StreamConfig& stream_config);
+
+ // Copies data from the buffer.
+ void CopyTo(AudioFrame* frame) const;
void CopyTo(const StreamConfig& stream_config, float* const* data);
- // Splits the signal into different bands.
+ // Splits the buffer data into frequency bands.
void SplitIntoFrequencyBands();
- // Recombine the different bands into one signal.
+
+ // Recombines the frequency bands into a full-band signal.
void MergeFrequencyBands();
// Copies the split bands data into the integer two-dimensional array.
- void CopySplitChannelDataTo(size_t channel, int16_t* const* split_band_data);
+ void ExportSplitChannelData(size_t channel, int16_t* const* split_band_data);
// Copies the data in the integer two-dimensional array into the split_bands
// data.
- void CopySplitChannelDataFrom(size_t channel,
- const int16_t* const* split_band_data);
+ void ImportSplitChannelData(size_t channel,
+ const int16_t* const* split_band_data);
static const size_t kMaxSplitFrameLength = 160;
static const size_t kMaxNumBands = 3;
+ // Deprecated methods, will be removed soon.
+ float* const* channels_f() { return channels(); }
+ const float* const* channels_const_f() const { return channels_const(); }
+ const float* const* split_bands_const_f(size_t channel) const {
+ return split_bands_const(channel);
+ }
+ float* const* split_bands_f(size_t channel) { return split_bands(channel); }
+ const float* const* split_channels_const_f(Band band) const {
+ return split_channels_const(band);
+ }
+ void DeinterleaveFrom(const AudioFrame* frame) { CopyFrom(frame); }
+ void InterleaveTo(AudioFrame* frame) const { CopyTo(frame); }
+
private:
FRIEND_TEST_ALL_PREFIXES(AudioBufferTest,
SetNumChannelsSetsChannelBuffersNumChannels);
- // Called from DeinterleaveFrom() and CopyFrom().
- void InitForNewData();
+ void RestoreNumChannels();
- // The audio is passed into DeinterleaveFrom() or CopyFrom() with input
- // format (samples per channel and number of channels).
const size_t input_num_frames_;
- const size_t num_input_channels_;
- // The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
- // format.
- const size_t proc_num_frames_;
- const size_t num_proc_channels_;
- // The audio is returned by InterleaveTo() and CopyTo() with output samples
- // per channels and the current number of channels. This last one can be
- // changed at any time using set_num_channels().
+ const size_t input_num_channels_;
+ const size_t buffer_num_frames_;
+ const size_t buffer_num_channels_;
const size_t output_num_frames_;
- size_t num_channels_;
+ size_t num_channels_;
size_t num_bands_;
size_t num_split_frames_;
- std::unique_ptr<IFChannelBuffer> data_;
- std::unique_ptr<IFChannelBuffer> split_data_;
+ std::unique_ptr<ChannelBuffer<float>> data_;
+ std::unique_ptr<ChannelBuffer<float>> split_data_;
std::unique_ptr<SplittingFilter> splitting_filter_;
- std::unique_ptr<IFChannelBuffer> input_buffer_;
- std::unique_ptr<IFChannelBuffer> output_buffer_;
- std::unique_ptr<ChannelBuffer<float>> process_buffer_;
+ std::unique_ptr<ChannelBuffer<float>> output_buffer_;
std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_;
std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_;
+ bool downmix_by_averaging_ = true;
+ size_t channel_for_downmixing_ = 0;
};
} // namespace webrtc