Revert "Simplification and refactoring of the AudioBuffer code"
This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128.
Reason for revert: internal test failures
Original change's description:
> 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}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index dd9b768..16d5616 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -23,142 +23,114 @@
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:
- AudioBuffer(size_t input_rate,
- size_t input_num_channels,
- size_t buffer_rate,
- size_t buffer_num_channels,
- size_t output_rate);
+ // 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);
virtual ~AudioBuffer();
- AudioBuffer(const AudioBuffer&) = delete;
- AudioBuffer& operator=(const AudioBuffer&) = delete;
-
- // 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.
+ 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;
- 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.
+ // Returns a pointer array to the full-band channels.
// Usage:
// channels()[channel][sample].
// Where:
- // 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(); }
+ // 0 <= channel < |num_proc_channels_|
+ // 0 <= sample < |proc_num_frames_|
+ float* const* channels_f();
+ const float* const* channels_const_f() const;
- // Returns pointer arrays to the bands for a specific channel.
+ // Returns a pointer array to the bands for a specific channel.
// Usage:
// split_bands(channel)[band][sample].
// Where:
- // 0 <= channel < |buffer_num_channels_|
+ // 0 <= channel < |num_proc_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_split_frames_|
- 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);
- }
+ float* const* split_bands_f(size_t channel);
+ const float* const* split_bands_const_f(size_t channel) const;
// Returns a pointer array to the channels for a specific band.
// Usage:
// split_channels(band)[channel][sample].
// Where:
// 0 <= band < |num_bands_|
- // 0 <= channel < |buffer_num_channels_|
+ // 0 <= channel < |num_proc_channels_|
// 0 <= sample < |num_split_frames_|
- 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;
- }
- }
+ const float* const* split_channels_const_f(Band band) const;
- // Copies data into the buffer.
- void CopyFrom(const AudioFrame* frame);
+ // 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.
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 buffer data into frequency bands.
+ // Splits the signal into different bands.
void SplitIntoFrequencyBands();
-
- // Recombines the frequency bands into a full-band signal.
+ // Recombine the different bands into one signal.
void MergeFrequencyBands();
// Copies the split bands data into the integer two-dimensional array.
- void ExportSplitChannelData(size_t channel, int16_t* const* split_band_data);
+ void CopySplitChannelDataTo(size_t channel, int16_t* const* split_band_data);
// Copies the data in the integer two-dimensional array into the split_bands
// data.
- void ImportSplitChannelData(size_t channel,
- const int16_t* const* split_band_data);
+ void CopySplitChannelDataFrom(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);
- void RestoreNumChannels();
+ // Called from DeinterleaveFrom() and CopyFrom().
+ void InitForNewData();
+ // 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 input_num_channels_;
- const size_t buffer_num_frames_;
- const size_t buffer_num_channels_;
+ 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 output_num_frames_;
-
size_t num_channels_;
+
size_t num_bands_;
size_t num_split_frames_;
- std::unique_ptr<ChannelBuffer<float>> data_;
- std::unique_ptr<ChannelBuffer<float>> split_data_;
+ std::unique_ptr<IFChannelBuffer> data_;
+ std::unique_ptr<IFChannelBuffer> split_data_;
std::unique_ptr<SplittingFilter> splitting_filter_;
- std::unique_ptr<ChannelBuffer<float>> output_buffer_;
+ std::unique_ptr<IFChannelBuffer> input_buffer_;
+ std::unique_ptr<IFChannelBuffer> output_buffer_;
+ std::unique_ptr<ChannelBuffer<float>> process_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