From a181c9ad17dd749eaa384f48d5c953fcc66d435d Mon Sep 17 00:00:00 2001 From: Alejandro Luebs Date: Thu, 30 Jun 2016 15:33:37 -0700 Subject: [PATCH] Keep track of the user-facing number of channels in a ChannelBuffer Before this change the ChannelBuffer had a fixed number of channels. This meant for example that when the Beamformer would reduce the number of channels to one, the merging filter bank was still merging all the channels, which was unnecessary since they were not processed and just discarded later. This change doesn't change the signal at all. It just reflects the number of channels in the ChannelBuffer, reducing the complexity. R=henrik.lundin@webrtc.org, peah@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/2053773002 . Cr-Commit-Position: refs/heads/master@{#13352} --- webrtc/common_audio/BUILD.gn | 1 + webrtc/common_audio/channel_buffer.cc | 4 +- webrtc/common_audio/channel_buffer.h | 33 +++++++--- .../common_audio/channel_buffer_unittest.cc | 65 +++++++++++++++++++ webrtc/common_audio/common_audio.gyp | 1 + webrtc/modules/BUILD.gn | 1 + .../modules/audio_processing/audio_buffer.cc | 8 +++ .../modules/audio_processing/audio_buffer.h | 2 + .../audio_processing/audio_buffer_unittest.cc | 48 ++++++++++++++ .../audio_processing/splitting_filter.cc | 8 +-- webrtc/modules/modules.gyp | 1 + 11 files changed, 158 insertions(+), 14 deletions(-) create mode 100644 webrtc/common_audio/channel_buffer_unittest.cc create mode 100644 webrtc/modules/audio_processing/audio_buffer_unittest.cc diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn index 50c4439035..f64ece3be3 100644 --- a/webrtc/common_audio/BUILD.gn +++ b/webrtc/common_audio/BUILD.gn @@ -259,6 +259,7 @@ if (rtc_include_tests) { "audio_ring_buffer_unittest.cc", "audio_util_unittest.cc", "blocker_unittest.cc", + "channel_buffer_unittest.cc", "fir_filter_unittest.cc", "lapped_transform_unittest.cc", "real_fourier_unittest.cc", diff --git a/webrtc/common_audio/channel_buffer.cc b/webrtc/common_audio/channel_buffer.cc index 349d4245cc..cc0001b944 100644 --- a/webrtc/common_audio/channel_buffer.cc +++ b/webrtc/common_audio/channel_buffer.cc @@ -47,6 +47,7 @@ const ChannelBuffer* IFChannelBuffer::fbuf_const() const { void IFChannelBuffer::RefreshF() const { if (!fvalid_) { RTC_DCHECK(ivalid_); + fbuf_.set_num_channels(ibuf_.num_channels()); const int16_t* const* int_channels = ibuf_.channels(); float* const* float_channels = fbuf_.channels(); for (size_t i = 0; i < ibuf_.num_channels(); ++i) { @@ -62,8 +63,9 @@ void IFChannelBuffer::RefreshI() const { if (!ivalid_) { RTC_DCHECK(fvalid_); int16_t* const* int_channels = ibuf_.channels(); + ibuf_.set_num_channels(fbuf_.num_channels()); const float* const* float_channels = fbuf_.channels(); - for (size_t i = 0; i < ibuf_.num_channels(); ++i) { + for (size_t i = 0; i < fbuf_.num_channels(); ++i) { FloatS16ToS16(float_channels[i], ibuf_.num_frames(), int_channels[i]); diff --git a/webrtc/common_audio/channel_buffer.h b/webrtc/common_audio/channel_buffer.h index faecec9194..d6661de4ff 100644 --- a/webrtc/common_audio/channel_buffer.h +++ b/webrtc/common_audio/channel_buffer.h @@ -48,13 +48,14 @@ class ChannelBuffer { bands_(new T*[num_channels * num_bands]), num_frames_(num_frames), num_frames_per_band_(num_frames / num_bands), + num_allocated_channels_(num_channels), num_channels_(num_channels), num_bands_(num_bands) { - for (size_t i = 0; i < num_channels_; ++i) { + for (size_t i = 0; i < num_allocated_channels_; ++i) { for (size_t j = 0; j < num_bands_; ++j) { - channels_[j * num_channels_ + i] = + channels_[j * num_allocated_channels_ + i] = &data_[i * num_frames_ + j * num_frames_per_band_]; - bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i]; + bands_[i * num_bands_ + j] = channels_[j * num_allocated_channels_ + i]; } } } @@ -63,7 +64,7 @@ class ChannelBuffer { // Usage: // channels()[channel][sample]. // Where: - // 0 <= channel < |num_channels_| + // 0 <= channel < |num_allocated_channels_| // 0 <= sample < |num_frames_| T* const* channels() { return channels(0); } const T* const* channels() const { return channels(0); } @@ -73,11 +74,11 @@ class ChannelBuffer { // channels(band)[channel][sample]. // Where: // 0 <= band < |num_bands_| - // 0 <= channel < |num_channels_| + // 0 <= channel < |num_allocated_channels_| // 0 <= sample < |num_frames_per_band_| const T* const* channels(size_t band) const { RTC_DCHECK_LT(band, num_bands_); - return &channels_[band * num_channels_]; + return &channels_[band * num_allocated_channels_]; } T* const* channels(size_t band) { const ChannelBuffer* t = this; @@ -118,7 +119,12 @@ class ChannelBuffer { size_t num_frames_per_band() const { return num_frames_per_band_; } size_t num_channels() const { return num_channels_; } size_t num_bands() const { return num_bands_; } - size_t size() const {return num_frames_ * num_channels_; } + size_t size() const {return num_frames_ * num_allocated_channels_; } + + void set_num_channels(size_t num_channels) { + RTC_DCHECK_LE(num_channels, num_allocated_channels_); + num_channels_ = num_channels; + } void SetDataForTesting(const T* data, size_t size) { RTC_CHECK_EQ(size, this->size()); @@ -131,7 +137,10 @@ class ChannelBuffer { std::unique_ptr bands_; const size_t num_frames_; const size_t num_frames_per_band_; - const size_t num_channels_; + // Number of channels the internal buffer holds. + const size_t num_allocated_channels_; + // Number of channels the user sees. + size_t num_channels_; const size_t num_bands_; }; @@ -152,7 +161,13 @@ class IFChannelBuffer { size_t num_frames() const { return ibuf_.num_frames(); } size_t num_frames_per_band() const { return ibuf_.num_frames_per_band(); } - size_t num_channels() const { return ibuf_.num_channels(); } + size_t num_channels() const { + return ivalid_ ? ibuf_.num_channels() : fbuf_.num_channels(); + } + void set_num_channels(size_t num_channels) { + ibuf_.set_num_channels(num_channels); + fbuf_.set_num_channels(num_channels); + } size_t num_bands() const { return ibuf_.num_bands(); } private: diff --git a/webrtc/common_audio/channel_buffer_unittest.cc b/webrtc/common_audio/channel_buffer_unittest.cc new file mode 100644 index 0000000000..98731fc5e7 --- /dev/null +++ b/webrtc/common_audio/channel_buffer_unittest.cc @@ -0,0 +1,65 @@ +/* + * Copyright (c) 2016 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. + */ + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/common_audio/channel_buffer.h" + +namespace webrtc { + +namespace { + +const size_t kNumFrames = 480u; +const size_t kStereo = 2u; +const size_t kMono = 1u; + +void ExpectNumChannels(const IFChannelBuffer& ifchb, size_t num_channels) { + EXPECT_EQ(ifchb.ibuf_const()->num_channels(), num_channels); + EXPECT_EQ(ifchb.fbuf_const()->num_channels(), num_channels); + EXPECT_EQ(ifchb.num_channels(), num_channels); +} + +} // namespace + +TEST(ChannelBufferTest, SetNumChannelsSetsNumChannels) { + ChannelBuffer chb(kNumFrames, kStereo); + EXPECT_EQ(chb.num_channels(), kStereo); + chb.set_num_channels(kMono); + EXPECT_EQ(chb.num_channels(), kMono); +} + +TEST(IFChannelBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) { + IFChannelBuffer ifchb(kNumFrames, kStereo); + ExpectNumChannels(ifchb, kStereo); + ifchb.set_num_channels(kMono); + ExpectNumChannels(ifchb, kMono); +} + +TEST(IFChannelBufferTest, SettingNumChannelsOfOneChannelBufferSetsTheOther) { + IFChannelBuffer ifchb(kNumFrames, kStereo); + ExpectNumChannels(ifchb, kStereo); + ifchb.ibuf()->set_num_channels(kMono); + ExpectNumChannels(ifchb, kMono); + ifchb.fbuf()->set_num_channels(kStereo); + ExpectNumChannels(ifchb, kStereo); +} + +#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) +TEST(ChannelBufferTest, SetNumChannelsDeathTest) { + ChannelBuffer chb(kNumFrames, kMono); + EXPECT_DEATH(chb.set_num_channels(kStereo), "num_channels"); +} + +TEST(IFChannelBufferTest, SetNumChannelsDeathTest) { + IFChannelBuffer ifchb(kNumFrames, kMono); + EXPECT_DEATH(ifchb.ibuf()->set_num_channels(kStereo), "num_channels"); +} +#endif + +} // namespace webrtc diff --git a/webrtc/common_audio/common_audio.gyp b/webrtc/common_audio/common_audio.gyp index fcd7a411bf..41f6111290 100644 --- a/webrtc/common_audio/common_audio.gyp +++ b/webrtc/common_audio/common_audio.gyp @@ -244,6 +244,7 @@ 'audio_ring_buffer_unittest.cc', 'audio_util_unittest.cc', 'blocker_unittest.cc', + 'channel_buffer_unittest.cc', 'fir_filter_unittest.cc', 'lapped_transform_unittest.cc', 'real_fourier_unittest.cc', diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index 4eeaa59126..f7eaa4aaa7 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -112,6 +112,7 @@ if (rtc_include_tests) { # "audio_processing/agc/agc_unittest.cc", "audio_processing/agc/loudness_histogram_unittest.cc", "audio_processing/agc/mock_agc.h", + "audio_processing/audio_buffer_unittest.cc", "audio_processing/beamformer/array_util_unittest.cc", "audio_processing/beamformer/complex_matrix_unittest.cc", "audio_processing/beamformer/covariance_matrix_generator_unittest.cc", diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc index f9b9f631d2..13ece67f25 100644 --- a/webrtc/modules/audio_processing/audio_buffer.cc +++ b/webrtc/modules/audio_processing/audio_buffer.cc @@ -184,6 +184,10 @@ void AudioBuffer::InitForNewData() { reference_copied_ = false; activity_ = AudioFrame::kVadUnknown; num_channels_ = num_proc_channels_; + data_->set_num_channels(num_proc_channels_); + if (split_data_.get()) { + split_data_->set_num_channels(num_proc_channels_); + } } const int16_t* const* AudioBuffer::channels_const() const { @@ -345,6 +349,10 @@ size_t AudioBuffer::num_channels() const { void AudioBuffer::set_num_channels(size_t num_channels) { num_channels_ = num_channels; + data_->set_num_channels(num_channels); + if (split_data_.get()) { + split_data_->set_num_channels(num_channels); + } } size_t AudioBuffer::num_frames() const { diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h index d90219e913..d0e79db329 100644 --- a/webrtc/modules/audio_processing/audio_buffer.h +++ b/webrtc/modules/audio_processing/audio_buffer.h @@ -123,6 +123,8 @@ class AudioBuffer { void MergeFrequencyBands(); private: + FRIEND_TEST_ALL_PREFIXES(AudioBufferTest, + SetNumChannelsSetsChannelBuffersNumChannels); // Called from DeinterleaveFrom() and CopyFrom(). void InitForNewData(); diff --git a/webrtc/modules/audio_processing/audio_buffer_unittest.cc b/webrtc/modules/audio_processing/audio_buffer_unittest.cc new file mode 100644 index 0000000000..c6a5e1daad --- /dev/null +++ b/webrtc/modules/audio_processing/audio_buffer_unittest.cc @@ -0,0 +1,48 @@ +/* + * Copyright (c) 2016 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. + */ + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_processing/audio_buffer.h" + +namespace webrtc { + +namespace { + +const size_t kNumFrames = 480u; +const size_t kStereo = 2u; +const size_t kMono = 1u; + +void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) { + EXPECT_EQ(ab.data()->num_channels(), num_channels); + EXPECT_EQ(ab.data_f()->num_channels(), num_channels); + EXPECT_EQ(ab.split_data()->num_channels(), num_channels); + EXPECT_EQ(ab.split_data_f()->num_channels(), num_channels); + EXPECT_EQ(ab.num_channels(), num_channels); +} + +} // namespace + +TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) { + AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames); + ExpectNumChannels(ab, kStereo); + ab.set_num_channels(kMono); + ExpectNumChannels(ab, kMono); + ab.InitForNewData(); + ExpectNumChannels(ab, kStereo); +} + +#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) +TEST(AudioBufferTest, SetNumChannelsDeathTest) { + AudioBuffer ab(kNumFrames, kMono, kNumFrames, kMono, kNumFrames); + EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels"); +} +#endif + +} // namespace webrtc diff --git a/webrtc/modules/audio_processing/splitting_filter.cc b/webrtc/modules/audio_processing/splitting_filter.cc index e87a7f8621..b6f642c09d 100644 --- a/webrtc/modules/audio_processing/splitting_filter.cc +++ b/webrtc/modules/audio_processing/splitting_filter.cc @@ -72,8 +72,8 @@ void SplittingFilter::TwoBandsAnalysis(const IFChannelBuffer* data, void SplittingFilter::TwoBandsSynthesis(const IFChannelBuffer* bands, IFChannelBuffer* data) { - RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels()); - for (size_t i = 0; i < two_bands_states_.size(); ++i) { + RTC_DCHECK_LE(data->num_channels(), two_bands_states_.size()); + for (size_t i = 0; i < data->num_channels(); ++i) { WebRtcSpl_SynthesisQMF(bands->ibuf_const()->channels(0)[i], bands->ibuf_const()->channels(1)[i], bands->num_frames_per_band(), @@ -95,8 +95,8 @@ void SplittingFilter::ThreeBandsAnalysis(const IFChannelBuffer* data, void SplittingFilter::ThreeBandsSynthesis(const IFChannelBuffer* bands, IFChannelBuffer* data) { - RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels()); - for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) { + RTC_DCHECK_LE(data->num_channels(), three_band_filter_banks_.size()); + for (size_t i = 0; i < data->num_channels(); ++i) { three_band_filter_banks_[i]->Synthesis(bands->fbuf_const()->bands(i), bands->num_frames_per_band(), data->fbuf()->channels()[i]); diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 4e2a14eb35..dadc5a5e96 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -238,6 +238,7 @@ # 'audio_processing/agc/agc_unittest.cc', 'audio_processing/agc/loudness_histogram_unittest.cc', 'audio_processing/agc/mock_agc.h', + 'audio_processing/audio_buffer_unittest.cc', 'audio_processing/beamformer/array_util_unittest.cc', 'audio_processing/beamformer/complex_matrix_unittest.cc', 'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',