ACM: Adding unittests for the remixing functionality
On top of adding unittests for the remixing, the CL
moves the code tested to a separate file in order
to allow it to be tested.
Bug: webrtc:11007
Change-Id: I531736517bbcc715b3c1bf3a4256c42208c5b778
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155740
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29839}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index a4825c4..5f20c5c 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -29,6 +29,8 @@
sources = [
"acm2/acm_receiver.cc",
"acm2/acm_receiver.h",
+ "acm2/acm_remixing.cc",
+ "acm2/acm_remixing.h",
"acm2/acm_resampler.cc",
"acm2/acm_resampler.h",
"acm2/audio_coding_module.cc",
@@ -1972,6 +1974,7 @@
sources = [
"acm2/acm_receiver_unittest.cc",
+ "acm2/acm_remixing_unittest.cc",
"acm2/audio_coding_module_unittest.cc",
"acm2/call_statistics_unittest.cc",
"audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
diff --git a/modules/audio_coding/acm2/acm_remixing.cc b/modules/audio_coding/acm2/acm_remixing.cc
new file mode 100644
index 0000000..13709db
--- /dev/null
+++ b/modules/audio_coding/acm2/acm_remixing.cc
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2019 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 "modules/audio_coding/acm2/acm_remixing.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output) {
+ RTC_DCHECK_EQ(input.num_channels_, 2);
+ RTC_DCHECK_EQ(output.size(), input.samples_per_channel_);
+
+ if (input.muted()) {
+ std::fill(output.begin(), output.begin() + input.samples_per_channel_, 0);
+ } else {
+ const int16_t* const input_data = input.data();
+ for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+ output[n] = rtc::dchecked_cast<int16_t>(
+ (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+ }
+ }
+}
+
+void ReMixFrame(const AudioFrame& input,
+ size_t num_output_channels,
+ std::vector<int16_t>* output) {
+ const size_t output_size = num_output_channels * input.samples_per_channel_;
+ RTC_DCHECK(!(input.num_channels_ == 0 && num_output_channels > 0 &&
+ input.samples_per_channel_ > 0));
+
+ if (output->size() != output_size) {
+ output->resize(output_size);
+ }
+
+ // For muted frames, fill the frame with zeros.
+ if (input.muted()) {
+ std::fill(output->begin(), output->end(), 0);
+ return;
+ }
+
+ // Ensure that the special case of zero input channels is handled correctly
+ // (zero samples per channel is already handled correctly in the code below).
+ if (input.num_channels_ == 0) {
+ return;
+ }
+
+ const int16_t* const input_data = input.data();
+ size_t out_index = 0;
+
+ // When upmixing is needed and the input is mono copy the left channel
+ // into the left and right channels, and set any remaining channels to zero.
+ if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) {
+ for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+ (*output)[out_index++] = input_data[k];
+ (*output)[out_index++] = input_data[k];
+ for (size_t j = 2; j < num_output_channels; ++j) {
+ (*output)[out_index++] = 0;
+ }
+ RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+ }
+ RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+ return;
+ }
+
+ size_t in_index = 0;
+
+ // When upmixing is needed and the output is surround, copy the available
+ // channels directly, and set the remaining channels to zero.
+ if (input.num_channels_ < num_output_channels) {
+ for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+ for (size_t j = 0; j < input.num_channels_; ++j) {
+ (*output)[out_index++] = input_data[in_index++];
+ }
+ for (size_t j = input.num_channels_; j < num_output_channels; ++j) {
+ (*output)[out_index++] = 0;
+ }
+ RTC_DCHECK_EQ(in_index, (k + 1) * input.num_channels_);
+ RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+ }
+ RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_);
+ RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+
+ return;
+ }
+
+ // When downmixing is needed, and the input is stereo, average the channels.
+ if (input.num_channels_ == 2) {
+ for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+ (*output)[n] = rtc::dchecked_cast<int16_t>(
+ (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+ }
+ return;
+ }
+
+ // When downmixing is needed, and the input is multichannel, drop the surplus
+ // channels.
+ const size_t num_channels_to_drop = input.num_channels_ - num_output_channels;
+ for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+ for (size_t j = 0; j < num_output_channels; ++j) {
+ (*output)[out_index++] = input_data[in_index++];
+ }
+ in_index += num_channels_to_drop;
+ }
+}
+
+} // namespace webrtc
diff --git a/modules/audio_coding/acm2/acm_remixing.h b/modules/audio_coding/acm2/acm_remixing.h
new file mode 100644
index 0000000..661569b
--- /dev/null
+++ b/modules/audio_coding/acm2/acm_remixing.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2019 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 MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+// Stereo-to-mono downmixing. The length of the output must equal to the number
+// of samples per channel in the input.
+void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output);
+
+// Remixes the interleaved input frame to an interleaved output data vector. The
+// remixed data replaces the data in the output vector which is resized if
+// needed. The remixing supports any combination of input and output channels,
+// as well as any number of samples per channel.
+void ReMixFrame(const AudioFrame& input,
+ size_t num_output_channels,
+ std::vector<int16_t>* output);
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
diff --git a/modules/audio_coding/acm2/acm_remixing_unittest.cc b/modules/audio_coding/acm2/acm_remixing_unittest.cc
new file mode 100644
index 0000000..a1a816f
--- /dev/null
+++ b/modules/audio_coding/acm2/acm_remixing_unittest.cc
@@ -0,0 +1,191 @@
+/*
+ * Copyright (c) 2013 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 "modules/audio_coding/acm2/acm_remixing.h"
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using ::testing::AllOf;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::SizeIs;
+
+namespace webrtc {
+
+TEST(AcmRemixing, DownMixFrame) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 2;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ in_data[2 * k] = 2;
+ in_data[2 * k + 1] = 0;
+ }
+
+ DownMixFrame(in, out);
+
+ EXPECT_THAT(out, AllOf(SizeIs(480), Each(1)));
+}
+
+TEST(AcmRemixing, DownMixMutedFrame) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 2;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ in_data[2 * k] = 2;
+ in_data[2 * k + 1] = 0;
+ }
+
+ in.Mute();
+
+ DownMixFrame(in, out);
+
+ EXPECT_THAT(out, AllOf(SizeIs(480), Each(0)));
+}
+
+TEST(AcmRemixing, RemixMutedStereoFrameTo6Channels) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 2;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ in_data[2 * k] = 1;
+ in_data[2 * k + 1] = 2;
+ }
+ in.Mute();
+
+ ReMixFrame(in, 6, &out);
+ EXPECT_EQ(6 * 480u, out.size());
+
+ EXPECT_THAT(out, AllOf(SizeIs(in.samples_per_channel_ * 6), Each(0)));
+}
+
+TEST(AcmRemixing, RemixStereoFrameTo6Channels) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 2;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ in_data[2 * k] = 1;
+ in_data[2 * k + 1] = 2;
+ }
+
+ ReMixFrame(in, 6, &out);
+ EXPECT_EQ(6 * 480u, out.size());
+
+ std::vector<int16_t> expected_output(in.samples_per_channel_ * 6);
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ expected_output[6 * k] = 1;
+ expected_output[6 * k + 1] = 2;
+ }
+
+ EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+TEST(AcmRemixing, RemixMonoFrameTo6Channels) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 1;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ in_data[k] = 1;
+ }
+
+ ReMixFrame(in, 6, &out);
+ EXPECT_EQ(6 * 480u, out.size());
+
+ std::vector<int16_t> expected_output(in.samples_per_channel_ * 6, 0);
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ expected_output[6 * k] = 1;
+ expected_output[6 * k + 1] = 1;
+ }
+
+ EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+TEST(AcmRemixing, RemixStereoFrameToMono) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 2;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ in_data[2 * k] = 2;
+ in_data[2 * k + 1] = 0;
+ }
+
+ ReMixFrame(in, 1, &out);
+ EXPECT_EQ(480u, out.size());
+
+ EXPECT_THAT(out, AllOf(SizeIs(in.samples_per_channel_), Each(1)));
+}
+
+TEST(AcmRemixing, RemixMonoFrameToStereo) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 1;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ in_data[k] = 1;
+ }
+
+ ReMixFrame(in, 2, &out);
+ EXPECT_EQ(960u, out.size());
+
+ EXPECT_THAT(out, AllOf(SizeIs(2 * in.samples_per_channel_), Each(1)));
+}
+
+TEST(AcmRemixing, Remix3ChannelFrameToStereo) {
+ std::vector<int16_t> out(480, 0);
+ AudioFrame in;
+ in.num_channels_ = 3;
+ in.samples_per_channel_ = 480;
+
+ int16_t* const in_data = in.mutable_data();
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ for (size_t j = 0; j < 3; ++j) {
+ in_data[3 * k + j] = j;
+ }
+ }
+
+ ReMixFrame(in, 2, &out);
+ EXPECT_EQ(2 * 480u, out.size());
+
+ std::vector<int16_t> expected_output(in.samples_per_channel_ * 2);
+ for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+ for (size_t j = 0; j < 2; ++j) {
+ expected_output[2 * k + j] = static_cast<int>(j);
+ }
+ }
+
+ EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+} // namespace webrtc
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index efef3c0..b68579b 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -18,6 +18,7 @@
#include "absl/strings/match.h"
#include "api/array_view.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_remixing.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/include/module_common_types.h"
#include "modules/include/module_common_types_public.h"
@@ -199,110 +200,6 @@
webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
}
-// Stereo-to-mono can be used as in-place.
-void DownMix(const AudioFrame& frame,
- size_t length_out_buff,
- int16_t* out_buff) {
- RTC_DCHECK_EQ(frame.num_channels_, 2);
- RTC_DCHECK_GE(length_out_buff, frame.samples_per_channel_);
-
- if (!frame.muted()) {
- const int16_t* frame_data = frame.data();
- for (size_t n = 0; n < frame.samples_per_channel_; ++n) {
- out_buff[n] =
- static_cast<int16_t>((static_cast<int32_t>(frame_data[2 * n]) +
- static_cast<int32_t>(frame_data[2 * n + 1])) >>
- 1);
- }
- } else {
- std::fill(out_buff, out_buff + frame.samples_per_channel_, 0);
- }
-}
-
-// Remixes the input frame to an output data vector. The output vector is
-// resized if needed.
-void ReMix(const AudioFrame& input,
- size_t num_output_channels,
- std::vector<int16_t>* output) {
- const size_t output_size = num_output_channels * input.samples_per_channel_;
-
- if (output->size() != output_size) {
- output->resize(output_size);
- }
-
- // For muted frames, fill the frame with zeros.
- if (input.muted()) {
- std::fill(output->begin(), output->end(), 0);
- return;
- }
-
- // Ensure that the special case of zero input channels is handled correctly
- // (zero samples per channel is already handled correctly in the code below).
- if (input.num_channels_ == 0) {
- return;
- }
-
- const int16_t* input_data = input.data();
- size_t out_index = 0;
-
- // When upmixing is needed and the input is mono copy the left channel
- // into the left and right channels, and set any remaining channels to zero.
- if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) {
- for (size_t k = 0; k < input.samples_per_channel_; ++k) {
- (*output)[out_index++] = input_data[k];
- (*output)[out_index++] = input_data[k];
- for (size_t j = 2; j < num_output_channels; ++j) {
- (*output)[out_index++] = 0;
- }
- RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
- }
- RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
- return;
- }
-
- size_t in_index = 0;
-
- // When upmixing is needed and the output is surround, copy the available
- // channels directly, and set the remaining channels to zero.
- if (input.num_channels_ < num_output_channels) {
- for (size_t k = 0; k < input.samples_per_channel_; ++k) {
- for (size_t j = 0; j < input.num_channels_; ++j) {
- (*output)[out_index++] = input_data[in_index++];
- }
- for (size_t j = input.num_channels_; j < num_output_channels; ++j) {
- (*output)[out_index++] = 0;
- }
- RTC_DCHECK_EQ(in_index, (k + 1) * input.num_channels_);
- RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
- }
- RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_);
- RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
-
- return;
- }
-
- // When downmixing is needed, and the input is stereo, average the channels.
- if (input.num_channels_ == 2) {
- for (size_t n = 0; n < input.samples_per_channel_; ++n) {
- (*output)[n] =
- static_cast<int16_t>((static_cast<int32_t>(input_data[2 * n]) +
- static_cast<int32_t>(input_data[2 * n + 1])) >>
- 1);
- }
- return;
- }
-
- // When downmixing is needed, and the input is multichannel, drop the surplus
- // channels.
- const size_t num_channels_to_drop = input.num_channels_ - num_output_channels;
- for (size_t k = 0; k < input.samples_per_channel_; ++k) {
- for (size_t j = 0; j < num_output_channels; ++j) {
- (*output)[out_index++] = input_data[in_index++];
- }
- in_index += num_channels_to_drop;
- }
-}
-
void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
if (value != last_value_ || first_time_) {
first_time_ = false;
@@ -499,7 +396,7 @@
if (!same_num_channels) {
// Remixes the input frame to the output data and in the process resize the
// output data if needed.
- ReMix(*ptr_frame, current_num_channels, &input_data->buffer);
+ ReMixFrame(*ptr_frame, current_num_channels, &input_data->buffer);
// For pushing data to primary, point the |ptr_audio| to correct buffer.
input_data->audio = input_data->buffer.data();
@@ -567,21 +464,24 @@
*ptr_out = &preprocess_frame_;
preprocess_frame_.num_channels_ = in_frame.num_channels_;
- int16_t audio[WEBRTC_10MS_PCM_AUDIO];
+ preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
+ std::array<int16_t, WEBRTC_10MS_PCM_AUDIO> audio;
const int16_t* src_ptr_audio = in_frame.data();
if (down_mix) {
// If a resampling is required the output of a down-mix is written into a
// local buffer, otherwise, it will be written to the output frame.
int16_t* dest_ptr_audio =
- resample ? audio : preprocess_frame_.mutable_data();
- DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio);
+ resample ? audio.data() : preprocess_frame_.mutable_data();
+ RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel_);
+ DownMixFrame(in_frame,
+ rtc::ArrayView<int16_t>(
+ dest_ptr_audio, preprocess_frame_.samples_per_channel_));
preprocess_frame_.num_channels_ = 1;
// Set the input of the resampler is the down-mixed signal.
- src_ptr_audio = audio;
+ src_ptr_audio = audio.data();
}
preprocess_frame_.timestamp_ = expected_codec_ts_;
- preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
// If it is required, we have to do a resampling.
if (resample) {
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 9f026e8..ab84c78 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -1638,7 +1638,7 @@
// send surround audio.
TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForMonoInput) {
constexpr int kSampleRateHz = 48000;
- constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
+ constexpr int kSamplesPerChannel = kSampleRateHz * 10 / 1000;
audio_format_ = SdpAudioFormat({"multiopus",
kSampleRateHz,
@@ -1692,7 +1692,7 @@
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
- audio_format_ = SdpAudioFormat("opus", kSampleRateHz, 2);
+ audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 2);
RegisterCodec();