Remove dependencies on modules:module_api from AudioProcessing.
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index 864188d..6fb0ffa 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <string.h>
-
#include "api/audio/audio_frame.h"
#include "rtc_base/checks.h"
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index d69f607..39840e5 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -11,6 +11,8 @@
#ifndef API_AUDIO_AUDIO_FRAME_H_
#define API_AUDIO_AUDIO_FRAME_H_
+#include <stddef.h>
+
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)
diff --git a/api/audio/test/BUILD.gn b/api/audio/test/BUILD.gn
new file mode 100644
index 0000000..9a79778
--- /dev/null
+++ b/api/audio/test/BUILD.gn
@@ -0,0 +1,27 @@
+# Copyright (c) 2018 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+ import("//build/config/android/config.gni")
+ import("//build/config/android/rules.gni")
+}
+
+if (rtc_include_tests) {
+ rtc_source_set("audio_api_unittests") {
+ testonly = true
+ sources = [
+ "audio_frame_unittest.cc",
+ ]
+ deps = [
+ "..:audio_frame_api",
+ "../../../rtc_base:rtc_base_approved",
+ "../../../test:test_support",
+ ]
+ }
+}
diff --git a/api/audio/test/audio_frame_unittest.cc b/api/audio/test/audio_frame_unittest.cc
new file mode 100644
index 0000000..2a41e76
--- /dev/null
+++ b/api/audio/test/audio_frame_unittest.cc
@@ -0,0 +1,114 @@
+/*
+ * Copyright 2018 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 <string.h> // memcmp
+
+#include "api/audio/audio_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
+ const int16_t* frame_data = frame.data();
+ for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+ if (frame_data[i] != sample) {
+ return false;
+ }
+ }
+ return true;
+}
+
+constexpr uint32_t kTimestamp = 27;
+constexpr int kSampleRateHz = 16000;
+constexpr size_t kNumChannels = 1;
+constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
+
+} // namespace
+
+TEST(AudioFrameTest, FrameStartsMuted) {
+ AudioFrame frame;
+ EXPECT_TRUE(frame.muted());
+ EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
+ AudioFrame frame;
+ frame.mutable_data();
+ EXPECT_FALSE(frame.muted());
+ EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
+ AudioFrame frame;
+ int16_t* frame_data = frame.mutable_data();
+ for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+ frame_data[i] = 17;
+ }
+ ASSERT_TRUE(AllSamplesAre(17, frame));
+ frame.Mute();
+ EXPECT_TRUE(frame.muted());
+ EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, UpdateFrame) {
+ AudioFrame frame;
+ int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
+ frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
+ AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
+
+ EXPECT_EQ(kTimestamp, frame.timestamp_);
+ EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel_);
+ EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz_);
+ EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
+ EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
+ EXPECT_EQ(kNumChannels, frame.num_channels_);
+
+ EXPECT_FALSE(frame.muted());
+ EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
+
+ frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
+ kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+ kNumChannels);
+ EXPECT_TRUE(frame.muted());
+ EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, CopyFrom) {
+ AudioFrame frame1;
+ AudioFrame frame2;
+
+ int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
+ frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel,
+ kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+ kNumChannels);
+ frame1.CopyFrom(frame2);
+
+ EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
+ EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
+ EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
+ EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
+ EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
+ EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
+
+ EXPECT_EQ(frame2.muted(), frame1.muted());
+ EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
+
+ frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
+ kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+ kNumChannels);
+ frame1.CopyFrom(frame2);
+
+ EXPECT_EQ(frame2.muted(), frame1.muted());
+ EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
+}
+
+} // namespace webrtc