mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 13:50:40 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions

Remove dependencies on modules:module_api from AudioProcessing.

Browse source 
- 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}
This commit is contained in:
Fredrik Solenberg 2018-04-12 22:44:09 +02:00 committed by Commit Bot
parent 0ab56511f1
commit bbf21a3fd6
101 changed files with 220 additions and 214 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
BUILD.gn
View file

@ -425,6 +425,7 @@ if (rtc_include_tests) {
deps = [
":webrtc_common",
"api:rtc_api_unittests",
"api/audio/test:audio_api_unittests",
"api/audio_codecs/test:audio_codecs_api_unittests",
"p2p:libstunprober_unittests",
"p2p:rtc_p2p_unittests",

2
api/audio/audio_frame.cc
View file

@ -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"

2
api/audio/audio_frame.h
View file

@ -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)

27
api/audio/test/BUILD.gn Normal file
View file

@ -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",
]
}
}

114
api/audio/test/audio_frame_unittest.cc Normal file
View file

@ -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

4
audio/BUILD.gn
View file

@ -52,6 +52,7 @@ rtc_static_library("audio") {
"../api:optional",
"../api:transport_api",
"../api/audio:aec3_factory",
"../api/audio:audio_frame_api",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
@ -62,7 +63,6 @@ rtc_static_library("audio") {
"../common_audio:common_audio_c",
"../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/audio_coding",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:audio_network_adaptor_config",
@ -127,13 +127,13 @@ if (rtc_include_tests) {
":audio",
":audio_end_to_end_test",
"../api:mock_audio_mixer",
"../api/audio:audio_frame_api",
"../call:mock_call_interfaces",
"../call:mock_rtp_interfaces",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../common_audio",
"../logging:mocks",
"../modules:module_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing_statistics",

2
audio/audio_level.cc
View file

@ -10,8 +10,8 @@
#include "audio/audio_level.h"
#include "api/audio/audio_frame.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/include/module_common_types.h"
namespace webrtc {
namespace voe {

1
audio/audio_receive_stream.h
View file

@ -15,6 +15,7 @@
#include <vector>
#include "api/audio/audio_mixer.h"
#include "api/rtp_headers.h"
#include "audio/audio_state.h"
#include "call/audio_receive_stream.h"
#include "call/rtp_packet_sink_interface.h"

1
audio/channel.cc
View file

@ -26,7 +26,6 @@
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_payload_registry.h"

2
audio/remix_resample.cc
View file

@ -10,11 +10,11 @@
#include "audio/remix_resample.h"
#include "api/audio/audio_frame.h"
#include "audio/utility/audio_frame_operations.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

4
audio/remix_resample.h
View file

@ -11,12 +11,10 @@
#ifndef AUDIO_REMIX_RESAMPLE_H_
#define AUDIO_REMIX_RESAMPLE_H_
#include "api/audio/audio_frame.h"
#include "common_audio/resampler/include/push_resampler.h"
namespace webrtc {
class AudioFrame;
namespace voe {
// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|

2
audio/remix_resample_unittest.cc
View file

@ -12,8 +12,8 @@
#include "audio/remix_resample.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "test/gtest.h"

1
audio/transport_feedback_packet_loss_tracker.h
View file

@ -15,7 +15,6 @@
#include <vector>
#include "api/optional.h"
#include "modules/include/module_common_types.h"
namespace webrtc {

3
audio/utility/BUILD.gn
View file

@ -23,7 +23,7 @@ rtc_static_library("audio_frame_operations") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../modules:module_api",
"../../api/audio:audio_frame_api",
"../../modules/audio_coding:audio_format_conversion",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -38,7 +38,6 @@ if (rtc_include_tests) {
]
deps = [
":audio_frame_operations",
"../../modules:module_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../test:test_support",

1
audio/utility/audio_frame_operations.cc
View file

@ -12,7 +12,6 @@
#include <algorithm>
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"

3
audio/utility/audio_frame_operations.h
View file

@ -13,12 +13,11 @@
#include <stddef.h>
#include "api/audio/audio_frame.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
class AudioFrame;
// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
// Change reference parameters to pointers. Consider using a namespace rather
// than a class.

1
audio/utility/audio_frame_operations_unittest.cc
View file

@ -9,7 +9,6 @@
*/
#include "audio/utility/audio_frame_operations.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"

20
modules/audio_coding/BUILD.gn
View file

@ -97,7 +97,6 @@ rtc_source_set("audio_coding_module_typedefs") {
"include/audio_coding_module_typedefs.h",
]
deps = [
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
]
@ -136,12 +135,13 @@ rtc_static_library("audio_coding") {
}
deps = audio_coding_deps + [
"../../api/audio:audio_frame_api",
"..:module_api",
"../../common_audio:common_audio_c",
"../..:typedefs",
"../../rtc_base:deprecation",
"../../rtc_base:checks",
"../../system_wrappers:metrics_api",
"..:module_api",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
":audio_coding_module_typedefs",
@ -1086,6 +1086,7 @@ rtc_static_library("neteq") {
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../common_audio:common_audio_c",
@ -1129,11 +1130,11 @@ rtc_source_set("neteq_tools_minimal") {
deps = [
":neteq",
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
@ -1168,7 +1169,6 @@ rtc_source_set("neteq_test_tools") {
deps = [
":pcm16b",
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
@ -1220,6 +1220,7 @@ rtc_source_set("neteq_tools") {
}
deps = [
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
@ -1379,6 +1380,7 @@ if (rtc_include_tests) {
"../..:typedefs",
"../..:webrtc_common",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:rtc_base_approved",
"../../rtc_base/synchronization:rw_lock_wrapper",
@ -1438,6 +1440,7 @@ if (rtc_include_tests) {
defines = audio_coding_defines
deps = audio_coding_deps + [
"..:module_api",
":audio_coding",
":audio_format_conversion",
"../../api/audio_codecs:audio_codecs_api",
@ -1459,6 +1462,7 @@ if (rtc_include_tests) {
defines = audio_coding_defines
deps = audio_coding_deps + [
"../../api/audio:audio_frame_api",
"../../rtc_base:checks",
":audio_coding",
":neteq_tools",
@ -1490,6 +1494,7 @@ if (rtc_include_tests) {
"../..:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@ -1523,6 +1528,7 @@ if (rtc_include_tests) {
"../..:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@ -1602,7 +1608,6 @@ if (rtc_include_tests) {
testonly = true
defines = []
deps = [
"..:module_api",
"../..:typedefs",
"../../rtc_base:checks",
"../../test:fileutils",
@ -1713,9 +1718,9 @@ if (rtc_include_tests) {
":neteq",
":neteq_test_tools",
":pcm16b",
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
@ -1742,7 +1747,6 @@ if (rtc_include_tests) {
deps = [
":neteq",
":neteq_test_tools",
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
@ -1761,6 +1765,7 @@ if (rtc_include_tests) {
"../..:typedefs",
":audio_coding",
":neteq_input_audio_tools",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs/g711:audio_encoder_g711",
"../../api/audio_codecs/L16:audio_encoder_L16",
"../../api/audio_codecs/g722:audio_encoder_g722",
@ -2222,6 +2227,7 @@ if (rtc_include_tests) {
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",

1
modules/audio_coding/acm2/acm_receive_test.cc
View file

@ -21,6 +21,7 @@
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/include/module_common_types.h"
#include "test/gtest.h"
namespace webrtc {

1
modules/audio_coding/acm2/acm_receiver.cc
View file

@ -22,6 +22,7 @@
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"

2
modules/audio_coding/acm2/acm_receiver.h
View file

@ -16,6 +16,7 @@
#include <string>
#include <vector>
#include "api/audio/audio_frame.h"
#include "api/array_view.h"
#include "api/optional.h"
#include "common_audio/vad/include/webrtc_vad.h"
@ -23,7 +24,6 @@
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_coding/acm2/acm_receiver_unittest.cc
View file

@ -17,6 +17,7 @@
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "system_wrappers/include/clock.h"

1
modules/audio_coding/acm2/acm_send_test.h
View file

@ -14,6 +14,7 @@
#include <memory>
#include <vector>
#include "api/audio/audio_frame.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "rtc_base/constructormagic.h"

1
modules/audio_coding/acm2/audio_coding_module.cc
View file

@ -16,6 +16,7 @@
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/codec_manager.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"

1
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View file

@ -32,7 +32,6 @@
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/messagedigest.h"
#include "rtc_base/numerics/safe_conversions.h"

2
modules/audio_coding/acm2/call_statistics.h
View file

@ -11,8 +11,8 @@
#ifndef MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
#define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
#include "api/audio/audio_frame.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types.h"
//
// This class is for book keeping of calls to ACM. It is not useful to log API

1
modules/audio_coding/acm2/rent_a_codec_unittest.cc
View file

@ -10,6 +10,7 @@
#include <memory>
#include "common_types.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "rtc_base/arraysize.h"
#include "test/gtest.h"

9
modules/audio_coding/include/audio_coding_module.h
View file

@ -21,7 +21,6 @@
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/include/module.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/function_view.h"
#include "system_wrappers/include/clock.h"
@ -278,9 +277,7 @@ class AudioCodingModule {
//
// Input:
// -audio_frame : the input audio frame, containing raw audio
// sampling frequency etc.,
// c.f. module_common_types.h for definition of
// AudioFrame.
// sampling frequency etc.
//
// Return value:
// >= 0 number of bytes encoded.
@ -663,9 +660,7 @@ class AudioCodingModule {
//
// Output:
// -audio_frame : output audio frame which contains raw audio data
// and other relevant parameters, c.f.
// module_common_types.h for the definition of
// AudioFrame.
// and other relevant parameters.
// -muted : if true, the sample data in audio_frame is not
// populated, and must be interpreted as all zero.
//

1
modules/audio_coding/include/audio_coding_module_typedefs.h
View file

@ -13,7 +13,6 @@
#include <map>
#include "modules/include/module_common_types.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {

1
modules/audio_coding/neteq/decision_logic.cc
View file

@ -19,6 +19,7 @@
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/include/module_common_types.h"
namespace webrtc {

1
modules/audio_coding/neteq/decision_logic_normal.cc
View file

@ -20,7 +20,6 @@
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/include/module_common_types.h"
namespace webrtc {

1
modules/audio_coding/neteq/nack_tracker.cc
View file

@ -14,7 +14,6 @@
#include <algorithm> // For std::max.
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {

1
modules/audio_coding/neteq/nack_tracker.h
View file

@ -15,6 +15,7 @@
#include <map>
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/gtest_prod_util.h"
//

2
modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
View file

@ -12,13 +12,13 @@
#include <memory>
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "modules/include/module_common_types.h"
#include "test/gmock.h"
#include "test/testsupport/fileutils.h"

1
modules/audio_coding/neteq/neteq_impl.cc
View file

@ -41,7 +41,6 @@
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/timestamp_scaler.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"

2
modules/audio_coding/neteq/neteq_impl.h
View file

@ -15,6 +15,7 @@
#include <string>
#include "api/optional.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/defines.h"
#include "modules/audio_coding/neteq/expand_uma_logger.h"
@ -24,7 +25,6 @@
#include "modules/audio_coding/neteq/rtcp.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"

1
modules/audio_coding/neteq/neteq_impl_unittest.cc
View file

@ -27,7 +27,6 @@
#include "modules/audio_coding/neteq/preemptive_expand.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/audio_coding/neteq/timestamp_scaler.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gmock.h"
#include "test/gtest.h"

2
modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View file

@ -10,10 +10,10 @@
#include <memory>
#include "api/audio/audio_frame.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "modules/include/module_common_types.h"
#include "test/gmock.h"
namespace webrtc {

2
modules/audio_coding/neteq/neteq_stereo_unittest.cc
View file

@ -15,13 +15,13 @@
#include <string>
#include <list>
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "modules/include/module_common_types.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"

2
modules/audio_coding/neteq/neteq_unittest.cc
View file

@ -20,12 +20,12 @@
#include <string>
#include <vector>
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/messagedigest.h"
#include "rtc_base/numerics/safe_conversions.h"

2
modules/audio_coding/neteq/rtcp.cc
View file

@ -15,8 +15,6 @@
#include <algorithm>
#include "modules/include/module_common_types.h"
namespace webrtc {
void Rtcp::Init(uint16_t start_sequence_number) {

2
modules/audio_coding/neteq/sync_buffer.h
View file

@ -11,8 +11,8 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
#include "api/audio/audio_frame.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)

2
modules/audio_coding/neteq/tools/audio_sink.h
View file

@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
#include "modules/include/module_common_types.h"
#include "api/audio/audio_frame.h"
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_coding/neteq/tools/encode_neteq_input.h
View file

@ -15,7 +15,6 @@
#include "api/audio_codecs/audio_encoder.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
#include "modules/include/module_common_types.h"
namespace webrtc {
namespace test {

1
modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
View file

@ -17,6 +17,7 @@
#include <limits>
#include <utility>
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {

1
modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
View file

@ -11,6 +11,7 @@
#include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "rtc_base/format_macros.h"
#include "test/gtest.h"

1
modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
View file

@ -17,7 +17,6 @@
#include "api/audio_codecs/audio_decoder.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/include/module_common_types.h"
namespace webrtc {
namespace test {

7
modules/audio_coding/neteq/tools/neteq_performance_test.cc
View file

@ -10,13 +10,13 @@
#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/clock.h"
#include "test/testsupport/fileutils.h"
@ -103,7 +103,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
return -1;
payload_len = WebRtcPcm16b_Encode(input_samples.data(),
input_samples.size(), input_payload);
assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));
}
// Get output audio, but don't do anything with it.
@ -113,8 +113,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
if (error != NetEq::kOK)
return -1;
assert(out_frame.samples_per_channel_ ==
static_cast<size_t>(kSampRateHz * 10 / 1000));
RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);
static const int kOutputBlockSizeMs = 10;
time_now_ms += kOutputBlockSizeMs;

1
modules/audio_coding/neteq/tools/neteq_quality_test.h
View file

@ -19,7 +19,6 @@
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/flags.h"
#include "test/gtest.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_coding/neteq/tools/neteq_rtpplay.cc
View file

@ -32,7 +32,6 @@
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "test/testsupport/fileutils.h"

1
modules/audio_coding/neteq/tools/packet.cc
View file

@ -14,7 +14,6 @@
#include <memory>
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "rtc_base/checks.h"

1
modules/audio_coding/neteq/tools/rtp_encode.cc
View file

@ -21,6 +21,7 @@
#include <map>
#include <string>
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h"
#include "api/audio_codecs/g722/audio_encoder_g722.h"

1
modules/audio_coding/test/EncodeDecodeTest.h
View file

@ -18,6 +18,7 @@
#include "modules/audio_coding/test/ACMTest.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/audio_coding/test/RTPFile.h"
#include "modules/include/module_common_types.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {

1
modules/audio_coding/test/PCMFile.cc
View file

@ -14,7 +14,6 @@
#include <stdio.h>
#include <string.h>
#include "modules/include/module_common_types.h"
#include "test/gtest.h"
namespace webrtc {

2
modules/audio_coding/test/PCMFile.h
View file

@ -16,8 +16,8 @@
#include <string>
#include "api/audio/audio_frame.h"
#include "api/optional.h"
#include "modules/include/module_common_types.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {

3
modules/audio_coding/test/RTPFile.cc
View file

@ -19,8 +19,7 @@
# include <arpa/inet.h>
#endif
#include "audio_coding_module.h"
#include "rtc_base/synchronization/rw_lock_wrapper.h"
#include "modules/include/module_common_types.h"
// TODO(tlegrand): Consider removing usage of gtest.
#include "test/gtest.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_coding/test/RTPFile.h
View file

@ -15,7 +15,6 @@
#include <queue>
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/synchronization/rw_lock_wrapper.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_coding/test/insert_packet_with_timing.cc
View file

@ -19,7 +19,6 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/flags.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"

1
modules/audio_coding/test/target_delay_unittest.cc
View file

@ -10,6 +10,7 @@
#include <memory>
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"

1
modules/audio_device/BUILD.gn
View file

@ -177,7 +177,6 @@ rtc_source_set("audio_device_impl") {
":audio_device_api",
":audio_device_buffer",
":audio_device_generic",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",

5
modules/audio_mixer/BUILD.gn
View file

@ -36,7 +36,6 @@ rtc_static_library("audio_mixer_impl") {
deps = [
":audio_frame_manipulator",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
@ -67,7 +66,7 @@ rtc_static_library("audio_frame_manipulator") {
]
deps = [
"..:module_api",
"../../api/audio:audio_frame_api",
"../../audio/utility:audio_frame_operations",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -91,8 +90,8 @@ if (rtc_include_tests) {
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
"..:module_api",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../api/audio:audio_mixer_api",
"../../audio/utility:audio_frame_operations",
"../../rtc_base:checks",

1
modules/audio_mixer/audio_frame_manipulator.cc
View file

@ -10,7 +10,6 @@
#include "modules/audio_mixer/audio_frame_manipulator.h"
#include "audio/utility/audio_frame_operations.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {

2
modules/audio_mixer/audio_frame_manipulator.h
View file

@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
#define MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
#include "modules/include/module_common_types.h"
#include "api/audio/audio_frame.h"
namespace webrtc {

1
modules/audio_mixer/audio_frame_manipulator_unittest.cc
View file

@ -11,7 +11,6 @@
#include <algorithm>
#include "modules/audio_mixer/audio_frame_manipulator.h"
#include "modules/include/module_common_types.h"
#include "test/gtest.h"
namespace webrtc {

1
modules/audio_mixer/audio_mixer_impl.h
View file

@ -18,7 +18,6 @@
#include "modules/audio_mixer/frame_combiner.h"
#include "modules/audio_mixer/output_rate_calculator.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread_annotations.h"

1
modules/audio_mixer/frame_combiner.h
View file

@ -16,7 +16,6 @@
#include "modules/audio_processing/agc2/fixed_gain_controller.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
namespace webrtc {
class ApmDataDumper;

2
modules/audio_mixer/sine_wave_generator.h
View file

@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
#define MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
#include "modules/include/module_common_types.h"
#include "api/audio/audio_frame.h"
#include "rtc_base/checks.h"
namespace webrtc {

12
modules/audio_processing/BUILD.gn
View file

@ -122,12 +122,12 @@ rtc_static_library("audio_processing") {
":audio_generator_interface",
":audio_processing_c",
":audio_processing_statistics",
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
"../../api:optional",
"../../api/audio:aec3_config",
"../../api/audio:audio_frame_api",
"../../api/audio:echo_control",
"../../audio/utility:audio_frame_operations",
"../../common_audio:common_audio_c",
@ -214,6 +214,7 @@ rtc_source_set("aec_dump_interface") {
deps = [
":audio_frame_view",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../rtc_base:rtc_base_approved",
]
}
@ -526,7 +527,6 @@ if (rtc_include_tests) {
":audioproc_test_utils",
":file_audio_generator_unittests",
":mocks",
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
@ -581,6 +581,7 @@ if (rtc_include_tests) {
":audioproc_protobuf_utils",
":audioproc_test_utils",
":audioproc_unittest_proto",
"../../api/audio:audio_frame_api",
"../../rtc_base:rtc_task_queue",
"aec_dump",
"aec_dump:aec_dump_unittests",
@ -629,7 +630,6 @@ if (rtc_include_tests) {
":audio_processing",
":audioproc_test_utils",
"../../api:array_view",
"../../modules:module_api",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@ -668,8 +668,9 @@ if (rtc_include_tests) {
]
deps = [
"../../api:array_view",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../common_audio:common_audio",
"../../modules:module_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
@ -743,9 +744,9 @@ if (rtc_include_tests) {
deps = [
":audio_processing",
"..:module_api",
"../../api:array_view",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -766,7 +767,6 @@ if (rtc_include_tests) {
]
deps = [
":audio_processing",
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../common_audio:common_audio",

4
modules/audio_processing/aec_dump/BUILD.gn
View file

@ -29,7 +29,6 @@ rtc_source_set("mock_aec_dump") {
deps = [
"..:aec_dump_interface",
"../..:module_api",
"../../../test:test_support",
]
}
@ -63,7 +62,7 @@ if (rtc_enable_protobuf) {
deps = [
":aec_dump",
"..:aec_dump_interface",
"../../../modules:module_api",
"../../../api/audio:audio_frame_api",
"../../../rtc_base:checks",
"../../../rtc_base:protobuf_utils",
"../../../rtc_base:rtc_base_approved",
@ -83,7 +82,6 @@ if (rtc_enable_protobuf) {
":aec_dump_impl",
"..:aec_dump_interface",
"..:audioproc_debug_proto",
"../../../modules:module_api",
"../../../rtc_base:rtc_task_queue",
"../../../test:fileutils",
"../../../test:test_support",

2
modules/audio_processing/aec_dump/aec_dump_impl.h
View file

@ -15,10 +15,10 @@
#include <string>
#include <vector>
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/aec_dump/capture_stream_info.h"
#include "modules/audio_processing/aec_dump/write_to_file_task.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/race_checker.h"

1
modules/audio_processing/aec_dump/aec_dump_unittest.cc
View file

@ -12,7 +12,6 @@
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/task_queue.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"

2
modules/audio_processing/aec_dump/capture_stream_info.h
View file

@ -15,9 +15,9 @@
#include <utility>
#include <vector>
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/aec_dump/write_to_file_task.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/logging.h"

1
modules/audio_processing/aec_dump/mock_aec_dump.h
View file

@ -14,7 +14,6 @@
#include <memory>
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/include/module_common_types.h"
#include "test/gmock.h"
namespace webrtc {

1
modules/audio_processing/agc/agc.cc
View file

@ -18,7 +18,6 @@
#include "modules/audio_processing/agc/loudness_histogram.h"
#include "modules/audio_processing/agc/utility.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {

1
modules/audio_processing/agc/agc_manager_direct.cc
View file

@ -18,7 +18,6 @@
#include "modules/audio_processing/agc/gain_map_internal.h"
#include "modules/audio_processing/gain_control_impl.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"

1
modules/audio_processing/agc/loudness_histogram.cc
View file

@ -13,7 +13,6 @@
#include <cmath>
#include <cstring>
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {

1
modules/audio_processing/agc/mock_agc.h
View file

@ -13,7 +13,6 @@
#include "modules/audio_processing/agc/agc.h"
#include "modules/include/module_common_types.h"
#include "test/gmock.h"
namespace webrtc {

2
modules/audio_processing/audio_buffer.h
View file

@ -14,10 +14,10 @@
#include <memory>
#include <vector>
#include "api/audio/audio_frame.h"
#include "common_audio/channel_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/splitting_filter.h"
#include "modules/include/module_common_types.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {

5
modules/audio_processing/audio_processing_impl.cc
View file

@ -43,7 +43,6 @@
#include "modules/audio_processing/residual_echo_detector.h"
#include "modules/audio_processing/transient/transient_suppressor.h"
#include "modules/audio_processing/voice_detection_impl.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/system/file_wrapper.h"
#include "system_wrappers/include/metrics.h"
@ -1375,8 +1374,8 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
processing_config.reverse_output_stream() = output_config;
RETURN_ON_ERR(MaybeInitializeRender(processing_config));
assert(input_config.num_frames() ==
formats_.api_format.reverse_input_stream().num_frames());
RTC_DCHECK_EQ(input_config.num_frames(),
formats_.api_format.reverse_input_stream().num_frames());
if (aec_dump_) {
const size_t channel_size =

1
modules/audio_processing/audio_processing_impl_locking_unittest.cc
View file

@ -16,7 +16,6 @@
#include "api/array_view.h"
#include "modules/audio_processing/test/test_utils.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/event.h"
#include "rtc_base/platform_thread.h"

1
modules/audio_processing/audio_processing_impl_unittest.cc
View file

@ -11,7 +11,6 @@
#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/test/test_utils.h"
#include "modules/include/module_common_types.h"
#include "test/gmock.h"
#include "test/gtest.h"

1
modules/audio_processing/audio_processing_performance_unittest.cc
View file

@ -17,7 +17,6 @@
#include "api/array_view.h"
#include "modules/audio_processing/test/test_utils.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/platform_thread.h"

1
modules/audio_processing/audio_processing_unittest.cc
View file

@ -27,7 +27,6 @@
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/audio_processing/test/protobuf_utils.h"
#include "modules/audio_processing/test/test_utils.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/gtest_prod_util.h"

3
modules/audio_processing/include/aec_dump.h
View file

@ -16,12 +16,11 @@
#include <vector>
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class AudioFrame;
// Struct for passing current config from APM without having to
// include protobuf headers.
struct InternalAPMConfig {

1
modules/audio_processing/test/aec_dump_based_simulator.cc
View file

@ -14,6 +14,7 @@
#include "modules/audio_processing/test/protobuf_utils.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
namespace test {

1
modules/audio_processing/test/fake_recording_device.cc
View file

@ -12,6 +12,7 @@
#include <algorithm>
#include "api/optional.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"

2
modules/audio_processing/test/fake_recording_device.h
View file

@ -15,9 +15,9 @@
#include <memory>
#include <vector>
#include "api/audio/audio_frame.h"
#include "api/array_view.h"
#include "common_audio/channel_buffer.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {

2
modules/audio_processing/test/test_utils.h
View file

@ -18,10 +18,10 @@
#include <string>
#include <vector>
#include "api/audio/audio_frame.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/wav_file.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {

1
modules/audio_processing/transient/transient_suppression_test.cc
View file

@ -19,7 +19,6 @@
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc/agc.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/flags.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"

1
modules/audio_processing/typing_detection.h
View file

@ -11,7 +11,6 @@
#ifndef MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
#define MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
#include "modules/include/module_common_types.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {

2
modules/audio_processing/vad/BUILD.gn
View file

@ -35,7 +35,6 @@ rtc_static_library("vad") {
"voice_gmm_tables.h",
]
deps = [
"../..:module_api",
"../../..:typedefs",
"../../../audio/utility:audio_frame_operations",
"../../../common_audio",
@ -70,7 +69,6 @@ if (rtc_include_tests) {
]
deps = [
":vad",
"../..:module_api",
"../../../common_audio",
"../../../test:fileutils",
"../../../test:test_support",

1
modules/audio_processing/vad/pitch_based_vad.cc
View file

@ -17,7 +17,6 @@
#include "modules/audio_processing/vad/common.h"
#include "modules/audio_processing/vad/noise_gmm_tables.h"
#include "modules/audio_processing/vad/voice_gmm_tables.h"
#include "modules/include/module_common_types.h"
namespace webrtc {

1
modules/audio_processing/vad/standalone_vad.cc
View file

@ -11,7 +11,6 @@
#include "modules/audio_processing/vad/standalone_vad.h"
#include "audio/utility/audio_frame_operations.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_processing/vad/standalone_vad_unittest.cc
View file

@ -14,7 +14,6 @@
#include <memory>
#include "modules/include/module_common_types.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"

1
modules/audio_processing/vad/vad_audio_proc.cc
View file

@ -24,7 +24,6 @@ extern "C" {
#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
}
#include "modules/include/module_common_types.h"
namespace webrtc {

1
modules/audio_processing/vad/vad_audio_proc_unittest.cc
View file

@ -20,7 +20,6 @@
#include <string>
#include "modules/audio_processing/vad/common.h"
#include "modules/include/module_common_types.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"

98
modules/module_common_types_unittest.cc
View file

@ -10,108 +10,10 @@
#include "modules/include/module_common_types.h"
#include <string.h> // memcmp
#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)));
}
TEST(IsNewerSequenceNumber, Equal) {
EXPECT_FALSE(IsNewerSequenceNumber(0x0001, 0x0001));
}

2
rtc_tools/BUILD.gn
View file

@ -290,7 +290,7 @@ if (rtc_include_tests) {
}
deps = [
"../modules:module_api",
"../api/audio:audio_frame_api",
"../modules/audio_processing",
"../modules/audio_processing/vad",
"../rtc_base:rtc_base_approved",

2
rtc_tools/agc/activity_metric.cc
View file

@ -16,6 +16,7 @@
#include <algorithm>
#include <memory>
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/agc/agc.h"
#include "modules/audio_processing/agc/loudness_histogram.h"
#include "modules/audio_processing/agc/utility.h"
@ -23,7 +24,6 @@
#include "modules/audio_processing/vad/pitch_based_vad.h"
#include "modules/audio_processing/vad/standalone_vad.h"
#include "modules/audio_processing/vad/vad_audio_proc.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/flags.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "test/gtest.h"

2
test/fuzzers/BUILD.gn
View file

@ -430,7 +430,7 @@ rtc_static_library("audio_processing_fuzzer_helper") {
deps = [
":fuzz_data_helper",
"../../api:optional",
"../../modules:module_api",
"../../api/audio:audio_frame_api",
"../../modules/audio_processing",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",

Some files were not shown because too many files have changed in this diff Show more