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modules: more interface -> include renames

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This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
This commit is contained in:
Henrik Kjellander 2015-11-04 08:31:52 +01:00
parent 5af9a28bd6
commit ff761fba82
322 changed files with 4176 additions and 507 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/audio/audio_receive_stream.h
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@ -14,7 +14,7 @@
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/voice_engine/include/voe_base.h"
namespace webrtc {

4
webrtc/call/call.cc
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@ -25,9 +25,9 @@
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/config.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/utility/interface/process_thread.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"

2
webrtc/call/call_perf_tests.cc
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@ -19,7 +19,7 @@
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/rtp_to_ntp.h"

4
webrtc/call/congestion_controller.cc
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@ -20,8 +20,8 @@
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/process_thread.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/video_engine/call_stats.h"

2
webrtc/common_video/video_render_frames.cc
View File

@ -12,7 +12,7 @@
#include <assert.h>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"

2
webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
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@ -18,7 +18,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/examples/android/media_demo/jni/jni_helpers.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
#include "webrtc/modules/utility/include/helpers_android.h"
#include "webrtc/test/channel_transport/include/channel_transport.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_base.h"

2
webrtc/modules/audio_coding/BUILD.gn
View File

@ -45,7 +45,7 @@ source_set("rent_a_codec") {
config("audio_coding_config") {
include_dirs = [
"main/include",
"../interface",
"../include",
]
}

2
webrtc/modules/audio_coding/main/acm2/acm_receiver.h
View File

@ -24,7 +24,7 @@
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
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@ -33,7 +33,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"

2
webrtc/modules/audio_coding/main/acm2/call_statistics.h
View File

@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
//
// This class is for book keeping of calls to ACM. It is not useful to log API

2
webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
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@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

4
webrtc/modules/audio_coding/main/audio_coding_module.gypi
View File

@ -81,13 +81,13 @@
],
'include_dirs': [
'include',
'../../interface',
'../../include',
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
'../../interface',
'../../include',
'<(webrtc_root)',
],
},

2
webrtc/modules/audio_coding/main/include/audio_coding_module.h
View File

@ -18,7 +18,7 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
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@ -13,7 +13,7 @@
#include <map>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/test/Channel.h
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@ -14,7 +14,7 @@
#include <stdio.h>
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/test/PCMFile.cc
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@ -15,7 +15,7 @@
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/test/PCMFile.h
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@ -16,7 +16,7 @@
#include <string>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/test/RTPFile.h
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@ -15,7 +15,7 @@
#include <queue>
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
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@ -17,7 +17,7 @@
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/testsupport/fileutils.h"

2
webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
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@ -14,7 +14,7 @@
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"

2
webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
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@ -20,7 +20,7 @@
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/delay_manager.cc
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@ -17,7 +17,7 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/nack.cc
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@ -15,7 +15,7 @@
#include <algorithm> // For std::max.
#include "webrtc/base/checks.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/neteq_impl.cc
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@ -42,7 +42,7 @@
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
// Modify the code to obtain backwards bit-exactness. Once bit-exactness is no

2
webrtc/modules/audio_coding/neteq/packet.h
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@ -13,7 +13,7 @@
#include <list>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/rtcp.cc
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@ -15,7 +15,7 @@
#include <algorithm>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
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@ -14,7 +14,7 @@
#include <map>
#include <stdio.h>
#include "webrtc/typedefs.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
enum stereoModes {
stereoModeMono,

2
webrtc/modules/audio_coding/neteq/tools/audio_sink.h
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@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
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@ -14,7 +14,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace test {

2
webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
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@ -34,7 +34,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/rtp_file_reader.h"
#include "webrtc/test/testsupport/fileutils.h"

4
webrtc/modules/audio_coding/neteq/tools/packet.cc
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@ -12,8 +12,8 @@
#include <string.h>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
namespace webrtc {
namespace test {

2
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
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@ -18,7 +18,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD

2
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
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@ -16,7 +16,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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@ -20,7 +20,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/test/rtp_file_reader.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
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@ -18,7 +18,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
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@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

8
webrtc/modules/audio_conference_mixer/BUILD.gn
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@ -9,15 +9,15 @@
config("audio_conference_mixer_config") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
include_dirs = [
"interface",
"../interface",
"include",
"../include",
]
}
source_set("audio_conference_mixer") {
sources = [
"interface/audio_conference_mixer.h",
"interface/audio_conference_mixer_defines.h",
"include/audio_conference_mixer.h",
"include/audio_conference_mixer_defines.h",
"source/audio_conference_mixer_impl.cc",
"source/audio_conference_mixer_impl.h",
"source/audio_frame_manipulator.cc",

4
webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
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@ -17,8 +17,8 @@
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'interface/audio_conference_mixer.h',
'interface/audio_conference_mixer_defines.h',
'include/audio_conference_mixer.h',
'include/audio_conference_mixer_defines.h',
'source/audio_frame_manipulator.cc',
'source/audio_frame_manipulator.h',
'source/memory_pool.h',

77
webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h Normal file
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@ -0,0 +1,77 @@
/*
* Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
class AudioMixerOutputReceiver;
class MixerParticipant;
class Trace;
class AudioConferenceMixer : public Module
{
public:
enum {kMaximumAmountOfMixedParticipants = 3};
enum Frequency
{
kNbInHz = 8000,
kWbInHz = 16000,
kSwbInHz = 32000,
kFbInHz = 48000,
kLowestPossible = -1,
kDefaultFrequency = kWbInHz
};
// Factory method. Constructor disabled.
static AudioConferenceMixer* Create(int id);
virtual ~AudioConferenceMixer() {}
// Module functions
int64_t TimeUntilNextProcess() override = 0;
int32_t Process() override = 0;
// Register/unregister a callback class for receiving the mixed audio.
virtual int32_t RegisterMixedStreamCallback(
AudioMixerOutputReceiver* receiver) = 0;
virtual int32_t UnRegisterMixedStreamCallback() = 0;
// Add/remove participants as candidates for mixing.
virtual int32_t SetMixabilityStatus(MixerParticipant* participant,
bool mixable) = 0;
// Returns true if a participant is a candidate for mixing.
virtual bool MixabilityStatus(
const MixerParticipant& participant) const = 0;
// Inform the mixer that the participant should always be mixed and not
// count toward the number of mixed participants. Note that a participant
// must have been added to the mixer (by calling SetMixabilityStatus())
// before this function can be successfully called.
virtual int32_t SetAnonymousMixabilityStatus(
MixerParticipant* participant, bool mixable) = 0;
// Returns true if the participant is mixed anonymously.
virtual bool AnonymousMixabilityStatus(
const MixerParticipant& participant) const = 0;
// Set the minimum sampling frequency at which to mix. The mixing algorithm
// may still choose to mix at a higher samling frequency to avoid
// downsampling of audio contributing to the mixed audio.
virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0;
protected:
AudioConferenceMixer() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_

60
webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h Normal file
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@ -0,0 +1,60 @@
/*
* Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class MixHistory;
// A callback class that all mixer participants must inherit from/implement.
class MixerParticipant
{
public:
// The implementation of this function should update audioFrame with new
// audio every time it's called.
//
// If it returns -1, the frame will not be added to the mix.
virtual int32_t GetAudioFrame(int32_t id,
AudioFrame* audioFrame) = 0;
// Returns true if the participant was mixed this mix iteration.
bool IsMixed() const;
// This function specifies the sampling frequency needed for the AudioFrame
// for future GetAudioFrame(..) calls.
virtual int32_t NeededFrequency(int32_t id) const = 0;
MixHistory* _mixHistory;
protected:
MixerParticipant();
virtual ~MixerParticipant();
};
class AudioMixerOutputReceiver
{
public:
// This callback function provides the mixed audio for this mix iteration.
// Note that uniqueAudioFrames is an array of AudioFrame pointers with the
// size according to the size parameter.
virtual void NewMixedAudio(const int32_t id,
const AudioFrame& generalAudioFrame,
const AudioFrame** uniqueAudioFrames,
const uint32_t size) = 0;
protected:
AudioMixerOutputReceiver() {}
virtual ~AudioMixerOutputReceiver() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_

14
webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
View File

@ -8,12 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/interface/module_common_types.h"
#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
class AudioMixerOutputReceiver;
@ -74,4 +76,4 @@ protected:
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_

10
webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
View File

@ -8,10 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#include "webrtc/modules/interface/module_common_types.h"
#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -57,4 +59,4 @@ protected:
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_

4
webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
View File

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/utility/interface/audio_frame_operations.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"

4
webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
View File

@ -16,10 +16,10 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
class AudioProcessing;

2
webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
View File

@ -9,7 +9,7 @@
*/
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace {

4
webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
View File

@ -10,8 +10,8 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
namespace webrtc {

2
webrtc/modules/audio_device/BUILD.gn
View File

@ -10,7 +10,7 @@ import("../../build/webrtc.gni")
config("audio_device_config") {
include_dirs = [
"../interface",
"../include",
"include",
"dummy", # Contains dummy audio device implementations.
]

2
webrtc/modules/audio_device/android/audio_manager.cc
View File

@ -16,7 +16,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_device/android/audio_common.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
#include "webrtc/modules/utility/include/helpers_android.h"
#define TAG "AudioManager"
#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)

4
webrtc/modules/audio_device/android/audio_manager.h
View File

@ -19,8 +19,8 @@
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
#include "webrtc/modules/utility/interface/jvm_android.h"
#include "webrtc/modules/utility/include/helpers_android.h"
#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {

4
webrtc/modules/audio_device/android/audio_record_jni.h
View File

@ -17,8 +17,8 @@
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
#include "webrtc/modules/utility/interface/jvm_android.h"
#include "webrtc/modules/utility/include/helpers_android.h"
#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {

4
webrtc/modules/audio_device/android/audio_track_jni.h
View File

@ -18,8 +18,8 @@
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
#include "webrtc/modules/utility/interface/jvm_android.h"
#include "webrtc/modules/utility/include/helpers_android.h"
#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {

2
webrtc/modules/audio_device/android/build_info.cc
View File

@ -10,7 +10,7 @@
#include "webrtc/modules/audio_device/android/build_info.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
#include "webrtc/modules/utility/include/helpers_android.h"
namespace webrtc {

2
webrtc/modules/audio_device/android/build_info.h
View File

@ -14,7 +14,7 @@
#include <jni.h>
#include <string>
#include "webrtc/modules/utility/interface/jvm_android.h"
#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {

2
webrtc/modules/audio_device/android/ensure_initialized.cc
View File

@ -18,7 +18,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_device/android/audio_record_jni.h"
#include "webrtc/modules/audio_device/android/audio_track_jni.h"
#include "webrtc/modules/utility/interface/jvm_android.h"
#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {
namespace audiodevicemodule {

2
webrtc/modules/audio_device/android/opensles_player.h
View File

@ -22,7 +22,7 @@
#include "webrtc/modules/audio_device/android/opensles_common.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
#include "webrtc/modules/utility/include/helpers_android.h"
namespace webrtc {

4
webrtc/modules/audio_device/audio_device.gypi
View File

@ -20,13 +20,13 @@
],
'include_dirs': [
'.',
'../interface',
'../include',
'include',
'dummy', # Contains dummy audio device implementations.
],
'direct_dependent_settings': {
'include_dirs': [
'../interface',
'../include',
'include',
],
},

2
webrtc/modules/audio_device/include/audio_device.h
View File

@ -12,7 +12,7 @@
#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/include/module.h"
namespace webrtc {

2
webrtc/modules/audio_device/ios/audio_device_ios.mm
View File

@ -21,7 +21,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_device/fine_audio_buffer.h"
#include "webrtc/modules/utility/interface/helpers_ios.h"
#include "webrtc/modules/utility/include/helpers_ios.h"
namespace webrtc {

2
webrtc/modules/audio_device/test/audio_device_test_defines.h
View File

@ -13,7 +13,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/utility/interface/process_thread.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/trace.h"
#ifdef _WIN32

2
webrtc/modules/audio_processing/agc/agc.cc
View File

@ -19,7 +19,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/agc/histogram.h"
#include "webrtc/modules/audio_processing/agc/utility.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace {

2
webrtc/modules/audio_processing/agc/agc_manager_direct.cc
View File

@ -19,7 +19,7 @@
#include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
#include "webrtc/modules/audio_processing/gain_control_impl.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {

2
webrtc/modules/audio_processing/agc/agc_unittest.cc
View File

@ -13,7 +13,7 @@
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/tools/agc/test_utils.h"

2
webrtc/modules/audio_processing/agc/histogram.cc
View File

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

2
webrtc/modules/audio_processing/agc/mock_agc.h
View File

@ -14,7 +14,7 @@
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "gmock/gmock.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

2
webrtc/modules/audio_processing/audio_buffer.h
View File

@ -15,7 +15,7 @@
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/splitting_filter.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_processing/audio_processing_impl.cc
View File

@ -36,7 +36,7 @@ extern "C" {
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
#include "webrtc/modules/audio_processing/voice_detection_impl.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"

2
webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
View File

@ -14,7 +14,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
using ::testing::Invoke;
using ::testing::Return;

2
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
View File

@ -24,7 +24,7 @@
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"

2
webrtc/modules/audio_processing/test/process_test.cc
View File

@ -22,7 +22,7 @@
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/fileutils.h"

2
webrtc/modules/audio_processing/test/test_utils.h
View File

@ -22,7 +22,7 @@
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

2
webrtc/modules/audio_processing/transient/transient_suppression_test.cc
View File

@ -19,7 +19,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_processing/typing_detection.h
View File

@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_processing/vad/pitch_based_vad.cc
View File

@ -18,7 +18,7 @@
#include "webrtc/modules/audio_processing/vad/common.h"
#include "webrtc/modules/audio_processing/vad/noise_gmm_tables.h"
#include "webrtc/modules/audio_processing/vad/voice_gmm_tables.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

4
webrtc/modules/audio_processing/vad/standalone_vad.cc
View File

@ -12,8 +12,8 @@
#include <assert.h>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/utility/interface/audio_frame_operations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
View File

@ -14,7 +14,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"

2
webrtc/modules/audio_processing/vad/vad_audio_proc.cc
View File

@ -23,7 +23,7 @@ extern "C" {
#include "webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h"
}
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

2
webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
View File

@ -21,7 +21,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_processing/vad/common.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {

2
webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
View File

@ -14,7 +14,7 @@
#include <algorithm>
#include <utility>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

2
webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
View File

@ -14,7 +14,7 @@
#include <vector>
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
using webrtc::RtcpBandwidthObserver;
using webrtc::BitrateObserver;

4
webrtc/modules/bitrate_controller/include/bitrate_controller.h
View File

@ -17,8 +17,8 @@
#include <map>
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

2
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
View File

@ -15,7 +15,7 @@
#include <deque>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {

81
webrtc/modules/include/module.h Normal file
View File

@ -0,0 +1,81 @@
/*
* Copyright (c) 2012 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 WEBRTC_MODULES_INCLUDE_MODULE_H_
#define WEBRTC_MODULES_INCLUDE_MODULE_H_
#include "webrtc/typedefs.h"
namespace webrtc {
class ProcessThread;
class Module {
public:
// Returns the number of milliseconds until the module wants a worker
// thread to call Process.
// This method is called on the same worker thread as Process will
// be called on.
// TODO(tommi): Almost all implementations of this function, need to know
// the current tick count. Consider passing it as an argument. It could
// also improve the accuracy of when the next callback occurs since the
// thread that calls Process() will also have it's tick count reference
// which might not match with what the implementations use.
virtual int64_t TimeUntilNextProcess() = 0;
// Process any pending tasks such as timeouts.
// Called on a worker thread.
virtual int32_t Process() = 0;
// This method is called when the module is attached to a *running* process
// thread or detached from one. In the case of detaching, |process_thread|
// will be nullptr.
//
// This method will be called in the following cases:
//
// * Non-null process_thread:
// * ProcessThread::RegisterModule() is called while the thread is running.
// * ProcessThread::Start() is called and RegisterModule has previously
// been called. The thread will be started immediately after notifying
// all modules.
//
// * Null process_thread:
// * ProcessThread::DeRegisterModule() is called while the thread is
// running.
// * ProcessThread::Stop() was called and the thread has been stopped.
//
// NOTE: This method is not called from the worker thread itself, but from
// the thread that registers/deregisters the module or calls Start/Stop.
virtual void ProcessThreadAttached(ProcessThread* process_thread) {}
protected:
virtual ~Module() {}
};
// Reference counted version of the Module interface.
class RefCountedModule : public Module {
public:
// Increase the reference count by one.
// Returns the incremented reference count.
virtual int32_t AddRef() const = 0;
// Decrease the reference count by one.
// Returns the decreased reference count.
// Returns 0 if the last reference was just released.
// When the reference count reaches 0 the object will self-destruct.
virtual int32_t Release() const = 0;
protected:
~RefCountedModule() override = default;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_INCLUDE_MODULE_H_

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/*
* Copyright (c) 2012 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 WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
#include <assert.h>
#include <string.h> // memcpy
#include <algorithm>
#include <limits>
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/rotation.h"
#include "webrtc/typedefs.h"
namespace webrtc {
struct RTPAudioHeader {
uint8_t numEnergy; // number of valid entries in arrOfEnergy
uint8_t arrOfEnergy[kRtpCsrcSize]; // one energy byte (0-9) per channel
bool isCNG; // is this CNG
uint8_t channel; // number of channels 2 = stereo
};
const int16_t kNoPictureId = -1;
const int16_t kMaxOneBytePictureId = 0x7F; // 7 bits
const int16_t kMaxTwoBytePictureId = 0x7FFF; // 15 bits
const int16_t kNoTl0PicIdx = -1;
const uint8_t kNoTemporalIdx = 0xFF;
const uint8_t kNoSpatialIdx = 0xFF;
const uint8_t kNoGofIdx = 0xFF;
const size_t kMaxVp9RefPics = 3;
const size_t kMaxVp9FramesInGof = 0xFF; // 8 bits
const size_t kMaxVp9NumberOfSpatialLayers = 8;
const int kNoKeyIdx = -1;
struct RTPVideoHeaderVP8 {
void InitRTPVideoHeaderVP8() {
nonReference = false;
pictureId = kNoPictureId;
tl0PicIdx = kNoTl0PicIdx;
temporalIdx = kNoTemporalIdx;
layerSync = false;
keyIdx = kNoKeyIdx;
partitionId = 0;
beginningOfPartition = false;
}
bool nonReference; // Frame is discardable.
int16_t pictureId; // Picture ID index, 15 bits;
// kNoPictureId if PictureID does not exist.
int16_t tl0PicIdx; // TL0PIC_IDX, 8 bits;
// kNoTl0PicIdx means no value provided.
uint8_t temporalIdx; // Temporal layer index, or kNoTemporalIdx.
bool layerSync; // This frame is a layer sync frame.
// Disabled if temporalIdx == kNoTemporalIdx.
int keyIdx; // 5 bits; kNoKeyIdx means not used.
int partitionId; // VP8 partition ID
bool beginningOfPartition; // True if this packet is the first
// in a VP8 partition. Otherwise false
};
enum TemporalStructureMode {
kTemporalStructureMode1, // 1 temporal layer structure - i.e., IPPP...
kTemporalStructureMode2, // 2 temporal layers 0-1-0-1...
kTemporalStructureMode3 // 3 temporal layers 0-2-1-2-0-2-1-2...
};
struct GofInfoVP9 {
void SetGofInfoVP9(TemporalStructureMode tm) {
switch (tm) {
case kTemporalStructureMode1:
num_frames_in_gof = 1;
temporal_idx[0] = 0;
temporal_up_switch[0] = false;
num_ref_pics[0] = 1;
pid_diff[0][0] = 1;
break;
case kTemporalStructureMode2:
num_frames_in_gof = 2;
temporal_idx[0] = 0;
temporal_up_switch[0] = false;
num_ref_pics[0] = 1;
pid_diff[0][0] = 2;
temporal_idx[1] = 1;
temporal_up_switch[1] = true;
num_ref_pics[1] = 1;
pid_diff[1][0] = 1;
break;
case kTemporalStructureMode3:
num_frames_in_gof = 4;
temporal_idx[0] = 0;
temporal_up_switch[0] = false;
num_ref_pics[0] = 1;
pid_diff[0][0] = 4;
temporal_idx[1] = 2;
temporal_up_switch[1] = true;
num_ref_pics[1] = 1;
pid_diff[1][0] = 1;
temporal_idx[2] = 1;
temporal_up_switch[2] = true;
num_ref_pics[2] = 1;
pid_diff[2][0] = 2;
temporal_idx[3] = 2;
temporal_up_switch[3] = false;
num_ref_pics[3] = 2;
pid_diff[3][0] = 1;
pid_diff[3][1] = 2;
break;
default:
assert(false);
}
}
void CopyGofInfoVP9(const GofInfoVP9& src) {
num_frames_in_gof = src.num_frames_in_gof;
for (size_t i = 0; i < num_frames_in_gof; ++i) {
temporal_idx[i] = src.temporal_idx[i];
temporal_up_switch[i] = src.temporal_up_switch[i];
num_ref_pics[i] = src.num_ref_pics[i];
for (size_t r = 0; r < num_ref_pics[i]; ++r) {
pid_diff[i][r] = src.pid_diff[i][r];
}
}
}
size_t num_frames_in_gof;
uint8_t temporal_idx[kMaxVp9FramesInGof];
bool temporal_up_switch[kMaxVp9FramesInGof];
size_t num_ref_pics[kMaxVp9FramesInGof];
int16_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
};
struct RTPVideoHeaderVP9 {
void InitRTPVideoHeaderVP9() {
inter_pic_predicted = false;
flexible_mode = false;
beginning_of_frame = false;
end_of_frame = false;
ss_data_available = false;
picture_id = kNoPictureId;
max_picture_id = kMaxTwoBytePictureId;
tl0_pic_idx = kNoTl0PicIdx;
temporal_idx = kNoTemporalIdx;
spatial_idx = kNoSpatialIdx;
temporal_up_switch = false;
inter_layer_predicted = false;
gof_idx = kNoGofIdx;
num_ref_pics = 0;
num_spatial_layers = 1;
}
bool inter_pic_predicted; // This layer frame is dependent on previously
// coded frame(s).
bool flexible_mode; // This frame is in flexible mode.
bool beginning_of_frame; // True if this packet is the first in a VP9 layer
// frame.
bool end_of_frame; // True if this packet is the last in a VP9 layer frame.
bool ss_data_available; // True if SS data is available in this payload
// descriptor.
int16_t picture_id; // PictureID index, 15 bits;
// kNoPictureId if PictureID does not exist.
int16_t max_picture_id; // Maximum picture ID index; either 0x7F or 0x7FFF;
int16_t tl0_pic_idx; // TL0PIC_IDX, 8 bits;
// kNoTl0PicIdx means no value provided.
uint8_t temporal_idx; // Temporal layer index, or kNoTemporalIdx.
uint8_t spatial_idx; // Spatial layer index, or kNoSpatialIdx.
bool temporal_up_switch; // True if upswitch to higher frame rate is possible
// starting from this frame.
bool inter_layer_predicted; // Frame is dependent on directly lower spatial
// layer frame.
uint8_t gof_idx; // Index to predefined temporal frame info in SS data.
size_t num_ref_pics; // Number of reference pictures used by this layer
// frame.
int16_t pid_diff[kMaxVp9RefPics]; // P_DIFF signaled to derive the PictureID
// of the reference pictures.
int16_t ref_picture_id[kMaxVp9RefPics]; // PictureID of reference pictures.
// SS data.
size_t num_spatial_layers; // Always populated.
bool spatial_layer_resolution_present;
uint16_t width[kMaxVp9NumberOfSpatialLayers];
uint16_t height[kMaxVp9NumberOfSpatialLayers];
GofInfoVP9 gof;
};
// The packetization types that we support: single, aggregated, and fragmented.
enum H264PacketizationTypes {
kH264SingleNalu, // This packet contains a single NAL unit.
kH264StapA, // This packet contains STAP-A (single time
// aggregation) packets. If this packet has an
// associated NAL unit type, it'll be for the
// first such aggregated packet.
kH264FuA, // This packet contains a FU-A (fragmentation
// unit) packet, meaning it is a part of a frame
// that was too large to fit into a single packet.
};
struct RTPVideoHeaderH264 {
uint8_t nalu_type; // The NAL unit type. If this is a header for a
// fragmented packet, it's the NAL unit type of
// the original data. If this is the header for an
// aggregated packet, it's the NAL unit type of
// the first NAL unit in the packet.
H264PacketizationTypes packetization_type;
};
union RTPVideoTypeHeader {
RTPVideoHeaderVP8 VP8;
RTPVideoHeaderVP9 VP9;
RTPVideoHeaderH264 H264;
};
enum RtpVideoCodecTypes {
kRtpVideoNone,
kRtpVideoGeneric,
kRtpVideoVp8,
kRtpVideoVp9,
kRtpVideoH264
};
// Since RTPVideoHeader is used as a member of a union, it can't have a
// non-trivial default constructor.
struct RTPVideoHeader {
uint16_t width; // size
uint16_t height;
VideoRotation rotation;
bool isFirstPacket; // first packet in frame
uint8_t simulcastIdx; // Index if the simulcast encoder creating
// this frame, 0 if not using simulcast.
RtpVideoCodecTypes codec;
RTPVideoTypeHeader codecHeader;
};
union RTPTypeHeader {
RTPAudioHeader Audio;
RTPVideoHeader Video;
};
struct WebRtcRTPHeader {
RTPHeader header;
FrameType frameType;
RTPTypeHeader type;
// NTP time of the capture time in local timebase in milliseconds.
int64_t ntp_time_ms;
};
class RTPFragmentationHeader {
public:
RTPFragmentationHeader()
: fragmentationVectorSize(0),
fragmentationOffset(NULL),
fragmentationLength(NULL),
fragmentationTimeDiff(NULL),
fragmentationPlType(NULL) {};
~RTPFragmentationHeader() {
delete[] fragmentationOffset;
delete[] fragmentationLength;
delete[] fragmentationTimeDiff;
delete[] fragmentationPlType;
}
void CopyFrom(const RTPFragmentationHeader& src) {
if (this == &src) {
return;
}
if (src.fragmentationVectorSize != fragmentationVectorSize) {
// new size of vectors
// delete old
delete[] fragmentationOffset;
fragmentationOffset = NULL;
delete[] fragmentationLength;
fragmentationLength = NULL;
delete[] fragmentationTimeDiff;
fragmentationTimeDiff = NULL;
delete[] fragmentationPlType;
fragmentationPlType = NULL;
if (src.fragmentationVectorSize > 0) {
// allocate new
if (src.fragmentationOffset) {
fragmentationOffset = new size_t[src.fragmentationVectorSize];
}
if (src.fragmentationLength) {
fragmentationLength = new size_t[src.fragmentationVectorSize];
}
if (src.fragmentationTimeDiff) {
fragmentationTimeDiff = new uint16_t[src.fragmentationVectorSize];
}
if (src.fragmentationPlType) {
fragmentationPlType = new uint8_t[src.fragmentationVectorSize];
}
}
// set new size
fragmentationVectorSize = src.fragmentationVectorSize;
}
if (src.fragmentationVectorSize > 0) {
// copy values
if (src.fragmentationOffset) {
memcpy(fragmentationOffset, src.fragmentationOffset,
src.fragmentationVectorSize * sizeof(size_t));
}
if (src.fragmentationLength) {
memcpy(fragmentationLength, src.fragmentationLength,
src.fragmentationVectorSize * sizeof(size_t));
}
if (src.fragmentationTimeDiff) {
memcpy(fragmentationTimeDiff, src.fragmentationTimeDiff,
src.fragmentationVectorSize * sizeof(uint16_t));
}
if (src.fragmentationPlType) {
memcpy(fragmentationPlType, src.fragmentationPlType,
src.fragmentationVectorSize * sizeof(uint8_t));
}
}
}
void VerifyAndAllocateFragmentationHeader(const size_t size) {
assert(size <= std::numeric_limits<uint16_t>::max());
const uint16_t size16 = static_cast<uint16_t>(size);
if (fragmentationVectorSize < size16) {
uint16_t oldVectorSize = fragmentationVectorSize;
{
// offset
size_t* oldOffsets = fragmentationOffset;
fragmentationOffset = new size_t[size16];
memset(fragmentationOffset + oldVectorSize, 0,
sizeof(size_t) * (size16 - oldVectorSize));
// copy old values
memcpy(fragmentationOffset, oldOffsets,
sizeof(size_t) * oldVectorSize);
delete[] oldOffsets;
}
// length
{
size_t* oldLengths = fragmentationLength;
fragmentationLength = new size_t[size16];
memset(fragmentationLength + oldVectorSize, 0,
sizeof(size_t) * (size16 - oldVectorSize));
memcpy(fragmentationLength, oldLengths,
sizeof(size_t) * oldVectorSize);
delete[] oldLengths;
}
// time diff
{
uint16_t* oldTimeDiffs = fragmentationTimeDiff;
fragmentationTimeDiff = new uint16_t[size16];
memset(fragmentationTimeDiff + oldVectorSize, 0,
sizeof(uint16_t) * (size16 - oldVectorSize));
memcpy(fragmentationTimeDiff, oldTimeDiffs,
sizeof(uint16_t) * oldVectorSize);
delete[] oldTimeDiffs;
}
// payload type
{
uint8_t* oldTimePlTypes = fragmentationPlType;
fragmentationPlType = new uint8_t[size16];
memset(fragmentationPlType + oldVectorSize, 0,
sizeof(uint8_t) * (size16 - oldVectorSize));
memcpy(fragmentationPlType, oldTimePlTypes,
sizeof(uint8_t) * oldVectorSize);
delete[] oldTimePlTypes;
}
fragmentationVectorSize = size16;
}
}
uint16_t fragmentationVectorSize; // Number of fragmentations
size_t* fragmentationOffset; // Offset of pointer to data for each
// fragmentation
size_t* fragmentationLength; // Data size for each fragmentation
uint16_t* fragmentationTimeDiff; // Timestamp difference relative "now" for
// each fragmentation
uint8_t* fragmentationPlType; // Payload type of each fragmentation
private:
RTC_DISALLOW_COPY_AND_ASSIGN(RTPFragmentationHeader);
};
struct RTCPVoIPMetric {
// RFC 3611 4.7
uint8_t lossRate;
uint8_t discardRate;
uint8_t burstDensity;
uint8_t gapDensity;
uint16_t burstDuration;
uint16_t gapDuration;
uint16_t roundTripDelay;
uint16_t endSystemDelay;
uint8_t signalLevel;
uint8_t noiseLevel;
uint8_t RERL;
uint8_t Gmin;
uint8_t Rfactor;
uint8_t extRfactor;
uint8_t MOSLQ;
uint8_t MOSCQ;
uint8_t RXconfig;
uint16_t JBnominal;
uint16_t JBmax;
uint16_t JBabsMax;
};
// Types for the FEC packet masks. The type |kFecMaskRandom| is based on a
// random loss model. The type |kFecMaskBursty| is based on a bursty/consecutive
// loss model. The packet masks are defined in
// modules/rtp_rtcp/fec_private_tables_random(bursty).h
enum FecMaskType {
kFecMaskRandom,
kFecMaskBursty,
};
// Struct containing forward error correction settings.
struct FecProtectionParams {
int fec_rate;
bool use_uep_protection;
int max_fec_frames;
FecMaskType fec_mask_type;
};
// Interface used by the CallStats class to distribute call statistics.
// Callbacks will be triggered as soon as the class has been registered to a
// CallStats object using RegisterStatsObserver.
class CallStatsObserver {
public:
virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0;
virtual ~CallStatsObserver() {}
};
struct VideoContentMetrics {
VideoContentMetrics()
: motion_magnitude(0.0f),
spatial_pred_err(0.0f),
spatial_pred_err_h(0.0f),
spatial_pred_err_v(0.0f) {}
void Reset() {
motion_magnitude = 0.0f;
spatial_pred_err = 0.0f;
spatial_pred_err_h = 0.0f;
spatial_pred_err_v = 0.0f;
}
float motion_magnitude;
float spatial_pred_err;
float spatial_pred_err_h;
float spatial_pred_err_v;
};
/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
* allows for adding and subtracting frames while keeping track of the resulting
* states.
*
* Notes
* - The total number of samples in |data_| is
* samples_per_channel_ * num_channels_
*
* - Stereo data is interleaved starting with the left channel.
*
* - The +operator assume that you would never add exactly opposite frames when
* deciding the resulting state. To do this use the -operator.
*/
class AudioFrame {
public:
// Stereo, 32 kHz, 60 ms (2 * 32 * 60)
static const size_t kMaxDataSizeSamples = 3840;
enum VADActivity {
kVadActive = 0,
kVadPassive = 1,
kVadUnknown = 2
};
enum SpeechType {
kNormalSpeech = 0,
kPLC = 1,
kCNG = 2,
kPLCCNG = 3,
kUndefined = 4
};
AudioFrame();
virtual ~AudioFrame() {}
// Resets all members to their default state (except does not modify the
// contents of |data_|).
void Reset();
// |interleaved_| is not changed by this method.
void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
size_t samples_per_channel, int sample_rate_hz,
SpeechType speech_type, VADActivity vad_activity,
int num_channels = 1, uint32_t energy = -1);
AudioFrame& Append(const AudioFrame& rhs);
void CopyFrom(const AudioFrame& src);
void Mute();
AudioFrame& operator>>=(const int rhs);
AudioFrame& operator+=(const AudioFrame& rhs);
AudioFrame& operator-=(const AudioFrame& rhs);
int id_;
// RTP timestamp of the first sample in the AudioFrame.
uint32_t timestamp_;
// Time since the first frame in milliseconds.
// -1 represents an uninitialized value.
int64_t elapsed_time_ms_;
// NTP time of the estimated capture time in local timebase in milliseconds.
// -1 represents an uninitialized value.
int64_t ntp_time_ms_;
int16_t data_[kMaxDataSizeSamples];
size_t samples_per_channel_;
int sample_rate_hz_;
int num_channels_;
SpeechType speech_type_;
VADActivity vad_activity_;
// Note that there is no guarantee that |energy_| is correct. Any user of this
// member must verify that the value is correct.
// TODO(henrike) Remove |energy_|.
// See https://code.google.com/p/webrtc/issues/detail?id=3315.
uint32_t energy_;
bool interleaved_;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
};
inline AudioFrame::AudioFrame()
: data_() {
Reset();
}
inline void AudioFrame::Reset() {
id_ = -1;
// TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
// to an invalid value, or add a new member to indicate invalidity.
timestamp_ = 0;
elapsed_time_ms_ = -1;
ntp_time_ms_ = -1;
samples_per_channel_ = 0;
sample_rate_hz_ = 0;
num_channels_ = 0;
speech_type_ = kUndefined;
vad_activity_ = kVadUnknown;
energy_ = 0xffffffff;
interleaved_ = true;
}
inline void AudioFrame::UpdateFrame(int id,
uint32_t timestamp,
const int16_t* data,
size_t samples_per_channel,
int sample_rate_hz,
SpeechType speech_type,
VADActivity vad_activity,
int num_channels,
uint32_t energy) {
id_ = id;
timestamp_ = timestamp;
samples_per_channel_ = samples_per_channel;
sample_rate_hz_ = sample_rate_hz;
speech_type_ = speech_type;
vad_activity_ = vad_activity;
num_channels_ = num_channels;
energy_ = energy;
assert(num_channels >= 0);
const size_t length = samples_per_channel * num_channels;
assert(length <= kMaxDataSizeSamples);
if (data != NULL) {
memcpy(data_, data, sizeof(int16_t) * length);
} else {
memset(data_, 0, sizeof(int16_t) * length);
}
}
inline void AudioFrame::CopyFrom(const AudioFrame& src) {
if (this == &src) return;
id_ = src.id_;
timestamp_ = src.timestamp_;
elapsed_time_ms_ = src.elapsed_time_ms_;
ntp_time_ms_ = src.ntp_time_ms_;
samples_per_channel_ = src.samples_per_channel_;
sample_rate_hz_ = src.sample_rate_hz_;
speech_type_ = src.speech_type_;
vad_activity_ = src.vad_activity_;
num_channels_ = src.num_channels_;
energy_ = src.energy_;
interleaved_ = src.interleaved_;
assert(num_channels_ >= 0);
const size_t length = samples_per_channel_ * num_channels_;
assert(length <= kMaxDataSizeSamples);
memcpy(data_, src.data_, sizeof(int16_t) * length);
}
inline void AudioFrame::Mute() {
memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
}
inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
assert((num_channels_ > 0) && (num_channels_ < 3));
if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
data_[i] = static_cast<int16_t>(data_[i] >> rhs);
}
return *this;
}
inline AudioFrame& AudioFrame::Append(const AudioFrame& rhs) {
// Sanity check
assert((num_channels_ > 0) && (num_channels_ < 3));
assert(interleaved_ == rhs.interleaved_);
if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
if (num_channels_ != rhs.num_channels_) return *this;
if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
vad_activity_ = kVadActive;
} else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
vad_activity_ = kVadUnknown;
}
if (speech_type_ != rhs.speech_type_) {
speech_type_ = kUndefined;
}
size_t offset = samples_per_channel_ * num_channels_;
for (size_t i = 0; i < rhs.samples_per_channel_ * rhs.num_channels_; i++) {
data_[offset + i] = rhs.data_[i];
}
samples_per_channel_ += rhs.samples_per_channel_;
return *this;
}
namespace {
inline int16_t ClampToInt16(int32_t input) {
if (input < -0x00008000) {
return -0x8000;
} else if (input > 0x00007FFF) {
return 0x7FFF;
} else {
return static_cast<int16_t>(input);
}
}
}
inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
// Sanity check
assert((num_channels_ > 0) && (num_channels_ < 3));
assert(interleaved_ == rhs.interleaved_);
if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
if (num_channels_ != rhs.num_channels_) return *this;
bool noPrevData = false;
if (samples_per_channel_ != rhs.samples_per_channel_) {
if (samples_per_channel_ == 0) {
// special case we have no data to start with
samples_per_channel_ = rhs.samples_per_channel_;
noPrevData = true;
} else {
return *this;
}
}
if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
vad_activity_ = kVadActive;
} else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
vad_activity_ = kVadUnknown;
}
if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
if (noPrevData) {
memcpy(data_, rhs.data_,
sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
} else {
// IMPROVEMENT this can be done very fast in assembly
for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
int32_t wrap_guard =
static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
data_[i] = ClampToInt16(wrap_guard);
}
}
energy_ = 0xffffffff;
return *this;
}
inline AudioFrame& AudioFrame::operator-=(const AudioFrame& rhs) {
// Sanity check
assert((num_channels_ > 0) && (num_channels_ < 3));
assert(interleaved_ == rhs.interleaved_);
if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
if ((samples_per_channel_ != rhs.samples_per_channel_) ||
(num_channels_ != rhs.num_channels_)) {
return *this;
}
if ((vad_activity_ != kVadPassive) || rhs.vad_activity_ != kVadPassive) {
vad_activity_ = kVadUnknown;
}
speech_type_ = kUndefined;
for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
int32_t wrap_guard =
static_cast<int32_t>(data_[i]) - static_cast<int32_t>(rhs.data_[i]);
data_[i] = ClampToInt16(wrap_guard);
}
energy_ = 0xffffffff;
return *this;
}
inline bool IsNewerSequenceNumber(uint16_t sequence_number,
uint16_t prev_sequence_number) {
// Distinguish between elements that are exactly 0x8000 apart.
// If s1>s2 and |s1-s2| = 0x8000: IsNewer(s1,s2)=true, IsNewer(s2,s1)=false
// rather than having IsNewer(s1,s2) = IsNewer(s2,s1) = false.
if (static_cast<uint16_t>(sequence_number - prev_sequence_number) == 0x8000) {
return sequence_number > prev_sequence_number;
}
return sequence_number != prev_sequence_number &&
static_cast<uint16_t>(sequence_number - prev_sequence_number) < 0x8000;
}
inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
// Distinguish between elements that are exactly 0x80000000 apart.
// If t1>t2 and |t1-t2| = 0x80000000: IsNewer(t1,t2)=true,
// IsNewer(t2,t1)=false
// rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false.
if (static_cast<uint32_t>(timestamp - prev_timestamp) == 0x80000000) {
return timestamp > prev_timestamp;
}
return timestamp != prev_timestamp &&
static_cast<uint32_t>(timestamp - prev_timestamp) < 0x80000000;
}
inline uint16_t LatestSequenceNumber(uint16_t sequence_number1,
uint16_t sequence_number2) {
return IsNewerSequenceNumber(sequence_number1, sequence_number2)
? sequence_number1
: sequence_number2;
}
inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
}
// Utility class to unwrap a sequence number to a larger type, for easier
// handling large ranges. Note that sequence numbers will never be unwrapped
// to a negative value.
class SequenceNumberUnwrapper {
public:
SequenceNumberUnwrapper() : last_seq_(-1) {}
// Get the unwrapped sequence, but don't update the internal state.
int64_t UnwrapWithoutUpdate(uint16_t sequence_number) {
if (last_seq_ == -1)
return sequence_number;
uint16_t cropped_last = static_cast<uint16_t>(last_seq_);
int64_t delta = sequence_number - cropped_last;
if (IsNewerSequenceNumber(sequence_number, cropped_last)) {
if (delta < 0)
delta += (1 << 16); // Wrap forwards.
} else if (delta > 0 && (last_seq_ + delta - (1 << 16)) >= 0) {
// If sequence_number is older but delta is positive, this is a backwards
// wrap-around. However, don't wrap backwards past 0 (unwrapped).
delta -= (1 << 16);
}
return last_seq_ + delta;
}
// Only update the internal state to the specified last (unwrapped) sequence.
void UpdateLast(int64_t last_sequence) { last_seq_ = last_sequence; }
// Unwrap the sequence number and update the internal state.
int64_t Unwrap(uint16_t sequence_number) {
int64_t unwrapped = UnwrapWithoutUpdate(sequence_number);
UpdateLast(unwrapped);
return unwrapped;
}
private:
int64_t last_seq_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_

8
webrtc/modules/interface/module.h
View File

@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_INTERFACE_MODULE_H_
#define MODULES_INTERFACE_MODULE_H_
#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_
#define WEBRTC_MODULES_INCLUDE_MODULE_H_
#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
#include "webrtc/typedefs.h"
@ -78,4 +80,4 @@ class RefCountedModule : public Module {
} // namespace webrtc
#endif // MODULES_INTERFACE_MODULE_H_
#endif // WEBRTC_MODULES_INCLUDE_MODULE_H_

8
webrtc/modules/interface/module_common_types.h
View File

@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULE_COMMON_TYPES_H
#define MODULE_COMMON_TYPES_H
#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
#include <assert.h>
#include <string.h> // memcpy
@ -807,4 +809,4 @@ class SequenceNumberUnwrapper {
} // namespace webrtc
#endif // MODULE_COMMON_TYPES_H
#endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_

4
webrtc/modules/media_file/BUILD.gn
View File

@ -14,8 +14,8 @@ config("media_file_config") {
source_set("media_file") {
sources = [
"interface/media_file.h",
"interface/media_file_defines.h",
"include/media_file.h",
"include/media_file_defines.h",
"source/media_file_impl.cc",
"source/media_file_impl.h",
"source/media_file_utility.cc",

180
webrtc/modules/media_file/include/media_file.h Normal file
View File

@ -0,0 +1,180 @@
/*
* Copyright (c) 2012 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 WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/media_file/include/media_file_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class MediaFile : public Module
{
public:
// Factory method. Constructor disabled. id is the identifier for the
// MediaFile instance.
static MediaFile* CreateMediaFile(const int32_t id);
static void DestroyMediaFile(MediaFile* module);
// Put 10-60ms of audio data from file into the audioBuffer depending on
// codec frame size. dataLengthInBytes is both an input and output
// parameter. As input parameter it indicates the size of audioBuffer.
// As output parameter it indicates the number of bytes written to
// audioBuffer.
// Note: This API only play mono audio but can be used on file containing
// audio with more channels (in which case the audio will be converted to
// mono).
virtual int32_t PlayoutAudioData(
int8_t* audioBuffer,
size_t& dataLengthInBytes) = 0;
// Put 10-60ms, depending on codec frame size, of audio data from file into
// audioBufferLeft and audioBufferRight. The buffers contain the left and
// right channel of played out stereo audio.
// dataLengthInBytes is both an input and output parameter. As input
// parameter it indicates the size of both audioBufferLeft and
// audioBufferRight. As output parameter it indicates the number of bytes
// written to both audio buffers.
// Note: This API can only be successfully called for WAV files with stereo
// audio.
virtual int32_t PlayoutStereoData(
int8_t* audioBufferLeft,
int8_t* audioBufferRight,
size_t& dataLengthInBytes) = 0;
// Open the file specified by fileName (relative path is allowed) for
// reading. FileCallback::PlayNotification(..) will be called after
// notificationTimeMs of the file has been played if notificationTimeMs is
// greater than zero. If loop is true the file will be played until
// StopPlaying() is called. When end of file is reached the file is read
// from the start. format specifies the type of file fileName refers to.
// codecInst specifies the encoding of the audio data. Note that
// file formats that contain this information (like WAV files) don't need to
// provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
// specify what part of the file should be read. From startPointMs ms to
// stopPointMs ms.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo audio is only supported for WAV files.
virtual int32_t StartPlayingAudioFile(
const char* fileName,
const uint32_t notificationTimeMs = 0,
const bool loop = false,
const FileFormats format = kFileFormatPcm16kHzFile,
const CodecInst* codecInst = NULL,
const uint32_t startPointMs = 0,
const uint32_t stopPointMs = 0) = 0;
// Prepare for playing audio from stream.
// FileCallback::PlayNotification(..) will be called after
// notificationTimeMs of the file has been played if notificationTimeMs is
// greater than zero. format specifies the type of file fileName refers to.
// codecInst specifies the encoding of the audio data. Note that
// file formats that contain this information (like WAV files) don't need to
// provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
// specify what part of the file should be read. From startPointMs ms to
// stopPointMs ms.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo audio is only supported for WAV files.
virtual int32_t StartPlayingAudioStream(
InStream& stream,
const uint32_t notificationTimeMs = 0,
const FileFormats format = kFileFormatPcm16kHzFile,
const CodecInst* codecInst = NULL,
const uint32_t startPointMs = 0,
const uint32_t stopPointMs = 0) = 0;
// Stop playing from file or stream.
virtual int32_t StopPlaying() = 0;
// Return true if playing.
virtual bool IsPlaying() = 0;
// Set durationMs to the number of ms that has been played from file.
virtual int32_t PlayoutPositionMs(
uint32_t& durationMs) const = 0;
// Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
// to file. The audio frame size is determined by the codecInst.pacsize
// parameter of the last sucessfull StartRecordingAudioFile(..) call.
// Note: bufferLength must be exactly one frame.
virtual int32_t IncomingAudioData(
const int8_t* audioBuffer,
const size_t bufferLength) = 0;
// Open/creates file specified by fileName for writing (relative path is
// allowed). FileCallback::RecordNotification(..) will be called after
// notificationTimeMs of audio data has been recorded if
// notificationTimeMs is greater than zero.
// format specifies the type of file that should be created/opened.
// codecInst specifies the encoding of the audio data. maxSizeBytes
// specifies the number of bytes allowed to be written to file if it is
// greater than zero.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo is only supported for WAV files.
virtual int32_t StartRecordingAudioFile(
const char* fileName,
const FileFormats format,
const CodecInst& codecInst,
const uint32_t notificationTimeMs = 0,
const uint32_t maxSizeBytes = 0) = 0;
// Prepare for recording audio to stream.
// FileCallback::RecordNotification(..) will be called after
// notificationTimeMs of audio data has been recorded if
// notificationTimeMs is greater than zero.
// format specifies the type of file that stream should correspond to.
// codecInst specifies the encoding of the audio data.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo is only supported for WAV files.
virtual int32_t StartRecordingAudioStream(
OutStream& stream,
const FileFormats format,
const CodecInst& codecInst,
const uint32_t notificationTimeMs = 0) = 0;
// Stop recording to file or stream.
virtual int32_t StopRecording() = 0;
// Return true if recording.
virtual bool IsRecording() = 0;
// Set durationMs to the number of ms that has been recorded to file.
virtual int32_t RecordDurationMs(uint32_t& durationMs) = 0;
// Return true if recording or playing is stereo.
virtual bool IsStereo() = 0;
// Register callback to receive media file related notifications. Disables
// callbacks if callback is NULL.
virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0;
// Set durationMs to the size of the file (in ms) specified by fileName.
// format specifies the type of file fileName refers to. freqInHz specifies
// the sampling frequency of the file.
virtual int32_t FileDurationMs(
const char* fileName,
uint32_t& durationMs,
const FileFormats format,
const uint32_t freqInHz = 16000) = 0;
// Update codecInst according to the current audio codec being used for
// reading or writing.
virtual int32_t codec_info(CodecInst& codecInst) const = 0;
protected:
MediaFile() {}
virtual ~MediaFile() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_

51
webrtc/modules/media_file/include/media_file_defines.h Normal file
View File

@ -0,0 +1,51 @@
/*
* Copyright (c) 2011 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 WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Callback class for the MediaFile class.
class FileCallback
{
public:
virtual ~FileCallback(){}
// This function is called by MediaFile when a file has been playing for
// durationMs ms. id is the identifier for the MediaFile instance calling
// the callback.
virtual void PlayNotification(const int32_t id,
const uint32_t durationMs) = 0;
// This function is called by MediaFile when a file has been recording for
// durationMs ms. id is the identifier for the MediaFile instance calling
// the callback.
virtual void RecordNotification(const int32_t id,
const uint32_t durationMs) = 0;
// This function is called by MediaFile when a file has been stopped
// playing. id is the identifier for the MediaFile instance calling the
// callback.
virtual void PlayFileEnded(const int32_t id) = 0;
// This function is called by MediaFile when a file has been stopped
// recording. id is the identifier for the MediaFile instance calling the
// callback.
virtual void RecordFileEnded(const int32_t id) = 0;
protected:
FileCallback() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_

14
webrtc/modules/media_file/interface/media_file.h
View File

@ -8,13 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
#include "webrtc/common_types.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/media_file/interface/media_file_defines.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/media_file/include/media_file_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -177,4 +179,4 @@ protected:
virtual ~MediaFile() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_

10
webrtc/modules/media_file/interface/media_file_defines.h
View File

@ -8,11 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -48,4 +50,4 @@ protected:
FileCallback() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_

4
webrtc/modules/media_file/media_file.gypi
View File

@ -17,8 +17,8 @@
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
'interface/media_file.h',
'interface/media_file_defines.h',
'include/media_file.h',
'include/media_file_defines.h',
'source/media_file_impl.cc',
'source/media_file_impl.h',
'source/media_file_utility.cc',

6
webrtc/modules/media_file/source/media_file_impl.h
View File

@ -12,9 +12,9 @@
#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/media_file/interface/media_file.h"
#include "webrtc/modules/media_file/interface/media_file_defines.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/media_file/include/media_file.h"
#include "webrtc/modules/media_file/include/media_file_defines.h"
#include "webrtc/modules/media_file/source/media_file_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"

2
webrtc/modules/media_file/source/media_file_unittest.cc
View File

@ -9,7 +9,7 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/media_file/interface/media_file.h"
#include "webrtc/modules/media_file/include/media_file.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"

2
webrtc/modules/media_file/source/media_file_utility.cc
View File

@ -19,7 +19,7 @@
#include "webrtc/common_audio/wav_header.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"

2
webrtc/modules/media_file/source/media_file_utility.h
View File

@ -15,7 +15,7 @@
#include <stdio.h>
#include "webrtc/common_types.h"
#include "webrtc/modules/media_file/interface/media_file_defines.h"
#include "webrtc/modules/media_file/include/media_file_defines.h"
namespace webrtc {
class InStream;

2
webrtc/modules/module_common_types_unittest.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "testing/gtest/include/gtest/gtest.h"

4
webrtc/modules/pacing/include/paced_sender.h
View File

@ -16,8 +16,8 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/pacing/include/packet_router.h
View File

@ -19,7 +19,7 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

2
webrtc/modules/pacing/paced_sender.cc
View File

@ -16,7 +16,7 @@
#include <queue>
#include <set>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/pacing/bitrate_prober.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"

4
webrtc/modules/pacing/packet_router.cc
View File

@ -12,8 +12,8 @@
#include "webrtc/base/atomicops.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
namespace webrtc {

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