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Remove ACM1 and NetEq3 related targets from modules.gyp

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Make necessary changes to compile.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6008 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-04-29 08:58:46 +00:00
parent fdf2053787
commit 70e53fa34d
6 changed files with 61 additions and 24 deletions
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4
webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
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@ -13,15 +13,11 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
const char kLegacyAcmVersion[] = "acm1";
const char kExperimentalAcmVersion[] = "acm2";
// Create module
AudioCodingModule* AudioCodingModule::Create(int id) {
return Create(id, Clock::GetRealTimeClock());

61
webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
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@ -7,9 +7,29 @@
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'audio_coding_dependencies': [
'CNG',
'G711',
'G722',
'iLBC',
'iSAC',
'iSACFix',
'PCM16B',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'audio_coding_defines': [],
'conditions': [
['include_opus==1', {
'audio_coding_dependencies': ['webrtc_opus',],
'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
}],
],
},
'targets': [
{
'target_name': 'acm2',
'target_name': 'audio_coding_module',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
@ -93,4 +113,43 @@
],
},
],
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'delay_test',
'type': 'executable',
'dependencies': [
'audio_coding_module',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
'sources': [
'../test/delay_test.cc',
'../test/Channel.cc',
'../test/PCMFile.cc',
'../test/utility.cc',
],
}, # delay_test
{
'target_name': 'insert_packet_with_timing',
'type': 'executable',
'dependencies': [
'audio_coding_module',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
'sources': [
'../test/insert_packet_with_timing.cc',
'../test/Channel.cc',
'../test/PCMFile.cc',
],
}, # delay_test
],
}],
],
}

4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -1978,10 +1978,6 @@ int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
return receiver_.LeastRequiredDelayMs();
}
const char* AudioCodingModuleImpl::Version() const {
return kExperimentalAcmVersion;
}
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
receiver_.GetDecodingCallStatistics(call_stats);

2
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
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@ -35,8 +35,6 @@ class AudioCodingModuleImpl : public AudioCodingModule {
explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
~AudioCodingModuleImpl();
virtual const char* Version() const;
// Change the unique identifier of this object.
virtual int32_t ChangeUniqueId(const int32_t id);

9
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
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@ -74,10 +74,6 @@ class ACMVQMonCallback {
const uint16_t delayMS) = 0; // average delay in ms
};
// Version string for testing, to distinguish instances of ACM1 from ACM2.
extern const char kLegacyAcmVersion[];
extern const char kExperimentalAcmVersion[];
class AudioCodingModule: public Module {
protected:
AudioCodingModule() {}
@ -190,11 +186,6 @@ class AudioCodingModule: public Module {
//
static bool IsCodecValid(const CodecInst& codec);
// Returns the version of ACM. This facilitates distinguishing instances of
// ACM1 from ACM2 while testing. This API will be removed when ACM1 is
// completely removed.
virtual const char* Version() const = 0;
///////////////////////////////////////////////////////////////////////////
// Sender
//

5
webrtc/modules/modules.gyp
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@ -16,8 +16,7 @@
'audio_coding/codecs/isac/main/source/isac.gypi',
'audio_coding/codecs/isac/fix/source/isacfix.gypi',
'audio_coding/codecs/pcm16b/pcm16b.gypi',
'audio_coding/main/source/audio_coding_module.gypi',
'audio_coding/neteq/neteq.gypi',
'audio_coding/main/acm2/audio_coding_module.gypi',
'audio_coding/neteq4/neteq.gypi',
'audio_conference_mixer/source/audio_conference_mixer.gypi',
'audio_device/audio_device.gypi',
@ -76,7 +75,6 @@
'desktop_capture',
'iSACFix',
'media_file',
'NetEq',
'NetEq4',
'NetEq4TestTools',
'neteq_unittest_tools',
@ -107,7 +105,6 @@
'audio_coding/main/acm2/call_statistics_unittest.cc',
'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
'audio_coding/main/acm2/nack_unittest.cc',
'audio_coding/main/source/acm_neteq_unittest.cc',
'audio_coding/codecs/cng/cng_unittest.cc',
'audio_coding/codecs/isac/fix/source/filters_unittest.cc',
'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',