Create ACM2 instance when calling AudioCodingModule::Create

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5952 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/acm2/acm_receiver.cc

@@ -23,6 +23,7 @@
 #include "webrtc/modules/audio_coding/main/acm2/nack.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
+#include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
@@ -116,7 +117,7 @@ bool IsCng(int codec_id) {
 
 }  // namespace
 
-AcmReceiver::AcmReceiver()
+AcmReceiver::AcmReceiver(Clock* clock)
     : id_(0),
       neteq_config_(),
       neteq_(NetEq::Create(neteq_config_)),
@@ -128,6 +129,7 @@ AcmReceiver::AcmReceiver()
       current_sample_rate_hz_(neteq_config_.sample_rate_hz),
       nack_(),
       nack_enabled_(false),
+      clock_(clock),
       av_sync_(false),
       initial_delay_manager_(),
       missing_packets_sync_stream_(),
@@ -817,7 +819,7 @@ uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
   // We masked 6 most significant bits of 32-bit so there is no overflow in
   // the conversion from milliseconds to timestamp.
   const uint32_t now_in_ms = static_cast<uint32_t>(
-      TickTime::MillisecondTimestamp() & 0x03ffffff);
+      clock_->TimeInMilliseconds() & 0x03ffffff);
   return static_cast<uint32_t>(
       (decoder_sampling_rate / 1000) * now_in_ms);
 }

webrtc/modules/audio_coding/main/acm2/acm_receiver.h

@@ -27,6 +27,7 @@
 
 namespace webrtc {
 
+class Clock;
 struct CodecInst;
 class CriticalSectionWrapper;
 class RWLockWrapper;
@@ -47,9 +48,7 @@ class AcmReceiver {
   };
 
   // Constructor of the class
-  AcmReceiver();
+  explicit AcmReceiver(Clock* clock);
-
-  explicit AcmReceiver(NetEq* neteq);
 
   // Destructor of the class.
   ~AcmReceiver();
@@ -354,6 +353,7 @@ class AcmReceiver {
   int16_t audio_buffer_[AudioFrame::kMaxDataSizeSamples];
   scoped_ptr<Nack> nack_;
   bool nack_enabled_;
+  Clock* clock_;
 
   // Indicates if a non-zero initial delay is set, and the receiver is in
   // AV-sync mode.

webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc

@@ -17,6 +17,7 @@
 #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
+#include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -42,8 +43,8 @@ class AcmReceiverTest : public AudioPacketizationCallback,
                         public ::testing::Test {
  protected:
   AcmReceiverTest()
-      : receiver_(new AcmReceiver),
+      : receiver_(new AcmReceiver(Clock::GetRealTimeClock())),
-        acm_(new AudioCodingModuleImpl(0)),
+        acm_(new AudioCodingModuleImpl(0, Clock::GetRealTimeClock())),
         timestamp_(0),
         packet_sent_(false),
         last_packet_send_timestamp_(timestamp_),

webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc

@@ -24,11 +24,11 @@ const char kExperimentalAcmVersion[] = "acm2";
 
 // Create module
 AudioCodingModule* AudioCodingModule::Create(int id) {
-  return new acm1::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
+  return new acm2::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
 }
 
 AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
-  return new acm1::AudioCodingModuleImpl(id, clock);
+  return new acm2::AudioCodingModuleImpl(id, clock);
 }
 
 // Get number of supported codecs
@@ -101,7 +101,7 @@ AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
 }
 
 AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
-  return new acm2::AudioCodingModuleImpl(id);
+  return new acm2::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
 }
 
 }  // namespace webrtc

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

@@ -114,7 +114,7 @@ static int TimestampLessThan(uint32_t t1, uint32_t t2) {
 
 }  // namespace
 
-AudioCodingModuleImpl::AudioCodingModuleImpl(int id)
+AudioCodingModuleImpl::AudioCodingModuleImpl(int id, Clock* clock)
     : packetization_callback_(NULL),
       id_(id),
       expected_codec_ts_(0xD87F3F9F),
@@ -131,6 +131,7 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(int id)
       stereo_send_(false),
       current_send_codec_idx_(-1),
       send_codec_registered_(false),
+      receiver_(clock),
       acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       vad_callback_(NULL),
       is_first_red_(true),

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

@@ -22,6 +22,7 @@
 
 namespace webrtc {
 
+class Clock;
 class CriticalSectionWrapper;
 class RWLockWrapper;
 
@@ -32,7 +33,7 @@ class ACMGenericCodec;
 
 class AudioCodingModuleImpl : public AudioCodingModule {
  public:
-  explicit AudioCodingModuleImpl(int id);
+  AudioCodingModuleImpl(int id, Clock* clock);
   ~AudioCodingModuleImpl();
 
   virtual const char* Version() const;