Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d

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

@@ -475,10 +475,17 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   call_stats_.DecodedByNetEq(audio_frame->speech_type_);
 
   // Computes the RTP timestamp of the first sample in |audio_frame| from
-  // |PlayoutTimestamp|, which is the timestamp of the last sample of
+  // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
   // |audio_frame|.
-  audio_frame->timestamp_ =
-      PlayoutTimestamp() - audio_frame->samples_per_channel_;
+  uint32_t playout_timestamp = 0;
+  if (GetPlayoutTimestamp(&playout_timestamp)) {
+    audio_frame->timestamp_ =
+        playout_timestamp - audio_frame->samples_per_channel_;
+  } else {
+    // Remain 0 until we have a valid |playout_timestamp|.
+    audio_frame->timestamp_ = 0;
+  }
+
   return 0;
 }
 
@@ -596,13 +603,14 @@ void AcmReceiver::set_id(int id) {
   id_ = id;
 }
 
-uint32_t AcmReceiver::PlayoutTimestamp() {
+bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
   if (av_sync_) {
     assert(initial_delay_manager_.get());
-    if (initial_delay_manager_->buffering())
-      return initial_delay_manager_->playout_timestamp();
+    if (initial_delay_manager_->buffering()) {
+      return initial_delay_manager_->GetPlayoutTimestamp(timestamp);
+    }
   }
-  return neteq_->PlayoutTimestamp();
+  return neteq_->GetPlayoutTimestamp(timestamp);
 }
 
 int AcmReceiver::last_audio_codec_id() const {

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

@@ -242,9 +242,10 @@ class AcmReceiver {
   void set_id(int id);  // TODO(turajs): can be inline.
 
   //
-  // Returns the RTP timestamp of the last sample delivered by GetAudio().
+  // Gets the RTP timestamp of the last sample delivered by GetAudio().
+  // Returns true if the RTP timestamp is valid, otherwise false.
   //
-  uint32_t PlayoutTimestamp();
+  bool GetPlayoutTimestamp(uint32_t* timestamp);
 
   //
   // Return the index of the codec associated with the last non-CNG/non-DTMF

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

@@ -1776,7 +1776,6 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
   }
 
   audio_frame->id_ = id_;
-  audio_frame->timestamp_ = 0;
   return 0;
 }
 
@@ -1917,8 +1916,7 @@ int AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
 }
 
 int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
-  *timestamp = receiver_.PlayoutTimestamp();
-  return 0;
+  return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1;
 }
 
 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {

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

@@ -219,6 +219,14 @@ void InitialDelayManager::LatePackets(
   return;
 }
 
+bool InitialDelayManager::GetPlayoutTimestamp(uint32_t* playout_timestamp) {
+  if (!buffering_) {
+    return false;
+  }
+  *playout_timestamp = playout_timestamp_;
+  return true;
+}
+
 void InitialDelayManager::DisableBuffering() {
   buffering_ = false;
 }

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

@@ -65,8 +65,9 @@ class InitialDelayManager {
   // sequence of late (or perhaps missing) packets is computed.
   void LatePackets(uint32_t timestamp_now, SyncStream* sync_stream);
 
-  // Playout timestamp, valid when buffering.
-  uint32_t playout_timestamp() { return playout_timestamp_; }
+  // Get playout timestamp.
+  // Returns true if the timestamp is valid (when buffering), otherwise false.
+  bool GetPlayoutTimestamp(uint32_t* playout_timestamp);
 
   // True if buffered audio is less than the given initial delay (specified at
   // the constructor). Buffering might be disabled by the client of this class.

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

@@ -359,7 +359,9 @@ TEST_F(InitialDelayManagerTest, BufferingAudio) {
     EXPECT_TRUE(manager_->buffering());
     const uint32_t expected_playout_timestamp = rtp_info_.header.timestamp -
         kInitDelayMs * kSamplingRateHz / 1000;
-    EXPECT_EQ(expected_playout_timestamp, manager_->playout_timestamp());
+    uint32_t actual_playout_timestamp = 0;
+    EXPECT_TRUE(manager_->GetPlayoutTimestamp(&actual_playout_timestamp));
+    EXPECT_EQ(expected_playout_timestamp, actual_playout_timestamp);
     NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
   }