Update to the neteq_rtpplay utility to support RtcEventLog input files.

webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

Index: webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 6bcd717279b8b5bf4ce8423eaec62b825908997c..c925fe013a9d25c04f515d43350fda0492a33506 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -30,6 +30,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"
 #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/system_wrappers/interface/trace.h"
@@ -137,6 +138,10 @@ DEFINE_string(ssrc,
               "",
               "Only use packets with this SSRC (decimal or hex, the latter "
               "starting with 0x)");
+DEFINE_bool(rtc_event_log,
+            false,
+            "Enables using a stored RtcEventLog as input "

[hlundin-webrtc, 2015/08/28 12:06:24]

No need to break this line.

[ivoc, 2015/09/01 10:03:50]

Fixed. The string was longer before and I forgot to undo the line break :-)

+            "file.");
 const bool hex_ssrc_dummy =
     google::RegisterFlagValidator(&FLAGS_ssrc, &ValidateSsrcValue);
 
@@ -384,8 +389,15 @@ int main(int argc, char* argv[]) {
   }
 
   printf("Input file: %s\n", argv[1]);
-  rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
-      webrtc::test::RtpFileSource::Create(argv[1]));
+  rtc::scoped_ptr<webrtc::test::PacketSource> file_source;
+  webrtc::test::RtcEventLogSource* event_log_source = nullptr;
+  if (FLAGS_rtc_event_log) {
+    event_log_source = webrtc::test::RtcEventLogSource::Create(argv[1]);

[minyue-webrtc, 2015/08/28 14:50:21]

Is there a way to detect the format automatically?

[ivoc, 2015/09/01 10:03:50]

Good idea. I added this feature by checking if the file starts with the magic
string "#!rtpplay1.0", which indicates it is an RTPdump file. Otherwise, it is
assumed the input file is in RtcEventLog format. I removed the command line
flag.

+    file_source.reset(event_log_source);
+  } else {
+    file_source.reset(webrtc::test::RtpFileSource::Create(argv[1]));
+  }
+
   assert(file_source.get());
 
   // Check if an SSRC value was provided.
@@ -414,6 +426,7 @@ int main(int argc, char* argv[]) {
     return 0;
   }
   bool packet_available = true;
+  bool output_event_available = true;

[minyue-webrtc, 2015/08/28 16:54:17]

I think 428 and 429 are too far, move to 500 if possile

[ivoc, 2015/09/01 10:03:50]

Done.

 
   // Check the sample rate.
   int sample_rate_hz = CodecSampleRate(packet->header().payloadType);
@@ -485,7 +498,16 @@ int main(int argc, char* argv[]) {
     next_output_time_ms +=
         kOutputBlockSizeMs - time_now_ms % kOutputBlockSizeMs;
   }
-  while (packet_available) {
+  if (FLAGS_rtc_event_log) {
+    next_output_time_ms = event_log_source->NextAudioOutputEventMs();

[minyue-webrtc, 2015/08/28 16:54:17]

why not keeping -1 or let NextAudioOutputEventMs give INT_MAX

[ivoc, 2015/09/01 10:03:50]

Keeping it at -1 would not work because later on the next simulation time is
calculated as the minimum of next_output_time_ms and next_input_time_ms, and we
don't want the time to go backwards to -1.
Good idea on returning INT_MAX from NextAudioOutputEventMs, I changed it.

+    if (next_output_time_ms == -1) {
+      output_event_available = false;
+      next_output_time_ms = INT_MAX;
+    }
+    start_time_ms = time_now_ms =
+        std::min(next_input_time_ms, next_output_time_ms);
+  }
+  while (packet_available || output_event_available) {
     // Check if it is time to insert packet.
     while (time_now_ms >= next_input_time_ms && packet_available) {
       assert(packet->virtual_payload_length_bytes() > 0);
@@ -535,7 +557,11 @@ int main(int argc, char* argv[]) {
       if (temp_packet) {
         packet.reset(temp_packet);
       } else {
+        // Set next input time to INT_MAX to prevent the time_now_ms from
+        // becoming stuck at the final value.
+        next_input_time_ms = INT_MAX;
         packet_available = false;
+        break;

[hlundin-webrtc, 2015/08/28 12:06:24]

What are you breaking out of, and why do you have to?

[ivoc, 2015/09/01 10:03:50]

This break is for the while loop on line 512, it is needed to avoid line 576,
which would update the value of next_input_time_ms to something other than
INT_MAX.

I restructured the code a bit to avoid the need for the break, by moving the if
(replace_payload) code into the if (temp_packet) block.

       }
       if (replace_payload) {

[hlundin-webrtc, 2015/08/28 12:06:25]

Did you try to run with a replacement audio file? I have a feeling you may have
to move 
next_input_time_ms = INT_MAX;
to after this if statement.

[ivoc, 2015/09/01 10:03:50]

It seems like next_input_time_ms is not used at all in this if-statement, so I
don't see why it would matter if I do the assignment before or afterwards? I
tried using replacement audio, and it seems to work as expected.

[hlundin-webrtc, 2015/09/02 08:48:51]

Acknowledged.

         // At this point |packet| contains the packet *after* |next_packet|.
@@ -551,7 +577,7 @@ int main(int argc, char* argv[]) {
     }
 
     // Check if it is time to get output audio.
-    if (time_now_ms >= next_output_time_ms) {
+    while (time_now_ms >= next_output_time_ms && output_event_available) {
       static const int kOutDataLen =
           kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
       int16_t out_data[kOutDataLen];
@@ -576,11 +602,24 @@ int main(int argc, char* argv[]) {
         exit(1);
       }
       next_output_time_ms += kOutputBlockSizeMs;
+      if (FLAGS_rtc_event_log) {
+        next_output_time_ms = event_log_source->NextAudioOutputEventMs();
+        if (next_output_time_ms == -1) {
+          output_event_available = false;
+          // Set the next output time to INT_MAX to prevent the time_now_ms
+          // from being set to -1.
+          next_output_time_ms = INT_MAX;
+          break;

[hlundin-webrtc, 2015/08/28 12:06:24]

Same as the break above.

[ivoc, 2015/09/01 10:03:50]

Here the break is for the while loop on line 580, but it's not strictly
necessary here. I will remove it.

+        }
+      } else {
+        if (!packet_available) {
+          output_event_available = false;
+        }
+      }
     }
     // Advance time to next event.
     time_now_ms = std::min(next_input_time_ms, next_output_time_ms);
   }
-
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n", time_now_ms - start_time_ms);