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..5021650d169039a49c9d676bd103f02417801138 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -19,6 +19,7 @@
 
 #include <algorithm>
 #include <iostream>
+#include <limits>
 #include <string>
 
 #include "google/gflags.h"
@@ -30,6 +31,7 @@
 #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"
@@ -384,8 +386,36 @@ 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]));
+
+  // Check if the magic string "#!rtpplay1.0" appears at the beginning of the

[minyue-webrtc, 2015/09/01 11:48:35]

nice!

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

Yes, but I suggest you also add RTPencode as a magic string. We used to use that
one for the rtpdump files that we created ourselves from scratch (as opposed to
converting from wireshark traces). See e.g., webrtc/test/rtp_file_reader.cc,
lines 145--.

[ivoc, 2015/09/03 08:01:31]

Ah, I did not know that. Thanks.

+  // file. If it does, the input file is an RTPdump file, otherwise we will
+  // assume it is an RtcEventLog file.
+  bool is_rtp_dump = false;

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

This is a lot of extra code to check what is essentially already checked inside
RtpFileSource. I suggest you modify RtpFileSource::OpenFile and
RtpFileSource::Create to not use FATAL and CHECK for failed files. Instead,
Create should return nullptr if the format is wrong. (You may want to do this in
a separate CL, and also go through all other uses of RtpFileSource::Create that
today blindly assumes it always works, and perhaps CHECK that we got
non-nullptr.)

Then you would not need all of this. Instead, simply try to open the file with
RtpFileSource::Create. If that fails, try with RtcEventLogSource::Create. If
that fails too: let slip the dogs of war.

[ivoc, 2015/09/03 08:01:31]

You're right, that does seem like a nicer approach. For now I replaced this code
with a temporary solution using a RtpFileReader directly to check if the file is
valid, and re-reading it if it is. I will make your suggested changes in a
follow-up CL.

+  {
+    const char magic_string[] = "#!rtpplay1.0";
+    const size_t magic_string_size = sizeof(magic_string) - 1;
+    char first_bytes[magic_string_size];
+    FILE* input_file = fopen(argv[1], "r");
+    if (input_file) {
+      size_t bytes_read = fread(first_bytes, 1, magic_string_size, input_file);
+      if (bytes_read == magic_string_size)
+        if (strncmp(magic_string, first_bytes, magic_string_size) == 0)
+          is_rtp_dump = true;
+      fclose(input_file);
+    } else {
+      std::cerr << "Unable to read input file." << std::endl;
+      return -1;
+    }
+  }
+  rtc::scoped_ptr<webrtc::test::PacketSource> file_source;
+  webrtc::test::RtcEventLogSource* event_log_source = nullptr;
+  if (is_rtp_dump) {
+    file_source.reset(webrtc::test::RtpFileSource::Create(argv[1]));
+  } else {
+    event_log_source = webrtc::test::RtcEventLogSource::Create(argv[1]);
+    file_source.reset(event_log_source);
+  }
+
   assert(file_source.get());
 
   // Check if an SSRC value was provided.
@@ -413,7 +443,6 @@ int main(int argc, char* argv[]) {
     webrtc::Trace::ReturnTrace();
     return 0;
   }
-  bool packet_available = true;
 
   // Check the sample rate.
   int sample_rate_hz = CodecSampleRate(packet->header().payloadType);
@@ -475,17 +504,29 @@ int main(int argc, char* argv[]) {
 
   // This is the main simulation loop.
   // Set the simulation clock to start immediately with the first packet.
-  int start_time_ms = packet->time_ms();
-  int time_now_ms = packet->time_ms();
-  int next_input_time_ms = time_now_ms;
-  int next_output_time_ms = time_now_ms;
+  int64_t start_time_ms = packet->time_ms();

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

packet->time_ms() actually returns a double. Suggest you use
rtc::checked_cast<int64_t>(...)

[ivoc, 2015/09/03 08:01:31]

Done.

+  int64_t time_now_ms = packet->time_ms();
+  int64_t next_input_time_ms = time_now_ms;
+  int64_t next_output_time_ms = time_now_ms;
   if (time_now_ms % kOutputBlockSizeMs != 0) {
     // Make sure that next_output_time_ms is rounded up to the next multiple
     // of kOutputBlockSizeMs. (Legacy bit-exactness.)
     next_output_time_ms +=
         kOutputBlockSizeMs - time_now_ms % kOutputBlockSizeMs;
   }
-  while (packet_available) {
+
+  bool packet_available = true;
+  bool output_event_available = true;
+  if (!is_rtp_dump) {
+    next_output_time_ms = event_log_source->NextAudioOutputEventMs();
+    if (next_output_time_ms == std::numeric_limits<int64_t>::max())
+      output_event_available = false;
+    start_time_ms = time_now_ms =
+        std::min(next_input_time_ms, next_output_time_ms);
+  }
+  while (packet_available || output_event_available) {
+    // Advance time to next event.
+    time_now_ms = std::min(next_input_time_ms, next_output_time_ms);
     // 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);
@@ -534,24 +575,27 @@ int main(int argc, char* argv[]) {
       webrtc::test::Packet* temp_packet = file_source->NextPacket();
       if (temp_packet) {
         packet.reset(temp_packet);
+        if (replace_payload) {
+          // At this point |packet| contains the packet *after* |next_packet|.
+          // Swap Packet objects between |packet| and |next_packet|.
+          packet.swap(next_packet);
+          // Swap the status indicators unless they're already the same.
+          if (packet_available != next_packet_available) {
+            packet_available = !packet_available;
+            next_packet_available = !next_packet_available;
+          }
+        }
+        next_input_time_ms = packet->time_ms();

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

checked_cast here too

[ivoc, 2015/09/03 08:01:31]

Added. I also added one in the call to neteq->InsertPacket on line 551.

       } else {
+        // Set next input time to the maximum value of int64_t to prevent the
+        // time_now_ms from becoming stuck at the final value.
+        next_input_time_ms = std::numeric_limits<int64_t>::max();
         packet_available = false;
       }
-      if (replace_payload) {
-        // At this point |packet| contains the packet *after* |next_packet|.
-        // Swap Packet objects between |packet| and |next_packet|.
-        packet.swap(next_packet);
-        // Swap the status indicators unless they're already the same.
-        if (packet_available != next_packet_available) {
-          packet_available = !packet_available;
-          next_packet_available = !next_packet_available;
-        }
-      }
-      next_input_time_ms = packet->time_ms();
     }
 
     // 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];
@@ -575,14 +619,20 @@ int main(int argc, char* argv[]) {
         webrtc::Trace::ReturnTrace();
         exit(1);
       }
-      next_output_time_ms += kOutputBlockSizeMs;
+      if (is_rtp_dump) {
+        next_output_time_ms += kOutputBlockSizeMs;
+        if (!packet_available)
+          output_event_available = false;
+      } else {
+        next_output_time_ms = event_log_source->NextAudioOutputEventMs();
+        if (next_output_time_ms == std::numeric_limits<int64_t>::max())
+          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);
+  printf("Produced %i ms of audio\n",
+         static_cast<int>(time_now_ms - start_time_ms));
 
   delete neteq;
   webrtc::Trace::ReturnTrace();