Update to the neteq_rtpplay utility to support RtcEventLog input files.

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

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 12 matching lines @@
 
 #include "google/gflags.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #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"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 using webrtc::NetEq;
 using webrtc::WebRtcRTPHeader;
 
 namespace {
@@ skipping 87 matching lines @@
 const bool cn_swb48_dummy =
     google::RegisterFlagValidator(&FLAGS_cn_swb48, &ValidatePayloadType);
 DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and "
     "codec");
 DEFINE_string(replacement_audio_file, "",
               "A PCM file that will be used to populate ""dummy"" RTP packets");
 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);
 
 // Maps a codec type to a printable name string.
 std::string CodecName(webrtc::NetEqDecoder codec) {
   switch (codec) {
     case webrtc::kDecoderPCMu:
       return "PCM-u";
     case webrtc::kDecoderPCMa:
       return "PCM-a";
@@ skipping 227 matching lines @@
     if (FLAGS_codec_map) {
       // We have already printed the codec map. Just end the program.
       return 0;
     }
     // Print usage information.
     std::cout << google::ProgramUsage();
     return 0;
   }
 
   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.
   if (!FLAGS_ssrc.empty()) {
     uint32_t ssrc;
     CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
     file_source->SelectSsrc(ssrc);
   }
 
   // Check if a replacement audio file was provided, and if so, open it.
   bool replace_payload = false;
   rtc::scoped_ptr<webrtc::test::InputAudioFile> replacement_audio_file;
   if (!FLAGS_replacement_audio_file.empty()) {
     replacement_audio_file.reset(
         new webrtc::test::InputAudioFile(FLAGS_replacement_audio_file));
     replace_payload = true;
   }
 
   // Read first packet.
   rtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
   if (!packet) {
     printf(
         "Warning: input file is empty, or the filters did not match any "
         "packets\n");
     webrtc::Trace::ReturnTrace();
     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);
   if (sample_rate_hz <= 0) {
     printf("Warning: Invalid sample rate from RTP packet.\n");
     webrtc::Trace::ReturnTrace();
     return 0;
   }
 
   // Open the output file now that we know the sample rate. (Rate is only needed
@@ skipping 51 matching lines @@
   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;
   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) {
+  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);
       // Parse RTP header.
       WebRtcRTPHeader rtp_header;
       packet->ConvertHeader(&rtp_header);
       const uint8_t* payload_ptr = packet->payload();
       size_t payload_len = packet->payload_length_bytes();
       if (replace_payload) {
         payload_len = ReplacePayload(replacement_audio_file.get(),
@@ skipping 29 matching lines @@
                     << std::endl;
           std::cerr << "  TS = " << rtp_header.header.timestamp << std::endl;
         }
       }
 
       // Get next packet from file.
       webrtc::test::Packet* temp_packet = file_source->NextPacket();
       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|.
         // 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];
       int num_channels;
       int samples_per_channel;
       int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                    &num_channels, NULL);
       if (error != NetEq::kOK) {
         std::cerr << "GetAudio returned error code " <<
             neteq->LastError() << std::endl;
       } else {
         // Calculate sample rate from output size.
         sample_rate_hz = 1000 * samples_per_channel / kOutputBlockSizeMs;
       }
 
       // Write to file.
       // TODO(hlundin): Make writing to file optional.
       size_t write_len = samples_per_channel * num_channels;
       if (!output->WriteArray(out_data, write_len)) {
         std::cerr << "Error while writing to file" << std::endl;
         webrtc::Trace::ReturnTrace();
         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);
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;
 }