Add wav output capability to neteq_rtpplay

This CL makes neteq_rtpplay capable of writing to wav files as well as
pcm files. This is done through the new class OutputWavFile, which
wraps a WavWriter object in an AudioSink interface.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7740 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/neteq.gypi

@@ -171,6 +171,7 @@
           'type': 'static_library',
           'dependencies': [
             'rtp_rtcp',
+            '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
             '<(webrtc_root)/test/test.gyp:rtp_test_utils',
           ],
           'direct_dependent_settings': {
@@ -191,6 +192,7 @@
             'tools/input_audio_file.cc',
             'tools/input_audio_file.h',
             'tools/output_audio_file.h',
+            'tools/output_wav_file.h',
             'tools/packet.cc',
             'tools/packet.h',
             'tools/packet_source.h',

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

@@ -26,6 +26,8 @@
 #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/rtp_file_source.h"
 #include "webrtc/modules/interface/module_common_types.h"
@@ -163,7 +165,7 @@ int main(int argc, char* argv[]) {
   std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
       "Run " + program_name + " --helpshort for usage.\n"
       "Example usage:\n" + program_name +
-      " input.rtp output.pcm\n";
+      " input.rtp output.{pcm, wav}\n";
   google::SetUsageMessage(usage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
@@ -193,13 +195,6 @@ int main(int argc, char* argv[]) {
     file_source->SelectSsrc(ssrc);
   }
 
-  FILE* out_file = fopen(argv[2], "wb");
-  if (!out_file) {
-    std::cerr << "Cannot open output file " << argv[2] << std::endl;
-    exit(1);
-  }
-  std::cout << "Output file: " << argv[2] << std::endl;
-
   // Check if a replacement audio file was provided, and if so, open it.
   bool replace_payload = false;
   webrtc::scoped_ptr<webrtc::test::InputAudioFile> replacement_audio_file;
@@ -209,19 +204,6 @@ int main(int argc, char* argv[]) {
     replace_payload = true;
   }
 
-  // Enable tracing.
-  webrtc::Trace::CreateTrace();
-  webrtc::Trace::SetTraceFile((webrtc::test::OutputPath() +
-      "neteq_trace.txt").c_str());
-  webrtc::Trace::set_level_filter(webrtc::kTraceAll);
-
-  // Initialize NetEq instance.
-  int sample_rate_hz = 16000;
-  NetEq::Config config;
-  config.sample_rate_hz = sample_rate_hz;
-  NetEq* neteq = NetEq::Create(config);
-  RegisterPayloadTypes(neteq);
-
   // Read first packet.
   webrtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
   if (!packet) {
@@ -233,6 +215,44 @@ int main(int argc, char* argv[]) {
   }
   bool packet_available = true;
 
+  // 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
+  // for wav files.)
+  // Check output file type.
+  std::string output_file_name = argv[2];
+  webrtc::scoped_ptr<webrtc::test::AudioSink> output;
+  if (output_file_name.size() >= 4 &&
+      output_file_name.substr(output_file_name.size() - 4) == ".wav") {
+    // Open a wav file.
+    output.reset(
+        new webrtc::test::OutputWavFile(output_file_name, sample_rate_hz));
+  } else {
+    // Open a pcm file.
+    output.reset(new webrtc::test::OutputAudioFile(output_file_name));
+  }
+
+  std::cout << "Output file: " << argv[2] << std::endl;
+
+  // Enable tracing.
+  webrtc::Trace::CreateTrace();
+  webrtc::Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "neteq_trace.txt").c_str());
+  webrtc::Trace::set_level_filter(webrtc::kTraceAll);
+
+  // Initialize NetEq instance.
+  NetEq::Config config;
+  config.sample_rate_hz = sample_rate_hz;
+  NetEq* neteq = NetEq::Create(config);
+  RegisterPayloadTypes(neteq);
+
+
   // Set up variables for audio replacement if needed.
   webrtc::scoped_ptr<webrtc::test::Packet> next_packet;
   bool next_packet_available = false;
@@ -349,8 +369,7 @@ int main(int argc, char* argv[]) {
       // Write to file.
       // TODO(hlundin): Make writing to file optional.
       size_t write_len = samples_per_channel * num_channels;
-      if (fwrite(out_data, sizeof(out_data[0]), write_len, out_file) !=
-          write_len) {
+      if (!output->WriteArray(out_data, write_len)) {
         std::cerr << "Error while writing to file" << std::endl;
         webrtc::Trace::ReturnTrace();
         exit(1);
@@ -363,7 +382,6 @@ int main(int argc, char* argv[]) {
 
   std::cout << "Simulation done" << std::endl;
 
-  fclose(out_file);
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;

webrtc/modules/audio_coding/neteq/tools/output_wav_file.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+class OutputWavFile : public AudioSink {
+ public:
+  // Creates an OutputWavFile, opening a file named |file_name| for writing.
+  // The output file is a PCM encoded wav file.
+  OutputWavFile(const std::string& file_name, int sample_rate_hz)
+      : wav_writer_(file_name, sample_rate_hz, 1) {}
+
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+    wav_writer_.WriteSamples(audio, num_samples);
+    return true;
+  }
+
+ private:
+  WavWriter wav_writer_;
+
+  DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_