Receiver bit-exactness test for AudioCoding Module

This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.

The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d

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

@@ -41,6 +41,23 @@ class AudioSink {
   DISALLOW_COPY_AND_ASSIGN(AudioSink);
 };
 
+// Forks the output audio to two AudioSink objects.
+class AudioSinkFork : public AudioSink {
+ public:
+  AudioSinkFork(AudioSink* left, AudioSink* right)
+      : left_sink_(left), right_sink_(right) {}
+
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+    return left_sink_->WriteArray(audio, num_samples) &&
+           right_sink_->WriteArray(audio, num_samples);
+  }
+
+ private:
+  AudioSink* left_sink_;
+  AudioSink* right_sink_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
+};
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_

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

@@ -11,7 +11,10 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
 
+#include <bitset>
+
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace test {
@@ -28,6 +31,13 @@ class PacketSource {
   // depleted, or if an error occurred.
   virtual Packet* NextPacket() = 0;
 
+  virtual void FilterOutPayloadType(uint8_t payload_type) {
+    filter_.set(payload_type, true);
+  }
+
+ protected:
+  std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
+
  private:
   DISALLOW_COPY_AND_ASSIGN(PacketSource);
 };

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

@@ -92,6 +92,10 @@ Packet* RtpFileSource::NextPacket() {
       assert(false);
       return NULL;
     }
+    if (filter_.test(packet->header().payloadType)) {
+      // This payload type should be filtered out. Continue to the next packet.
+      continue;
+    }
     return packet.release();
   }
   return NULL;