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      1 /*
      2  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
     13 
     14 #include <vector>
     15 
     16 #include "webrtc/base/constructormagic.h"
     17 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
     18 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
     19 #include "webrtc/system_wrappers/interface/clock.h"
     20 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
     21 
     22 namespace webrtc {
     23 
     24 namespace test {
     25 class InputAudioFile;
     26 class Packet;
     27 
     28 class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
     29  public:
     30   AcmSendTest(InputAudioFile* audio_source,
     31               int source_rate_hz,
     32               int test_duration_ms);
     33   virtual ~AcmSendTest() {}
     34 
     35   // Registers the send codec. Returns true on success, false otherwise.
     36   bool RegisterCodec(int codec_type,
     37                      int channels,
     38                      int payload_type,
     39                      int frame_size_samples);
     40 
     41   // Returns the next encoded packet. Returns NULL if the test duration was
     42   // exceeded. Ownership of the packet is handed over to the caller.
     43   // Inherited from PacketSource.
     44   virtual Packet* NextPacket() OVERRIDE;
     45 
     46   // Inherited from AudioPacketizationCallback.
     47   virtual int32_t SendData(
     48       FrameType frame_type,
     49       uint8_t payload_type,
     50       uint32_t timestamp,
     51       const uint8_t* payload_data,
     52       uint16_t payload_len_bytes,
     53       const RTPFragmentationHeader* fragmentation) OVERRIDE;
     54 
     55  private:
     56   static const int kBlockSizeMs = 10;
     57 
     58   // Creates a Packet object from the last packet produced by ACM (and received
     59   // through the SendData method as a callback). Ownership of the new Packet
     60   // object is transferred to the caller.
     61   Packet* CreatePacket();
     62 
     63   SimulatedClock clock_;
     64   scoped_ptr<AudioCoding> acm_;
     65   InputAudioFile* audio_source_;
     66   int source_rate_hz_;
     67   const int input_block_size_samples_;
     68   AudioFrame input_frame_;
     69   bool codec_registered_;
     70   int test_duration_ms_;
     71   // The following member variables are set whenever SendData() is called.
     72   FrameType frame_type_;
     73   int payload_type_;
     74   uint32_t timestamp_;
     75   uint16_t sequence_number_;
     76   std::vector<uint8_t> last_payload_vec_;
     77 
     78   DISALLOW_COPY_AND_ASSIGN(AcmSendTest);
     79 };
     80 
     81 }  // namespace test
     82 }  // namespace webrtc
     83 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
     84