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      1 /*
      2  *  Copyright (c) 2012 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_NETEQ_TOOLS_RTP_GENERATOR_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
     13 
     14 #include "webrtc/base/constructormagic.h"
     15 #include "webrtc/modules/interface/module_common_types.h"
     16 #include "webrtc/typedefs.h"
     17 
     18 namespace webrtc {
     19 namespace test {
     20 
     21 // Class for generating RTP headers.
     22 class RtpGenerator {
     23  public:
     24   RtpGenerator(int samples_per_ms,
     25                uint16_t start_seq_number = 0,
     26                uint32_t start_timestamp = 0,
     27                uint32_t start_send_time_ms = 0,
     28                uint32_t ssrc = 0x12345678)
     29       : seq_number_(start_seq_number),
     30         timestamp_(start_timestamp),
     31         next_send_time_ms_(start_send_time_ms),
     32         ssrc_(ssrc),
     33         samples_per_ms_(samples_per_ms),
     34         drift_factor_(0.0) {
     35   }
     36 
     37   virtual ~RtpGenerator() {}
     38 
     39   // Writes the next RTP header to |rtp_header|, which will be of type
     40   // |payload_type|. Returns the send time for this packet (in ms). The value of
     41   // |payload_length_samples| determines the send time for the next packet.
     42   virtual uint32_t GetRtpHeader(uint8_t payload_type,
     43                                 size_t payload_length_samples,
     44                                 WebRtcRTPHeader* rtp_header);
     45 
     46   void set_drift_factor(double factor);
     47 
     48  protected:
     49   uint16_t seq_number_;
     50   uint32_t timestamp_;
     51   uint32_t next_send_time_ms_;
     52   const uint32_t ssrc_;
     53   const int samples_per_ms_;
     54   double drift_factor_;
     55 
     56  private:
     57   DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
     58 };
     59 
     60 class TimestampJumpRtpGenerator : public RtpGenerator {
     61  public:
     62   TimestampJumpRtpGenerator(int samples_per_ms,
     63                             uint16_t start_seq_number,
     64                             uint32_t start_timestamp,
     65                             uint32_t jump_from_timestamp,
     66                             uint32_t jump_to_timestamp)
     67       : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp),
     68         jump_from_timestamp_(jump_from_timestamp),
     69         jump_to_timestamp_(jump_to_timestamp) {}
     70 
     71   uint32_t GetRtpHeader(uint8_t payload_type,
     72                         size_t payload_length_samples,
     73                         WebRtcRTPHeader* rtp_header) OVERRIDE;
     74 
     75  private:
     76   uint32_t jump_from_timestamp_;
     77   uint32_t jump_to_timestamp_;
     78   DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
     79 };
     80 
     81 }  // namespace test
     82 }  // namespace webrtc
     83 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
     84