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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
81 lines
2.7 KiB
C++
81 lines
2.7 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
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#include "api/rtp_headers.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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namespace test {
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// Class for generating RTP headers.
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class RtpGenerator {
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public:
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RtpGenerator(int samples_per_ms,
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uint16_t start_seq_number = 0,
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uint32_t start_timestamp = 0,
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uint32_t start_send_time_ms = 0,
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uint32_t ssrc = 0x12345678)
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: seq_number_(start_seq_number),
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timestamp_(start_timestamp),
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next_send_time_ms_(start_send_time_ms),
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ssrc_(ssrc),
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samples_per_ms_(samples_per_ms),
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drift_factor_(0.0) {}
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virtual ~RtpGenerator() {}
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// Writes the next RTP header to |rtp_header|, which will be of type
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// |payload_type|. Returns the send time for this packet (in ms). The value of
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// |payload_length_samples| determines the send time for the next packet.
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virtual uint32_t GetRtpHeader(uint8_t payload_type,
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size_t payload_length_samples,
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RTPHeader* rtp_header);
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void set_drift_factor(double factor);
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protected:
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uint16_t seq_number_;
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uint32_t timestamp_;
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uint32_t next_send_time_ms_;
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const uint32_t ssrc_;
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const int samples_per_ms_;
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double drift_factor_;
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
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};
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class TimestampJumpRtpGenerator : public RtpGenerator {
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public:
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TimestampJumpRtpGenerator(int samples_per_ms,
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uint16_t start_seq_number,
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uint32_t start_timestamp,
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uint32_t jump_from_timestamp,
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uint32_t jump_to_timestamp)
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: RtpGenerator(samples_per_ms, start_seq_number, start_timestamp),
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jump_from_timestamp_(jump_from_timestamp),
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jump_to_timestamp_(jump_to_timestamp) {}
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uint32_t GetRtpHeader(uint8_t payload_type,
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size_t payload_length_samples,
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RTPHeader* rtp_header) override;
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private:
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uint32_t jump_from_timestamp_;
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uint32_t jump_to_timestamp_;
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RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
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