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Split out RtpSource from libjingle_peerconnection_api

Browse Source 
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.

Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
This commit is contained in:
Niels Möller
2019-09-02 15:16:49 +02:00
committed by Commit Bot
parent d112c75801
commit a837030f8f
20 changed files with 124 additions and 76 deletions
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15
api/rtp_receiver_interface.cc
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@ -12,21 +12,6 @@
namespace webrtc {
RtpSource::RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
absl::optional<uint8_t> audio_level,
uint32_t rtp_timestamp)
: timestamp_ms_(timestamp_ms),
source_id_(source_id),
source_type_(source_type),
audio_level_(audio_level),
rtp_timestamp_(rtp_timestamp) {}
RtpSource::RtpSource(const RtpSource&) = default;
RtpSource& RtpSource::operator=(const RtpSource&) = default;
RtpSource::~RtpSource() = default;
std::vector<std::string> RtpReceiverInterface::stream_ids() const {
return {};
}