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Move ownership of RTPSenderAudio to ChannelSend.

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This change takes out responsibility for packetization from the
RtpRtcp class, and deletes the method RtpRtcp::SendOutgoingData.

Video packetization was similarly moved in cl
https://webrtc-review.googlesource.com/c/src/+/123187

Bug: webrtc:7135
Change-Id: I0953125a5ca22a2ce51761b83693e0bb8ea74cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27000}
This commit is contained in:
Niels Möller
2019-03-06 16:47:29 +01:00
committed by Commit Bot
parent 232b3fda92
commit ee5ccbc57f
11 changed files with 57 additions and 164 deletions
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32
modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -32,8 +32,6 @@
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/gtest_prod_util.h"
@ -64,12 +62,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
void SetRemoteSSRC(uint32_t ssrc) override;
// Sender part.
void RegisterAudioSendPayload(int payload_type,
absl::string_view payload_name,
int frequency,
int channels,
int rate) override;
void RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) override;
@ -137,18 +129,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
void SetAsPartOfAllocation(bool part_of_allocation) override;
// Used by the codec module to deliver a video or audio frame for
// packetization.
bool SendOutgoingData(FrameType frame_type,
int8_t payload_type,
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_header,
uint32_t* transport_frame_id_out) override;
bool OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
@ -270,17 +250,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
bool RtcpXrRrtrStatus() const override;
// Audio part.
// Send a TelephoneEvent tone using RFC 2833 (4733).
int32_t SendTelephoneEventOutband(uint8_t key,
uint16_t time_ms,
uint8_t level) override;
// Store the audio level in d_bov for header-extension-for-audio-level-
// indication.
int32_t SetAudioLevel(uint8_t level_d_bov) override;
// Video part.
// Set method for requesting a new key frame.
@ -346,7 +315,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
bool TimeToSendFullNackList(int64_t now) const;
std::unique_ptr<RTPSender> rtp_sender_;
std::unique_ptr<RTPSenderAudio> audio_;
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;