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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
Download Patch File Download Diff File Expand all files Collapse all files

43
modules/rtp_rtcp/include/rtp_rtcp.h
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@ -146,11 +146,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
// FEC/ULP/RED overhead (when FEC is enabled).
virtual size_t MaxRtpPacketSize() const = 0;
virtual void RegisterAudioSendPayload(int payload_type,
absl::string_view payload_name,
int frequency,
int channels,
int rate) = 0;
virtual void RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) = 0;
@ -257,27 +252,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
virtual RTPSender* RtpSender() = 0;
virtual const RTPSender* RtpSender() const = 0;
// Used by the codec module to deliver a video or audio frame for
// packetization.
// |frame_type| - type of frame to send
// |payload_type| - payload type of frame to send
// |timestamp| - timestamp of frame to send
// |payload_data| - payload buffer of frame to send
// |payload_size| - size of payload buffer to send
// |fragmentation| - fragmentation offset data for fragmented frames such
// as layers or RED
// |transport_frame_id_out| - set to RTP timestamp.
// Returns true on success.
virtual bool SendOutgoingData(FrameType frame_type,
int8_t payload_type,
uint32_t timestamp,
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) = 0;
// Record that a frame is about to be sent. Returns true on success, and false
// if the module isn't ready to send.
virtual bool OnSendingRtpFrame(uint32_t timestamp,
@ -431,23 +405,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
virtual void SetVideoBitrateAllocation(
const VideoBitrateAllocation& bitrate) = 0;
// **************************************************************************
// Audio
// **************************************************************************
// Sends a TelephoneEvent tone using RFC 2833 (4733).
// Returns -1 on failure else 0.
virtual int32_t SendTelephoneEventOutband(uint8_t key,
uint16_t time_ms,
uint8_t level) = 0;
// Store the audio level in dBov for header-extension-for-audio-level-
// indication.
// This API shall be called before transmision of an RTP packet to ensure
// that the |level| part of the extended RTP header is updated.
// return -1 on failure else 0.
virtual int32_t SetAudioLevel(uint8_t level_dbov) = 0;
// **************************************************************************
// Video
// **************************************************************************