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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

50
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -116,9 +116,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
configuration.extmap_allow_mixed,
configuration.field_trials ? *configuration.field_trials
: default_trials));
if (configuration.audio) {
audio_ = absl::make_unique<RTPSenderAudio>(clock_, rtp_sender_.get());
}
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
@ -270,17 +268,6 @@ void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet,
rtcp_receiver_.IncomingPacket(rtcp_packet, length);
}
void ModuleRtpRtcpImpl::RegisterAudioSendPayload(int payload_type,
absl::string_view payload_name,
int frequency,
int channels,
int rate) {
RTC_DCHECK(audio_);
rtcp_sender_.SetRtpClockRate(payload_type, frequency);
RTC_CHECK_EQ(0, audio_->RegisterAudioPayload(payload_name, payload_type,
frequency, channels, rate));
}
void ModuleRtpRtcpImpl::RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) {
rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
@ -425,30 +412,6 @@ void ModuleRtpRtcpImpl::SetAsPartOfAllocation(bool part_of_allocation) {
rtp_sender_->SetAsPartOfAllocation(part_of_allocation);
}
bool ModuleRtpRtcpImpl::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) {
OnSendingRtpFrame(time_stamp, capture_time_ms, payload_type,
kVideoFrameKey == frame_type);
const uint32_t rtp_timestamp = time_stamp + rtp_sender_->TimestampOffset();
if (transport_frame_id_out)
*transport_frame_id_out = rtp_timestamp;
RTC_DCHECK(audio_);
RTC_DCHECK(fragmentation == nullptr);
return audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
payload_data, payload_size);
}
bool ModuleRtpRtcpImpl::OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
@ -787,17 +750,6 @@ bool ModuleRtpRtcpImpl::SendFeedbackPacket(
return rtcp_sender_.SendFeedbackPacket(packet);
}
// Send a TelephoneEvent tone using RFC 2833 (4733).
int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(const uint8_t key,
const uint16_t time_ms,
const uint8_t level) {
return audio_ ? audio_->SendTelephoneEvent(key, time_ms, level) : -1;
}
int32_t ModuleRtpRtcpImpl::SetAudioLevel(const uint8_t level_d_bov) {
return audio_ ? audio_->SetAudioLevel(level_d_bov) : -1;
}
int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
const KeyFrameRequestMethod method) {
key_frame_req_method_ = method;