Move ownership of RTPSenderAudio to ChannelSend.
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}diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4e06002..6d1c9e2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -116,9 +116,7 @@
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 @@
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 @@
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 @@
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;