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RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs

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This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.

BUG=webrtc:8159

Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
This commit is contained in:
Karl Wiberg
2017-10-04 12:38:53 +02:00
committed by Commit Bot
parent 83ccca1864
commit c62f6c7121
22 changed files with 205 additions and 259 deletions
Download Patch File Download Diff File Expand all files Collapse all files

43
modules/rtp_rtcp/source/rtp_receiver_impl.cc
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@ -19,6 +19,7 @@
#include <vector>
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@ -57,6 +58,11 @@ RtpReceiver* RtpReceiver::CreateAudioReceiver(
RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
}
int32_t RtpReceiver::RegisterReceivePayload(const CodecInst& audio_codec) {
return RegisterReceivePayload(audio_codec.pltype,
CodecInstToSdp(audio_codec));
}
RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry,
@ -81,7 +87,9 @@ RtpReceiverImpl::~RtpReceiverImpl() {
}
}
int32_t RtpReceiverImpl::RegisterReceivePayload(const CodecInst& audio_codec) {
int32_t RtpReceiverImpl::RegisterReceivePayload(
int payload_type,
const SdpAudioFormat& audio_format) {
rtc::CritScope lock(&critical_section_rtp_receiver_);
// TODO(phoglund): Try to streamline handling of the RED codec and some other
@ -89,11 +97,12 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(const CodecInst& audio_codec) {
// payload or not.
bool created_new_payload = false;
int32_t result = rtp_payload_registry_->RegisterReceivePayload(
audio_codec, &created_new_payload);
payload_type, audio_format, &created_new_payload);
if (created_new_payload) {
if (rtp_media_receiver_->OnNewPayloadTypeCreated(audio_codec) != 0) {
LOG(LS_ERROR) << "Failed to register payload: " << audio_codec.plname
<< "/" << static_cast<int>(audio_codec.pltype);
if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_type,
audio_format) != 0) {
LOG(LS_ERROR) << "Failed to register payload: " << audio_format.name
<< "/" << payload_type;
return -1;
}
}
@ -240,10 +249,7 @@ bool RtpReceiverImpl::GetLatestTimestamps(uint32_t* timestamp,
// Implementation note: must not hold critsect when called.
void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
bool new_ssrc = false;
bool re_initialize_decoder = false;
char payload_name[RTP_PAYLOAD_NAME_SIZE];
size_t channels = 1;
uint32_t rate = 0;
rtc::Optional<AudioPayload> reinitialize_audio_payload;
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
@ -262,18 +268,16 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
if (ssrc_ != 0) {
// Do we have the same codec? Then re-initialize coder.
if (rtp_header.payloadType == last_received_payload_type) {
re_initialize_decoder = true;
const auto payload = rtp_payload_registry_->PayloadTypeToPayload(
rtp_header.payloadType);
if (!payload) {
return;
}
payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
if (payload->typeSpecific.is_audio()) {
channels = payload->typeSpecific.audio_payload().channels;
rate = payload->typeSpecific.audio_payload().rate;
reinitialize_audio_payload.emplace(
payload->typeSpecific.audio_payload());
} else {
// OnInitializeDecoder() is only used for audio.
}
}
}
@ -287,11 +291,10 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
cb_rtp_feedback_->OnIncomingSSRCChanged(rtp_header.ssrc);
}
if (re_initialize_decoder) {
if (-1 ==
cb_rtp_feedback_->OnInitializeDecoder(
rtp_header.payloadType, payload_name,
rtp_header.payload_type_frequency, channels, rate)) {
if (reinitialize_audio_payload) {
if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
rtp_header.payloadType, reinitialize_audio_payload->format,
reinitialize_audio_payload->rate)) {
// New stream, same codec.
LOG(LS_ERROR) << "Failed to create decoder for payload type: "
<< static_cast<int>(rtp_header.payloadType);