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WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

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This removes one more place where we were unable to handle codecs not
in the built-in set.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
This commit is contained in:
kwiberg
2017-03-24 03:16:04 -07:00
committed by Commit bot
parent f137e97adb
commit 1724cfbdba
28 changed files with 221 additions and 150 deletions
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26
webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
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@ -16,6 +16,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
namespace webrtc {
@ -119,6 +120,31 @@ RTPPayloadRegistry::RTPPayloadRegistry()
RTPPayloadRegistry::~RTPPayloadRegistry() = default;
void RTPPayloadRegistry::SetAudioReceivePayloads(
std::map<int, SdpAudioFormat> codecs) {
rtc::CritScope cs(&crit_sect_);
#if RTC_DCHECK_IS_ON
RTC_DCHECK(!used_for_video_);
used_for_audio_ = true;
#endif
payload_type_map_.clear();
for (const auto& kv : codecs) {
const int& rtp_payload_type = kv.first;
const SdpAudioFormat& audio_format = kv.second;
const CodecInst ci = SdpToCodecInst(rtp_payload_type, audio_format);
RTC_DCHECK(IsPayloadTypeValid(rtp_payload_type));
payload_type_map_.insert(
std::make_pair(rtp_payload_type, CreatePayloadType(ci)));
}
// Clear the value of last received payload type since it might mean
// something else now.
last_received_payload_type_ = -1;
last_received_media_payload_type_ = -1;
}
int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
bool* created_new_payload) {
rtc::CritScope cs(&crit_sect_);