28deb90728
This is a reland of 3409cfa378e75c0c08d900e0848147929249a62b Needed to change RtpVideoStreamReceiver to stop deregistering a payload type if two payload types refer to the same codec (which now happens, with the packetization mode 0/1 payload types). It's not clear why this was being done in the first place. Original change's description: > Start supporting H264 packetization mode 0. > > The work was already done to support it, but it wasn't being negotiated > in SDP. > > This means we'll now see 8 H264 payload types instead of 4; one for each > combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX. > This could be problematic in the future, since we're starting to run > out of dynamic payload types (using 25 of 32). > > Bug: chromium:600254 > Change-Id: Ief2340db77c796f12980445b547b87e939170fae > Reviewed-on: https://webrtc-review.googlesource.com/77264 > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23372} Bug: chromium:600254 Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259 Reviewed-on: https://webrtc-review.googlesource.com/78399 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23494}
83 lines
2.6 KiB
C++
83 lines
2.6 KiB
C++
/*
|
|
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
|
|
#define MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
|
|
|
|
#include <map>
|
|
#include <set>
|
|
|
|
#include "api/audio_codecs/audio_format.h"
|
|
#include "api/optional.h"
|
|
#include "modules/rtp_rtcp/source/rtp_utility.h"
|
|
#include "rtc_base/criticalsection.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class VideoCodec;
|
|
|
|
class RTPPayloadRegistry {
|
|
public:
|
|
RTPPayloadRegistry();
|
|
~RTPPayloadRegistry();
|
|
|
|
// TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
|
|
// and simplify the code. http://crbug/webrtc/6743.
|
|
|
|
// Replace all audio receive payload types with the given map.
|
|
void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
|
|
|
|
int32_t RegisterReceivePayload(int payload_type,
|
|
const SdpAudioFormat& audio_format,
|
|
bool* created_new_payload_type);
|
|
int32_t RegisterReceivePayload(const VideoCodec& video_codec);
|
|
|
|
int32_t DeRegisterReceivePayload(int8_t payload_type);
|
|
|
|
int GetPayloadTypeFrequency(uint8_t payload_type) const;
|
|
|
|
rtc::Optional<RtpUtility::Payload> PayloadTypeToPayload(
|
|
uint8_t payload_type) const;
|
|
|
|
void ResetLastReceivedPayloadTypes() {
|
|
rtc::CritScope cs(&crit_sect_);
|
|
last_received_payload_type_ = -1;
|
|
}
|
|
|
|
int8_t last_received_payload_type() const {
|
|
rtc::CritScope cs(&crit_sect_);
|
|
return last_received_payload_type_;
|
|
}
|
|
void set_last_received_payload_type(int8_t last_received_payload_type) {
|
|
rtc::CritScope cs(&crit_sect_);
|
|
last_received_payload_type_ = last_received_payload_type;
|
|
}
|
|
|
|
private:
|
|
// Prunes the payload type map of the specific payload type, if it exists.
|
|
void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
|
|
const SdpAudioFormat& audio_format);
|
|
|
|
rtc::CriticalSection crit_sect_;
|
|
std::map<int, RtpUtility::Payload> payload_type_map_;
|
|
int8_t last_received_payload_type_;
|
|
|
|
// As a first step in splitting this class up in separate cases for audio and
|
|
// video, DCHECK that no instance is used for both audio and video.
|
|
#if RTC_DCHECK_IS_ON
|
|
bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false;
|
|
bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false;
|
|
#endif
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
|