Use PayloadTypePicker in WebRtcVoiceEngine
This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code.
Bug: webrtc:360058654
Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42989}
diff --git a/call/BUILD.gn b/call/BUILD.gn
index a74e8ac..f53d0bd 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -39,7 +39,6 @@
deps = [
":audio_sender_interface",
":payload_type",
- ":payload_type",
":receive_stream_interface",
":rtp_interfaces",
":video_stream_api",
diff --git a/call/call.cc b/call/call.cc
index cf94cb8..2b77517 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -114,6 +114,11 @@
cricket::Codec codec) override {
return payload_type_picker_.SuggestMapping(codec, nullptr);
}
+ RTCError AddLocalMapping(const std::string& mid,
+ PayloadType payload_type,
+ const cricket::Codec& codec) override {
+ return RTCError::OK();
+ }
private:
PayloadTypePicker payload_type_picker_;
diff --git a/call/payload_type.h b/call/payload_type.h
index 428096b..c0520e8 100644
--- a/call/payload_type.h
+++ b/call/payload_type.h
@@ -37,6 +37,11 @@
// or a newly suggested one.
virtual RTCErrorOr<PayloadType> SuggestPayloadType(const std::string& mid,
cricket::Codec codec) = 0;
+ // Register a payload type as mapped to a specific codec for this MID
+ // at this time.
+ virtual RTCError AddLocalMapping(const std::string& mid,
+ PayloadType payload_type,
+ const cricket::Codec& codec) = 0;
};
} // namespace webrtc
diff --git a/call/payload_type_picker.cc b/call/payload_type_picker.cc
index 4c5936b..765d716 100644
--- a/call/payload_type_picker.cc
+++ b/call/payload_type_picker.cc
@@ -136,6 +136,14 @@
RTCErrorOr<PayloadType> PayloadTypePicker::SuggestMapping(
cricket::Codec codec,
const PayloadTypeRecorder* excluder) {
+ // Test compatibility: If the codec contains a PT, and it is free, use it.
+ // This saves having to rewrite tests that set the codec ID themselves.
+ // Codecs with unassigned IDs should have -1 as their id.
+ if (codec.id >= 0 && codec.id <= kLastDynamicPayloadTypeUpperRange &&
+ seen_payload_types_.count(PayloadType(codec.id)) == 0) {
+ AddMapping(PayloadType(codec.id), codec);
+ return PayloadType(codec.id);
+ }
// The first matching entry is returned, unless excluder
// maps it to something different.
for (auto entry : entries_) {
@@ -149,6 +157,7 @@
return entry.payload_type();
}
}
+ // Assign the first free payload type.
RTCErrorOr<PayloadType> found_pt = FindFreePayloadType(seen_payload_types_);
if (found_pt.ok()) {
AddMapping(found_pt.value(), codec);
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 32af232..74d935d 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -344,6 +344,8 @@
"../rtc_base:stringutils",
"../rtc_base/network:sent_packet",
"../video/config:encoder_config",
+ "//third_party/abseil-cpp/absl/functional:any_invocable",
+ "//third_party/abseil-cpp/absl/strings:string_view",
]
}
@@ -592,6 +594,8 @@
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
+ "../call:payload_type",
+ "../call:payload_type_picker",
"../call:receive_stream_interface",
"../call:rtp_interfaces",
"../call:video_stream_api",
@@ -652,8 +656,6 @@
sources = [
"engine/adm_helpers.cc",
"engine/adm_helpers.h",
- "engine/payload_type_mapper.cc",
- "engine/payload_type_mapper.h",
"engine/webrtc_media_engine.cc",
"engine/webrtc_media_engine.h",
"engine/webrtc_video_engine.cc",
@@ -786,12 +788,23 @@
":video_common",
"../api:call_api",
"../api:fec_controller_api",
+ "../api:frame_transformer_interface",
+ "../api:rtc_error",
+ "../api:rtp_headers",
"../api:rtp_parameters",
+ "../api:rtp_sender_interface",
"../api:scoped_refptr",
+ "../api/adaptation:resource_adaptation_api",
+ "../api/audio:audio_frame_api",
"../api/audio:audio_processing",
+ "../api/audio_codecs:audio_codecs_api",
+ "../api/crypto:frame_decryptor_interface",
"../api/environment",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
+ "../api/transport:bitrate_settings",
+ "../api/transport/rtp:rtp_source",
+ "../api/units:time_delta",
"../api/units:timestamp",
"../api/video:encoded_image",
"../api/video:video_bitrate_allocation",
@@ -800,9 +813,13 @@
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../call:mock_rtp_interfaces",
+ "../call:payload_type",
+ "../call:payload_type_picker",
+ "../call:rtp_interfaces",
"../call:video_stream_api",
"../common_video",
"../modules/audio_processing",
+ "../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:video_coding_utility",
@@ -820,11 +837,13 @@
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:timeutils",
+ "../rtc_base/network:sent_packet",
"../rtc_base/synchronization:mutex",
"../rtc_base/third_party/sigslot",
"../test:explicit_key_value_config",
"../test:scoped_key_value_config",
"../test:test_support",
+ "../video/config:encoder_config",
"../video/config:streams_config",
"//testing/gtest",
"//third_party/abseil-cpp/absl/algorithm:container",
@@ -882,6 +901,7 @@
":rtc_audio_video",
":rtc_internal_video_codecs",
":rtc_media",
+ ":rtc_media_config",
":rtc_media_tests_utils",
":rtc_sdp_video_format_utils",
":rtc_simulcast_encoder_adapter",
@@ -891,27 +911,38 @@
":video_adapter",
":video_broadcaster",
":video_common",
+ "../api:audio_options_api",
+ "../api:call_api",
"../api:create_simulcast_test_fixture_api",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
+ "../api:make_ref_counted",
"../api:mock_encoder_selector",
"../api:mock_video_bitrate_allocator",
"../api:mock_video_bitrate_allocator_factory",
"../api:mock_video_codec_factory",
"../api:mock_video_encoder",
"../api:priority",
+ "../api:ref_count",
+ "../api:rtc_error",
+ "../api:rtp_headers",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:simulcast_test_fixture_api",
+ "../api/audio:audio_processing",
+ "../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
+ "../api/crypto:options",
"../api/environment",
"../api/environment:environment_factory",
"../api/rtc_event_log",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
+ "../api/transport:bitrate_settings",
"../api/transport:field_trial_based_config",
+ "../api/transport/rtp:rtp_source",
"../api/units:time_delta",
"../api/units:timestamp",
"../api/video:builtin_video_bitrate_allocator_factory",
@@ -950,6 +981,8 @@
"../rtc_base:async_packet_socket",
"../rtc_base:byte_order",
"../rtc_base:checks",
+ "../rtc_base:copy_on_write_buffer",
+ "../rtc_base:dscp",
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../rtc_base:macromagic",
@@ -995,7 +1028,6 @@
"base/video_common_unittest.cc",
"engine/internal_decoder_factory_unittest.cc",
"engine/internal_encoder_factory_unittest.cc",
- "engine/payload_type_mapper_unittest.cc",
"engine/simulcast_encoder_adapter_unittest.cc",
"engine/webrtc_media_engine_unittest.cc",
"engine/webrtc_video_engine_unittest.cc",
diff --git a/media/DEPS b/media/DEPS
index bcdbcac..ffa3c11 100644
--- a/media/DEPS
+++ b/media/DEPS
@@ -34,8 +34,7 @@
".*fake_webrtc_call\.cc": [
"+video/config",
],
- # temporary
- ".*webrtc_voice_engine\.cc": [
- "+pc/payload_type_picker.h",
+ ".*fake_webrtc_call\.h": [
+ "+video/config",
],
}
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index b7f5987..73d2b25 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -11,6 +11,9 @@
#ifndef MEDIA_BASE_MEDIA_CHANNEL_H_
#define MEDIA_BASE_MEDIA_CHANNEL_H_
+#include <cstddef>
+#include <cstdint>
+#include <functional>
#include <map>
#include <memory>
#include <optional>
@@ -19,6 +22,8 @@
#include <utility>
#include <vector>
+#include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_options.h"
@@ -27,37 +32,34 @@
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
+#include "api/media_types.h"
#include "api/rtc_error.h"
+#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
-#include "api/task_queue/pending_task_safety_flag.h"
-#include "api/transport/data_channel_transport_interface.h"
+#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
+#include "api/video/recordable_encoded_frame.h"
#include "api/video/video_content_type.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video/video_timing.h"
#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/video_encoder_factory.h"
-#include "call/video_receive_stream.h"
#include "common_video/include/quality_limitation_reason.h"
#include "media/base/codec.h"
-#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/async_packet_socket.h"
-#include "rtc_base/buffer.h"
#include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/dscp.h"
-#include "rtc_base/logging.h"
+#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
-#include "video/config/video_encoder_config.h"
namespace rtc {
class Timing;
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 64baab7..c4e5f61 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -20,24 +20,56 @@
#ifndef MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
#define MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
+#include <cstddef>
+#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
+#include "api/adaptation/resource.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/crypto/frame_decryptor_interface.h"
#include "api/environment/environment.h"
+#include "api/frame_transformer_interface.h"
+#include "api/media_types.h"
+#include "api/rtc_error.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_sender_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/transport/bitrate_settings.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "api/video_codecs/video_codec.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
#include "call/flexfec_receive_stream.h"
+#include "call/packet_receiver.h"
+#include "call/payload_type.h"
+#include "call/payload_type_picker.h"
+#include "call/rtp_transport_controller_send_interface.h"
#include "call/test/mock_rtp_transport_controller_send.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
+#include "media/base/codec.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/buffer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/network/sent_packet.h"
+#include "test/gmock.h"
+#include "video/config/video_encoder_config.h"
namespace cricket {
class FakeAudioSendStream final : public webrtc::AudioSendStream {
@@ -363,6 +395,26 @@
webrtc::FlexfecReceiveStream::Config config_;
};
+// Fake payload type suggester.
+// This is injected into FakeCall at initialization.
+class FakePayloadTypeSuggester : public webrtc::PayloadTypeSuggester {
+ public:
+ webrtc::RTCErrorOr<webrtc::PayloadType> SuggestPayloadType(
+ const std::string& mid,
+ cricket::Codec codec) override {
+ // Ignores mid argument.
+ return pt_picker_.SuggestMapping(codec, nullptr);
+ }
+ webrtc::RTCError AddLocalMapping(const std::string& mid,
+ webrtc::PayloadType payload_type,
+ const cricket::Codec& codec) override {
+ return webrtc::RTCError::OK();
+ }
+
+ private:
+ webrtc::PayloadTypePicker pt_picker_;
+};
+
class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
public:
explicit FakeCall(const webrtc::Environment& env);
@@ -371,6 +423,10 @@
webrtc::TaskQueueBase* network_thread);
~FakeCall() override;
+ webrtc::PayloadTypeSuggester* GetPayloadTypeSuggester() {
+ return &pt_suggester_;
+ }
+
webrtc::MockRtpTransportControllerSend* GetMockTransportControllerSend() {
return &transport_controller_send_;
}
@@ -501,6 +557,8 @@
int num_created_send_streams_;
int num_created_receive_streams_;
+
+ FakePayloadTypeSuggester pt_suggester_;
};
} // namespace cricket
diff --git a/media/engine/payload_type_mapper.cc b/media/engine/payload_type_mapper.cc
deleted file mode 100644
index 9db2281..0000000
--- a/media/engine/payload_type_mapper.cc
+++ /dev/null
@@ -1,160 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "media/engine/payload_type_mapper.h"
-
-#include <utility>
-
-#include "absl/strings/ascii.h"
-#include "api/audio_codecs/audio_format.h"
-#include "media/base/codec.h"
-#include "media/base/media_constants.h"
-
-namespace cricket {
-
-webrtc::SdpAudioFormat AudioCodecToSdpAudioFormat(const Codec& ac) {
- return webrtc::SdpAudioFormat(ac.name, ac.clockrate, ac.channels, ac.params);
-}
-
-PayloadTypeMapper::PayloadTypeMapper()
- // RFC 3551 reserves payload type numbers in the range 96-127 exclusively
- // for dynamic assignment. Once those are used up, it is recommended that
- // payload types unassigned by the RFC are used for dynamic payload type
- // mapping, before any static payload ids. At this point, we only support
- // mapping within the exclusive range.
- : next_unused_payload_type_(96),
- max_payload_type_(127),
- mappings_(
- {// Static payload type assignments according to RFC 3551.
- {{kPcmuCodecName, 8000, 1}, 0},
- {{"GSM", 8000, 1}, 3},
- {{"G723", 8000, 1}, 4},
- {{"DVI4", 8000, 1}, 5},
- {{"DVI4", 16000, 1}, 6},
- {{"LPC", 8000, 1}, 7},
- {{kPcmaCodecName, 8000, 1}, 8},
- {{kG722CodecName, 8000, 1}, 9},
- {{kL16CodecName, 44100, 2}, 10},
- {{kL16CodecName, 44100, 1}, 11},
- {{"QCELP", 8000, 1}, 12},
- {{kCnCodecName, 8000, 1}, 13},
- // RFC 4566 is a bit ambiguous on the contents of the "encoding
- // parameters" field, which, for audio, encodes the number of
- // channels. It is "optional and may be omitted if the number of
- // channels is one". Does that necessarily imply that an omitted
- // encoding parameter means one channel? Since RFC 3551 doesn't
- // specify a value for this parameter for MPA, I've included both 0
- // and 1 here, to increase the chances it will be correctly used if
- // someone implements an MPEG audio encoder/decoder.
- {{"MPA", 90000, 0}, 14},
- {{"MPA", 90000, 1}, 14},
- {{"G728", 8000, 1}, 15},
- {{"DVI4", 11025, 1}, 16},
- {{"DVI4", 22050, 1}, 17},
- {{"G729", 8000, 1}, 18},
-
- // Payload type assignments currently used by WebRTC.
- // Includes data to reduce collisions (and thus reassignments)
- {{kIlbcCodecName, 8000, 1}, 102},
- {{kCnCodecName, 16000, 1}, 105},
- {{kCnCodecName, 32000, 1}, 106},
- {{kOpusCodecName,
- 48000,
- 2,
- {{kCodecParamMinPTime, "10"},
- {kCodecParamUseInbandFec, kParamValueTrue}}},
- 111},
- // RED for opus is assigned in the lower range, starting at the top.
- // Note that the FMTP refers to the opus payload type.
- {{kRedCodecName,
- 48000,
- 2,
- {{kCodecParamNotInNameValueFormat, "111/111"}}},
- 63},
- // TODO(solenberg): Remove the hard coded 16k,32k,48k DTMF once we
- // assign payload types dynamically for send side as well.
- {{kDtmfCodecName, 48000, 1}, 110},
- {{kDtmfCodecName, 32000, 1}, 112},
- {{kDtmfCodecName, 16000, 1}, 113},
- {{kDtmfCodecName, 8000, 1}, 126}}) {
- // TODO(ossu): Try to keep this as change-proof as possible until we're able
- // to remove the payload type constants from everywhere in the code.
- for (const auto& mapping : mappings_) {
- used_payload_types_.insert(mapping.second);
- }
-}
-
-PayloadTypeMapper::~PayloadTypeMapper() = default;
-
-std::optional<int> PayloadTypeMapper::GetMappingFor(
- const webrtc::SdpAudioFormat& format) {
- auto iter = mappings_.find(format);
- if (iter != mappings_.end())
- return iter->second;
-
- for (; next_unused_payload_type_ <= max_payload_type_;
- ++next_unused_payload_type_) {
- int payload_type = next_unused_payload_type_;
- if (used_payload_types_.find(payload_type) == used_payload_types_.end()) {
- used_payload_types_.insert(payload_type);
- mappings_[format] = payload_type;
- ++next_unused_payload_type_;
- return payload_type;
- }
- }
-
- return std::nullopt;
-}
-
-std::optional<int> PayloadTypeMapper::FindMappingFor(
- const webrtc::SdpAudioFormat& format) const {
- auto iter = mappings_.find(format);
- if (iter != mappings_.end())
- return iter->second;
-
- return std::nullopt;
-}
-
-std::optional<Codec> PayloadTypeMapper::ToAudioCodec(
- const webrtc::SdpAudioFormat& format) {
- // TODO(ossu): We can safely set bitrate to zero here, since that field is
- // not presented in the SDP. It is used to ferry around some target bitrate
- // values for certain codecs (ISAC and Opus) and in ways it really
- // shouldn't. It should be removed once we no longer use CodecInsts in the
- // ACM or NetEq.
- auto opt_payload_type = GetMappingFor(format);
- if (opt_payload_type) {
- Codec codec =
- cricket::CreateAudioCodec(*opt_payload_type, format.name,
- format.clockrate_hz, format.num_channels);
- codec.params = format.parameters;
- return std::move(codec);
- }
-
- return std::nullopt;
-}
-
-bool PayloadTypeMapper::SdpAudioFormatOrdering::operator()(
- const webrtc::SdpAudioFormat& a,
- const webrtc::SdpAudioFormat& b) const {
- if (a.clockrate_hz == b.clockrate_hz) {
- if (a.num_channels == b.num_channels) {
- int name_cmp =
- absl::AsciiStrToLower(a.name).compare(absl::AsciiStrToLower(b.name));
- if (name_cmp == 0)
- return a.parameters < b.parameters;
- return name_cmp < 0;
- }
- return a.num_channels < b.num_channels;
- }
- return a.clockrate_hz < b.clockrate_hz;
-}
-
-} // namespace cricket
diff --git a/media/engine/payload_type_mapper.h b/media/engine/payload_type_mapper.h
deleted file mode 100644
index 246b04e..0000000
--- a/media/engine/payload_type_mapper.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2016 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 MEDIA_ENGINE_PAYLOAD_TYPE_MAPPER_H_
-#define MEDIA_ENGINE_PAYLOAD_TYPE_MAPPER_H_
-
-#include <map>
-#include <optional>
-#include <set>
-
-#include "api/audio_codecs/audio_format.h"
-#include "media/base/codec.h"
-
-namespace cricket {
-
-webrtc::SdpAudioFormat AudioCodecToSdpAudioFormat(const Codec& ac);
-
-class PayloadTypeMapper {
- public:
- PayloadTypeMapper();
- ~PayloadTypeMapper();
-
- // Finds the current payload type for `format` or assigns a new one, if no
- // current mapping exists. Will return an empty value if it was unable to
- // create a mapping, i.e. if all dynamic payload type ids have been used up.
- std::optional<int> GetMappingFor(const webrtc::SdpAudioFormat& format);
-
- // Finds the current payload type for `format`, if any. Returns an empty value
- // if no payload type mapping exists for the format.
- std::optional<int> FindMappingFor(const webrtc::SdpAudioFormat& format) const;
-
- // Like GetMappingFor, but fills in an AudioCodec structure with the necessary
- // information instead.
- std::optional<Codec> ToAudioCodec(const webrtc::SdpAudioFormat& format);
-
- private:
- struct SdpAudioFormatOrdering {
- bool operator()(const webrtc::SdpAudioFormat& a,
- const webrtc::SdpAudioFormat& b) const;
- };
-
- int next_unused_payload_type_;
- int max_payload_type_;
- std::map<webrtc::SdpAudioFormat, int, SdpAudioFormatOrdering> mappings_;
- std::set<int> used_payload_types_;
-};
-
-} // namespace cricket
-#endif // MEDIA_ENGINE_PAYLOAD_TYPE_MAPPER_H_
diff --git a/media/engine/payload_type_mapper_unittest.cc b/media/engine/payload_type_mapper_unittest.cc
deleted file mode 100644
index 62cca1e..0000000
--- a/media/engine/payload_type_mapper_unittest.cc
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "media/engine/payload_type_mapper.h"
-
-#include <set>
-#include <string>
-
-#include "absl/strings/string_view.h"
-#include "media/base/media_constants.h"
-#include "test/gmock.h"
-#include "test/gtest.h"
-
-namespace cricket {
-
-class PayloadTypeMapperTest : public ::testing::Test {
- protected:
- PayloadTypeMapper mapper_;
-};
-
-TEST_F(PayloadTypeMapperTest, StaticPayloadTypes) {
- EXPECT_EQ(0, mapper_.FindMappingFor({"pcmu", 8000, 1}));
- EXPECT_EQ(3, mapper_.FindMappingFor({"gsm", 8000, 1}));
- EXPECT_EQ(4, mapper_.FindMappingFor({"g723", 8000, 1}));
- EXPECT_EQ(5, mapper_.FindMappingFor({"dvi4", 8000, 1}));
- EXPECT_EQ(6, mapper_.FindMappingFor({"dvi4", 16000, 1}));
- EXPECT_EQ(7, mapper_.FindMappingFor({"lpc", 8000, 1}));
- EXPECT_EQ(8, mapper_.FindMappingFor({"pcma", 8000, 1}));
- EXPECT_EQ(9, mapper_.FindMappingFor({"g722", 8000, 1}));
- EXPECT_EQ(10, mapper_.FindMappingFor({"l16", 44100, 2}));
- EXPECT_EQ(11, mapper_.FindMappingFor({"l16", 44100, 1}));
- EXPECT_EQ(12, mapper_.FindMappingFor({"qcelp", 8000, 1}));
- EXPECT_EQ(13, mapper_.FindMappingFor({"cn", 8000, 1}));
- EXPECT_EQ(14, mapper_.FindMappingFor({"mpa", 90000, 0}));
- EXPECT_EQ(14, mapper_.FindMappingFor({"mpa", 90000, 1}));
- EXPECT_EQ(15, mapper_.FindMappingFor({"g728", 8000, 1}));
- EXPECT_EQ(16, mapper_.FindMappingFor({"dvi4", 11025, 1}));
- EXPECT_EQ(17, mapper_.FindMappingFor({"dvi4", 22050, 1}));
- EXPECT_EQ(18, mapper_.FindMappingFor({"g729", 8000, 1}));
-}
-
-TEST_F(PayloadTypeMapperTest, WebRTCPayloadTypes) {
- // Tests that the payload mapper knows about the audio formats we've
- // been using in WebRTC, with their hard coded values.
- EXPECT_EQ(102, mapper_.FindMappingFor({kIlbcCodecName, 8000, 1}));
- EXPECT_EQ(105, mapper_.FindMappingFor({kCnCodecName, 16000, 1}));
- EXPECT_EQ(106, mapper_.FindMappingFor({kCnCodecName, 32000, 1}));
- EXPECT_EQ(111, mapper_.FindMappingFor(
- {kOpusCodecName,
- 48000,
- 2,
- {{"minptime", "10"}, {"useinbandfec", "1"}}}));
- EXPECT_EQ(
- 63, mapper_.FindMappingFor({kRedCodecName, 48000, 2, {{"", "111/111"}}}));
- // TODO(solenberg): Remove 16k, 32k, 48k DTMF checks once these payload types
- // are dynamically assigned.
- EXPECT_EQ(110, mapper_.FindMappingFor({kDtmfCodecName, 48000, 1}));
- EXPECT_EQ(112, mapper_.FindMappingFor({kDtmfCodecName, 32000, 1}));
- EXPECT_EQ(113, mapper_.FindMappingFor({kDtmfCodecName, 16000, 1}));
- EXPECT_EQ(126, mapper_.FindMappingFor({kDtmfCodecName, 8000, 1}));
-}
-
-TEST_F(PayloadTypeMapperTest, ValidDynamicPayloadTypes) {
- // RFC 3551 says:
- // "This profile reserves payload type numbers in the range 96-127
- // exclusively for dynamic assignment. Applications SHOULD first use
- // values in this range for dynamic payload types. Those applications
- // which need to define more than 32 dynamic payload types MAY bind
- // codes below 96, in which case it is RECOMMENDED that unassigned
- // payload type numbers be used first. However, the statically assigned
- // payload types are default bindings and MAY be dynamically bound to
- // new encodings if needed."
-
- // Tests that the payload mapper uses values in the dynamic payload type range
- // (96 - 127) before any others and that the values returned are all valid.
- bool has_been_below_96 = false;
- std::set<int> used_payload_types;
- for (int i = 0; i != 256; ++i) {
- std::string format_name = "unknown_format_" + std::to_string(i);
- webrtc::SdpAudioFormat format(format_name.c_str(), i * 100, (i % 2) + 1);
- auto opt_payload_type = mapper_.GetMappingFor(format);
- bool mapper_is_full = false;
-
- // There's a limited number of slots for payload types. We're fine with not
- // being able to map them all.
- if (opt_payload_type) {
- int payload_type = *opt_payload_type;
- EXPECT_FALSE(mapper_is_full) << "Mapping should not fail sporadically";
- EXPECT_EQ(used_payload_types.find(payload_type), used_payload_types.end())
- << "Payload types must not be reused";
- used_payload_types.insert(payload_type);
- EXPECT_GE(payload_type, 0) << "Negative payload types are invalid";
- EXPECT_LE(payload_type, 127) << "Payload types above 127 are invalid";
- EXPECT_FALSE(payload_type >= 96 && has_been_below_96);
- if (payload_type < 96)
- has_been_below_96 = true;
-
- EXPECT_EQ(payload_type, mapper_.FindMappingFor(format))
- << "Mapping must be permanent after successful call to "
- "GetMappingFor";
- EXPECT_EQ(payload_type, mapper_.GetMappingFor(format))
- << "Subsequent calls to GetMappingFor must return the same value";
- } else {
- mapper_is_full = true;
- }
- }
-
- // Also, we must've been able to map at least one dynamic payload type.
- EXPECT_FALSE(used_payload_types.empty())
- << "Mapper must support at least one user-defined payload type";
-}
-
-TEST_F(PayloadTypeMapperTest, ToAudioCodec) {
- webrtc::SdpAudioFormat format("unknown_format", 4711, 17);
- auto opt_payload_type = mapper_.GetMappingFor(format);
- EXPECT_TRUE(opt_payload_type);
- auto opt_audio_codec = mapper_.ToAudioCodec(format);
- EXPECT_TRUE(opt_audio_codec);
-
- if (opt_payload_type && opt_audio_codec) {
- int payload_type = *opt_payload_type;
- const Codec& codec = *opt_audio_codec;
-
- EXPECT_EQ(codec.id, payload_type);
- EXPECT_EQ(codec.name, format.name);
- EXPECT_EQ(codec.clockrate, format.clockrate_hz);
- EXPECT_EQ(codec.channels, format.num_channels);
- EXPECT_THAT(codec.params, ::testing::ContainerEq(format.parameters));
- }
-}
-
-} // namespace cricket
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index e65d0eb..93533da 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -12,54 +12,77 @@
#include <algorithm>
#include <atomic>
+#include <cstddef>
#include <cstdint>
#include <functional>
#include <initializer_list>
#include <iterator>
+#include <map>
#include <memory>
#include <optional>
+#include <set>
#include <string>
-#include <type_traits>
#include <utility>
#include <vector>
#include "absl/algorithm/algorithm.h"
#include "absl/algorithm/container.h"
+#include "absl/functional/any_invocable.h"
#include "absl/functional/bind_front.h"
#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
#include "api/audio/audio_frame.h"
#include "api/audio/audio_frame_processor.h"
+#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_options.h"
#include "api/call/audio_sink.h"
+#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
#include "api/field_trials_view.h"
+#include "api/frame_transformer_interface.h"
#include "api/make_ref_counted.h"
#include "api/media_types.h"
#include "api/priority.h"
+#include "api/rtc_error.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
+#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_direction.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/bitrate_settings.h"
+#include "api/transport/rtp/rtp_source.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "call/audio_receive_stream.h"
+#include "call/audio_send_stream.h"
+#include "call/audio_state.h"
+#include "call/call.h"
#include "call/packet_receiver.h"
+#include "call/payload_type_picker.h"
#include "call/rtp_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "media/base/audio_source.h"
#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_config.h"
#include "media/base/media_constants.h"
+#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
#include "media/engine/adm_helpers.h"
-#include "media/engine/payload_type_mapper.h"
#include "media/engine/webrtc_media_engine.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
-#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
@@ -72,6 +95,7 @@
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/strings/string_format.h"
+#include "rtc_base/system/file_wrapper.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
@@ -332,6 +356,91 @@
return true;
}
+webrtc::SdpAudioFormat AudioCodecToSdpAudioFormat(const Codec& ac) {
+ return webrtc::SdpAudioFormat(ac.name, ac.clockrate, ac.channels, ac.params);
+}
+
+// Assign the payload types for the codecs of this voice engine.
+// This is a "preliminary" pass, done to prime the
+// payload type picker with a normal set of PTs.
+// TODO: https://issues.webrtc.org/360058654 - remove.
+std::vector<Codec> LegacyCollectCodecs(
+ const std::vector<webrtc::AudioCodecSpec>& specs) {
+ // Once payload assignment is done in the channel not in the factory,
+ // this needs to be a variable. But then this function should go away.
+ webrtc::PayloadTypePicker pt_mapper;
+ std::vector<Codec> out;
+
+ // Only generate CN payload types for these clockrates:
+ std::map<int, bool, std::greater<int>> generate_cn = {
+ {8000, false}, {16000, false}, {32000, false}};
+ // Only generate telephone-event payload types for these clockrates:
+ std::map<int, bool, std::greater<int>> generate_dtmf = {
+ {8000, false}, {16000, false}, {32000, false}, {48000, false}};
+
+ for (const auto& spec : specs) {
+ cricket::Codec codec = CreateAudioCodec(spec.format);
+ auto pt_or_error = pt_mapper.SuggestMapping(codec, nullptr);
+ // We need to do some extra stuff before adding the main codecs to out.
+ if (pt_or_error.ok()) {
+ codec.id = pt_or_error.value();
+ if (spec.info.supports_network_adaption) {
+ codec.AddFeedbackParam(
+ FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
+ }
+
+ if (spec.info.allow_comfort_noise) {
+ // Generate a CN entry if the decoder allows it and we support the
+ // clockrate.
+ auto cn = generate_cn.find(spec.format.clockrate_hz);
+ if (cn != generate_cn.end()) {
+ cn->second = true;
+ }
+ }
+
+ // Generate a telephone-event entry if we support the clockrate.
+ auto dtmf = generate_dtmf.find(spec.format.clockrate_hz);
+ if (dtmf != generate_dtmf.end()) {
+ dtmf->second = true;
+ }
+
+ out.push_back(codec);
+
+ // TODO(hta): Don't assign RED codecs until we know that the PT for Opus
+ // is final
+ if (codec.name == kOpusCodecName) {
+ std::string red_fmtp =
+ rtc::ToString(codec.id) + "/" + rtc::ToString(codec.id);
+ cricket::Codec red_codec =
+ CreateAudioCodec({kRedCodecName, 48000, 2, {{"", red_fmtp}}});
+ red_codec.id = pt_mapper.SuggestMapping(red_codec, nullptr).value();
+ out.push_back(red_codec);
+ }
+ }
+ }
+
+ // Add CN codecs after "proper" audio codecs.
+ for (const auto& cn : generate_cn) {
+ if (cn.second) {
+ cricket::Codec cn_codec = CreateAudioCodec({kCnCodecName, cn.first, 1});
+ cn_codec.id = pt_mapper.SuggestMapping(cn_codec, nullptr).value();
+ out.push_back(cn_codec);
+ }
+ }
+
+ // Add telephone-event codecs last.
+ for (const auto& dtmf : generate_dtmf) {
+ if (dtmf.second) {
+ cricket::Codec dtmf_codec =
+ CreateAudioCodec({kDtmfCodecName, dtmf.first, 1});
+ dtmf_codec.id = pt_mapper.SuggestMapping(dtmf_codec, nullptr).value();
+ out.push_back(dtmf_codec);
+ }
+ }
+
+ return out;
+}
+
} // namespace
WebRtcVoiceEngine::WebRtcVoiceEngine(
@@ -383,13 +492,13 @@
// Load our audio codec lists.
RTC_LOG(LS_VERBOSE) << "Supported send codecs in order of preference:";
- send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders());
+ send_codecs_ = LegacyCollectCodecs(encoder_factory_->GetSupportedEncoders());
for (const Codec& codec : send_codecs_) {
RTC_LOG(LS_VERBOSE) << ToString(codec);
}
RTC_LOG(LS_VERBOSE) << "Supported recv codecs in order of preference:";
- recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders());
+ recv_codecs_ = LegacyCollectCodecs(decoder_factory_->GetSupportedDecoders());
for (const Codec& codec : recv_codecs_) {
RTC_LOG(LS_VERBOSE) << ToString(codec);
}
@@ -708,84 +817,6 @@
return audio_state_.get();
}
-std::vector<Codec> WebRtcVoiceEngine::CollectCodecs(
- const std::vector<webrtc::AudioCodecSpec>& specs) const {
- PayloadTypeMapper mapper;
- std::vector<Codec> out;
-
- // Only generate CN payload types for these clockrates:
- std::map<int, bool, std::greater<int>> generate_cn = {
- {8000, false}, {16000, false}, {32000, false}};
- // Only generate telephone-event payload types for these clockrates:
- std::map<int, bool, std::greater<int>> generate_dtmf = {
- {8000, false}, {16000, false}, {32000, false}, {48000, false}};
-
- auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
- std::vector<Codec>* out) {
- std::optional<Codec> opt_codec = mapper.ToAudioCodec(format);
- if (opt_codec) {
- if (out) {
- out->push_back(*opt_codec);
- }
- } else {
- RTC_LOG(LS_ERROR) << "Unable to assign payload type to format: "
- << rtc::ToString(format);
- }
-
- return opt_codec;
- };
-
- for (const auto& spec : specs) {
- // We need to do some extra stuff before adding the main codecs to out.
- std::optional<Codec> opt_codec = map_format(spec.format, nullptr);
- if (opt_codec) {
- Codec& codec = *opt_codec;
- if (spec.info.supports_network_adaption) {
- codec.AddFeedbackParam(
- FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
- }
-
- if (spec.info.allow_comfort_noise) {
- // Generate a CN entry if the decoder allows it and we support the
- // clockrate.
- auto cn = generate_cn.find(spec.format.clockrate_hz);
- if (cn != generate_cn.end()) {
- cn->second = true;
- }
- }
-
- // Generate a telephone-event entry if we support the clockrate.
- auto dtmf = generate_dtmf.find(spec.format.clockrate_hz);
- if (dtmf != generate_dtmf.end()) {
- dtmf->second = true;
- }
-
- out.push_back(codec);
-
- if (codec.name == kOpusCodecName) {
- std::string red_fmtp =
- rtc::ToString(codec.id) + "/" + rtc::ToString(codec.id);
- map_format({kRedCodecName, 48000, 2, {{"", red_fmtp}}}, &out);
- }
- }
- }
-
- // Add CN codecs after "proper" audio codecs.
- for (const auto& cn : generate_cn) {
- if (cn.second) {
- map_format({kCnCodecName, cn.first, 1}, &out);
- }
- }
-
- // Add telephone-event codecs last.
- for (const auto& dtmf : generate_dtmf) {
- if (dtmf.second) {
- map_format({kDtmfCodecName, dtmf.first, 1}, &out);
- }
- }
-
- return out;
-}
// --------------------------------- WebRtcVoiceSendChannel ------------------
@@ -2151,11 +2182,20 @@
}
bool WebRtcVoiceReceiveChannel::SetRecvCodecs(
- const std::vector<Codec>& codecs) {
+ const std::vector<Codec>& codecs_in) {
RTC_DCHECK_RUN_ON(worker_thread_);
- // Set the payload types to be used for incoming media.
- RTC_LOG(LS_INFO) << "Setting receive voice codecs.";
+ auto codecs = codecs_in;
+ // Record the payload types used in the payload type suggester.
+ RTC_LOG(LS_INFO) << "Setting receive voice codecs. Mid is " << mid_;
+ for (auto& codec : codecs) {
+ auto error = call_->GetPayloadTypeSuggester()->AddLocalMapping(
+ mid_, codec.id, codec);
+ if (!error.ok()) {
+ RTC_LOG(LS_ERROR) << "Failed to register PT for " << codec.ToString();
+ return false;
+ }
+ }
if (!VerifyUniquePayloadTypes(codecs)) {
RTC_LOG(LS_ERROR) << "Codec payload types overlap.";
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 942c569..dd5b84b 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -34,13 +34,14 @@
#include "api/audio_codecs/audio_format.h"
#include "api/audio_options.h"
#include "api/call/audio_sink.h"
-#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/field_trials_view.h"
#include "api/frame_transformer_interface.h"
+#include "api/media_types.h"
#include "api/rtc_error.h"
+#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
@@ -57,12 +58,11 @@
#include "media/base/media_channel_impl.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
-#include "media/base/rtp_utils.h"
#include "media/base/stream_params.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/system/file_wrapper.h"
@@ -147,9 +147,6 @@
webrtc::AudioProcessing* apm() const;
webrtc::AudioState* audio_state();
- std::vector<Codec> CollectCodecs(
- const std::vector<webrtc::AudioCodecSpec>& specs) const;
-
webrtc::SequenceChecker signal_thread_checker_{
webrtc::SequenceChecker::kDetached};
webrtc::SequenceChecker worker_thread_checker_{
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 5e11f04..f21f407 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -79,6 +79,7 @@
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
+ "../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/units:timestamp",
"../call:rtp_interfaces",
@@ -1105,6 +1106,7 @@
"../api:ice_transport_interface",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
+ "../api:make_ref_counted",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_stats_api",
@@ -1115,16 +1117,20 @@
"../api:sequence_checker",
"../api:turn_customizer",
"../api/adaptation:resource_adaptation_api",
+ "../api/audio:audio_device",
"../api/crypto:options",
"../api/environment",
"../api/rtc_event_log",
"../api/task_queue:pending_task_safety_flag",
+ "../api/transport:bandwidth_estimation_settings",
"../api/transport:bitrate_settings",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
+ "../api/units:time_delta",
"../api/video:video_codec_constants",
"../call:call_interfaces",
"../call:payload_type_picker",
+ "../media:codec",
"../media:media_channel",
"../media:media_engine",
"../media:rid_description",
diff --git a/pc/channel.cc b/pc/channel.cc
index 45c6e74..6e4a80b 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -12,28 +12,44 @@
#include <algorithm>
#include <cstdint>
+#include <functional>
+#include <memory>
+#include <optional>
#include <string>
-#include <type_traits>
#include <utility>
+#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
+#include "api/crypto/crypto_options.h"
+#include "api/jsep.h"
+#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
-#include "api/units/timestamp.h"
+#include "api/task_queue/task_queue_base.h"
#include "media/base/codec.h"
+#include "media/base/media_channel.h"
#include "media/base/rid_description.h"
#include "media/base/rtp_utils.h"
+#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/dtls_transport_internal.h"
#include "pc/rtp_media_utils.h"
+#include "pc/rtp_transport_internal.h"
+#include "pc/session_description.h"
+#include "rtc_base/async_packet_socket.h"
#include "rtc_base/checks.h"
+#include "rtc_base/containers/flat_set.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"
+#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
+#include "rtc_base/socket.h"
#include "rtc_base/strings/string_format.h"
+#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
+#include "rtc_base/unique_id_generator.h"
namespace cricket {
namespace {
@@ -907,7 +923,9 @@
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
- RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
+ // Disabled because suggeting PTs takes thread jumps.
+ // TODO: https://issues.webrtc.org/360058654 - reenable after cleanup
+ // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
criteria_modified,
@@ -916,7 +934,7 @@
: std::nullopt,
error_desc);
- RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
+ // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
return success;
}
diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index 56840a3..4839ea7 100644
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h
@@ -17,7 +17,6 @@
#include <string>
#include <vector>
-#include "api/candidate.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/rtc_error.h"
@@ -243,6 +242,9 @@
const webrtc::PayloadTypeRecorder& local_payload_types() const {
return local_payload_types_;
}
+ webrtc::PayloadTypeRecorder& local_payload_types() {
+ return local_payload_types_;
+ }
void CommitPayloadTypes() {
RTC_DCHECK_RUN_ON(network_thread_);
local_payload_types_.Commit();
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 2780196..cb79860 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -12,25 +12,58 @@
#include <stddef.h>
+#include <cstdint>
#include <functional>
+#include <map>
#include <memory>
+#include <optional>
#include <string>
-#include <type_traits>
#include <utility>
+#include <vector>
#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "api/async_dns_resolver.h"
+#include "api/candidate.h"
#include "api/dtls_transport_interface.h"
#include "api/environment/environment.h"
+#include "api/ice_transport_interface.h"
+#include "api/jsep.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
+#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
+#include "api/transport/data_channel_transport_interface.h"
#include "api/transport/enums.h"
+#include "call/payload_type.h"
+#include "call/payload_type_picker.h"
+#include "media/base/codec.h"
#include "media/sctp/sctp_transport_internal.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/dtls_transport.h"
+#include "p2p/base/dtls_transport_internal.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
+#include "p2p/base/port_allocator.h"
+#include "p2p/base/transport_description.h"
+#include "p2p/base/transport_info.h"
+#include "pc/dtls_srtp_transport.h"
+#include "pc/dtls_transport.h"
+#include "pc/jsep_transport.h"
+#include "pc/rtp_transport.h"
+#include "pc/rtp_transport_internal.h"
+#include "pc/sctp_transport.h"
+#include "pc/session_description.h"
+#include "pc/srtp_transport.h"
+#include "pc/transport_stats.h"
#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"
+#include "rtc_base/rtc_certificate.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
@@ -225,6 +258,15 @@
RTCErrorOr<webrtc::PayloadType> JsepTransportController::SuggestPayloadType(
const std::string& mid,
cricket::Codec codec) {
+ // Because SDP processing runs on the signal thread and Call processing
+ // runs on the worker thread, we allow cross thread invocation until we
+ // can clean up the thread work.
+ if (!network_thread_->IsCurrent()) {
+ return network_thread_->BlockingCall([&] {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ return SuggestPayloadType(mid, codec);
+ });
+ }
RTC_DCHECK_RUN_ON(network_thread_);
const cricket::JsepTransport* transport = GetJsepTransportForMid(mid);
if (transport) {
@@ -245,6 +287,27 @@
return payload_type_picker_.SuggestMapping(codec, nullptr);
}
+RTCError JsepTransportController::AddLocalMapping(const std::string& mid,
+ PayloadType payload_type,
+ const cricket::Codec& codec) {
+ // Because SDP processing runs on the signal thread and Call processing
+ // runs on the worker thread, we allow cross thread invocation until we
+ // can clean up the thread work.
+ if (!network_thread_->IsCurrent()) {
+ return network_thread_->BlockingCall([&] {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ return AddLocalMapping(mid, payload_type, codec);
+ });
+ }
+ RTC_DCHECK_RUN_ON(network_thread_);
+ cricket::JsepTransport* transport = GetJsepTransportForMid(mid);
+ if (!transport) {
+ return RTCError(RTCErrorType::INVALID_PARAMETER,
+ "AddLocalMapping: no transport for mid");
+ }
+ return transport->local_payload_types().AddMapping(payload_type, codec);
+}
+
bool JsepTransportController::SetLocalCertificate(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
if (!network_thread_->IsCurrent()) {
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index 2846dab..b5e3ebf 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -243,6 +243,9 @@
// or a newly suggested one.
RTCErrorOr<PayloadType> SuggestPayloadType(const std::string& mid,
cricket::Codec codec) override;
+ RTCError AddLocalMapping(const std::string& mid,
+ PayloadType payload_type,
+ const cricket::Codec& codec) override;
// TODO(deadbeef): GetStats isn't const because all the way down to
// OpenSSLStreamAdapter, GetSslCipherSuite and GetDtlsSrtpCryptoSuite are not
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 32f21cf..b125f9e 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -13,54 +13,110 @@
#include <limits.h>
#include <stddef.h>
-#include <algorithm>
+#include <cstdint>
+#include <functional>
+#include <map>
#include <memory>
#include <optional>
#include <set>
#include <string>
#include <utility>
+#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
+#include "api/adaptation/resource.h"
+#include "api/audio/audio_device.h"
+#include "api/candidate.h"
+#include "api/crypto/crypto_options.h"
+#include "api/data_channel_interface.h"
+#include "api/dtls_transport_interface.h"
#include "api/environment/environment.h"
+#include "api/jsep.h"
#include "api/jsep_ice_candidate.h"
+#include "api/make_ref_counted.h"
+#include "api/media_stream_interface.h"
#include "api/media_types.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_error.h"
+#include "api/rtc_event_log_output.h"
#include "api/rtp_parameters.h"
+#include "api/rtp_receiver_interface.h"
+#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_direction.h"
+#include "api/rtp_transceiver_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sctp_transport_interface.h"
+#include "api/sequence_checker.h"
+#include "api/set_local_description_observer_interface.h"
+#include "api/set_remote_description_observer_interface.h"
+#include "api/stats/rtc_stats_collector_callback.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/transport/bandwidth_estimation_settings.h"
+#include "api/transport/bitrate_settings.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "api/transport/enums.h"
+#include "api/turn_customizer.h"
#include "api/uma_metrics.h"
+#include "api/units/time_delta.h"
#include "api/video/video_codec_constants.h"
#include "call/audio_state.h"
#include "call/packet_receiver.h"
-#include "media/base/media_channel.h"
+#include "media/base/codec.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/basic_async_resolver_factory.h"
-#include "p2p/base/connection.h"
#include "p2p/base/connection_info.h"
#include "p2p/base/dtls_transport_internal.h"
+#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/p2p_transport_channel.h"
+#include "p2p/base/port.h"
+#include "p2p/base/port_allocator.h"
+#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
+#include "pc/channel_interface.h"
+#include "pc/connection_context.h"
+#include "pc/data_channel_utils.h"
+#include "pc/dtls_transport.h"
#include "pc/ice_server_parsing.h"
+#include "pc/jsep_transport_controller.h"
+#include "pc/legacy_stats_collector.h"
+#include "pc/rtc_stats_collector.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_receiver_proxy.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_sender_proxy.h"
+#include "pc/rtp_transceiver.h"
+#include "pc/rtp_transmission_manager.h"
+#include "pc/rtp_transport_internal.h"
+#include "pc/sctp_data_channel.h"
#include "pc/sctp_transport.h"
+#include "pc/sdp_offer_answer.h"
+#include "pc/session_description.h"
#include "pc/simulcast_description.h"
-#include "pc/webrtc_session_description_factory.h"
+#include "pc/transceiver_list.h"
+#include "pc/transport_stats.h"
+#include "pc/usage_pattern.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/crypto_random.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/logging.h"
#include "rtc_base/net_helper.h"
#include "rtc_base/network.h"
#include "rtc_base/network_constants.h"
+#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket_address.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
+#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/unique_id_generator.h"
#include "system_wrappers/include/metrics.h"
@@ -695,6 +751,12 @@
kPeerConnectionAddressFamilyCounter_Max);
return InitializeTransportController_n(configuration, dependencies);
});
+ if (call_ptr_) {
+ worker_thread()->BlockingCall([this, tc = transport_controller_copy_] {
+ RTC_DCHECK_RUN_ON(worker_thread());
+ call_->SetPayloadTypeSuggester(tc);
+ });
+ }
configuration_ = configuration;
