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Change SSRC-passing for MediaChannel from external to callback

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This makes the handling somewhat more uniform, and is the same
for both video and audio channels.

Bug: webrtc:13931
Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40107}
This commit is contained in:
Harald Alvestrand 2023-05-22 12:49:05 +00:00 committed by WebRTC LUCI CQ
parent 5dc42050e1
commit 13897e67c8
13 changed files with 285 additions and 55 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

32
media/BUILD.gn
View File

@ -156,6 +156,7 @@ rtc_library("media_channel_impl") {
"../api:frame_transformer_interface",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:scoped_refptr",
@ -429,43 +430,66 @@ rtc_library("rtc_audio_video") {
":media_channel",
":media_channel_impl",
":media_constants",
":rid_description",
":rtc_media_base",
":rtc_media_config",
":rtp_utils",
":stream_params",
"../api:array_view",
"../api:audio_options_api",
"../api:call_api",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:priority",
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:transport_api",
"../api/audio:audio_frame_api",
"../api/audio:audio_frame_processor",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/transport:bitrate_settings",
"../api/transport:field_trial_based_config",
"../api/transport/rtp:rtp_source",
"../api/units:data_rate",
"../api/units:time_delta",
"../api/units:timestamp",
"../api/video:recordable_encoded_frame",
"../api/video:resolution",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator_factory",
"../api/video:video_codec_constants",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video:video_stream_encoder",
"../api/video_codecs:rtc_software_fallback_wrappers",
"../api/video_codecs:scalability_mode",
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
"../call:receive_stream_interface",
"../call:rtp_interfaces",
"../call:video_stream_api",
"../common_video",
"../common_video:frame_counts",
"../modules/async_audio_processing:async_audio_processing",
"../modules/audio_device",
"../modules/audio_device:audio_device_impl",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:api",
"../modules/audio_processing:audio_processing_statistics",
"../modules/audio_processing/aec_dump",
"../modules/audio_processing/agc:gain_control_interface",
"../modules/rtp_rtcp:rtp_rtcp_format",
@ -479,6 +503,7 @@ rtc_library("rtc_audio_video") {
"../rtc_base:byte_order",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:dscp",
"../rtc_base:event_tracer",
"../rtc_base:ignore_wundef",
"../rtc_base:logging",
@ -487,6 +512,7 @@ rtc_library("rtc_audio_video") {
"../rtc_base:race_checker",
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_conversions",
"../rtc_base:socket",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base:threading",
@ -495,15 +521,21 @@ rtc_library("rtc_audio_video") {
"../rtc_base/experiments:min_video_bitrate_experiment",
"../rtc_base/experiments:normalize_simulcast_size_experiment",
"../rtc_base/experiments:rate_control_settings",
"../rtc_base/network:sent_packet",
"../rtc_base/synchronization:mutex",
"../rtc_base/system:file_wrapper",
"../rtc_base/system:no_unique_address",
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/base64",
"../system_wrappers",
"../system_wrappers:metrics",
"../video/config:encoder_config",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",

7
media/base/fake_media_engine.h
View File

@ -126,6 +126,10 @@ class RtpHelper : public Base {
virtual absl::optional<uint32_t> GetUnsignaledSsrc() const {
return absl::nullopt;
}
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {}
virtual bool SetLocalSsrc(const StreamParams& sp) { return true; }
virtual void OnDemuxerCriteriaUpdatePending() {}
virtual void OnDemuxerCriteriaUpdateComplete() {}
@ -519,6 +523,9 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
}
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {}
bool SendCodecHasLntf() const override { return false; }
bool SendCodecHasNack() const override { return false; }
absl::optional<int> SendCodecRtxTime() const override {

7
media/base/media_channel.h
View File

@ -13,6 +13,7 @@
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
@ -267,6 +268,9 @@ class MediaSendChannelInterface
webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
}
virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
// Called whenever the list of sending SSRCs changes.
virtual void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) = 0;
};
class MediaReceiveChannelInterface
@ -292,8 +296,7 @@ class MediaReceiveChannelInterface
// Gets the current unsignaled receive stream's SSRC, if there is one.
virtual absl::optional<uint32_t> GetUnsignaledSsrc() const = 0;
// Sets the local SSRC for listening to incoming RTCP reports.
virtual bool SetLocalSsrc(const StreamParams& sp) = 0;
virtual void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) = 0;
// This is currently a workaround because of the demuxer state being managed
// across two separate threads. Once the state is consistently managed on
// the same thread (network), this workaround can be removed.

28
media/base/media_channel_impl.h
View File

@ -16,10 +16,12 @@
#include <functional>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/audio_options.h"
@ -30,6 +32,7 @@
#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"
@ -235,6 +238,10 @@ class VideoMediaChannel : public MediaChannel,
}
// Declared here in order to avoid "found by multiple paths" compile error
bool AddSendStream(const StreamParams& sp) override = 0;
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override = 0;
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override =
0;
// This fills the "bitrate parts" (rtx, video bitrate) of the
// BandwidthEstimationInfo, since that part that isn't possible to get
@ -305,6 +312,11 @@ class VoiceMediaChannel : public MediaChannel,
return nullptr;
}
// Declared here to avoid "found in multiple base-class subobjects" error
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override = 0;
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override =
0;
void SetExtmapAllowMixed(bool mixed) override {
MediaChannel::SetExtmapAllowMixed(mixed);
}
@ -410,6 +422,10 @@ class VoiceMediaSendChannel : public VoiceMediaSendChannelInterface {
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override {
return impl()->GetRtpSendParameters(ssrc);
}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
impl()->SetSsrcListChangedCallback(std::move(callback));
}
// Implementation of VoiceMediaSendChannel
bool SetSendParameters(const AudioSendParameters& params) override {
return impl()->SetSendParameters(params);
@ -493,8 +509,8 @@ class VoiceMediaReceiveChannel : public VoiceMediaReceiveChannelInterface {
void ResetUnsignaledRecvStream() override {
return impl()->ResetUnsignaledRecvStream();
}
bool SetLocalSsrc(const StreamParams& sp) override {
return impl()->SetLocalSsrc(sp);
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {
return impl()->ChooseReceiverReportSsrc(choices);
}
absl::optional<uint32_t> GetUnsignaledSsrc() const override {
return impl()->GetUnsignaledSsrc();
@ -665,6 +681,10 @@ class VideoMediaSendChannel : public VideoMediaSendChannelInterface {
absl::optional<int> SendCodecRtxTime() const override {
return impl()->SendCodecRtxTime();
}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
impl()->SetSsrcListChangedCallback(std::move(callback));
}
MediaChannel* ImplForTesting() override { return impl_; }
@ -730,8 +750,8 @@ class VideoMediaReceiveChannel : public VideoMediaReceiveChannelInterface {
absl::optional<uint32_t> GetUnsignaledSsrc() const override {
return impl()->GetUnsignaledSsrc();
}
bool SetLocalSsrc(const StreamParams& sp) override {
return impl()->SetLocalSsrc(sp);
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {
return impl()->ChooseReceiverReportSsrc(choices);
}
void OnDemuxerCriteriaUpdatePending() override {
impl()->OnDemuxerCriteriaUpdatePending();

71
media/engine/webrtc_video_engine.cc
View File

@ -14,33 +14,52 @@
#include <algorithm>
#include <cstdint>
#include <initializer_list>
#include <set>
#include <string>
#include <type_traits>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/container/inlined_vector.h"
#include "absl/functional/bind_front.h"
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/video/video_codec_constants.h"
#include "api/media_types.h"
#include "api/priority.h"
#include "api/rtp_transceiver_direction.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/resolution.h"
#include "api/video/video_codec_type.h"
#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/call.h"
#include "call/packet_receiver.h"
#include "call/receive_stream.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "common_video/frame_counts.h"
#include "common_video/include/quality_limitation_reason.h"
#include "media/base/media_constants.h"
#include "media/base/rid_description.h"
#include "media/base/rtp_utils.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/engine/webrtc_voice_engine.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "modules/video_coding/codecs/vp9/svc_config.h"
#include "modules/video_coding/svc/scalability_mode_util.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/dscp.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/field_trial_units.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/socket.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
@ -1284,6 +1303,22 @@ void WebRtcVideoChannel::SetReceiverReportSsrc(uint32_t ssrc) {
receive_stream->SetLocalSsrc(ssrc);
}
void WebRtcVideoChannel::ChooseReceiverReportSsrc(
const std::set<uint32_t>& choices) {
RTC_DCHECK_RUN_ON(&thread_checker_);
// If we can continue using the current receiver report, do so.
if (choices.find(rtcp_receiver_report_ssrc_) != choices.end()) {
return;
}
// Go back to the default if list has been emptied.
if (choices.empty()) {
SetReceiverReportSsrc(kDefaultRtcpReceiverReportSsrc);
return;
}
// Any number is as good as any other.
SetReceiverReportSsrc(*choices.begin());
}
bool WebRtcVideoChannel::GetSendCodec(VideoCodec* codec) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec()) {
@ -1417,8 +1452,10 @@ bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) {
// If legacy kBoth mode, tell my receiver part about its SSRC.
// In kSend mode, this is the responsibility of the caller.
if (role() == MediaChannel::Role::kBoth) {
if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
SetReceiverReportSsrc(ssrc);
ChooseReceiverReportSsrc(send_ssrcs_);
} else {
if (ssrc_list_changed_callback_) {
ssrc_list_changed_callback_(send_ssrcs_);
}
}
@ -1446,9 +1483,12 @@ bool WebRtcVideoChannel::RemoveSendStream(uint32_t ssrc) {
send_streams_.erase(it);
// Switch receiver report SSRCs, the one in use is no longer valid.
if (rtcp_receiver_report_ssrc_ == ssrc) {
SetReceiverReportSsrc(send_streams_.empty() ? kDefaultRtcpReceiverReportSsrc
: send_streams_.begin()->first);
if (role() == MediaChannel::Role::kBoth) {
ChooseReceiverReportSsrc(send_ssrcs_);
} else {
if (ssrc_list_changed_callback_) {
ssrc_list_changed_callback_(send_ssrcs_);
}
}
delete removed_stream;
@ -1617,15 +1657,6 @@ void WebRtcVideoChannel::ResetUnsignaledRecvStream() {
}
}
bool WebRtcVideoChannel::SetLocalSsrc(const StreamParams& sp) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(role() == MediaChannel::Role::kReceive);
if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
SetReceiverReportSsrc(sp.first_ssrc());
}
return true;
}
absl::optional<uint32_t> WebRtcVideoChannel::GetUnsignaledSsrc() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
absl::optional<uint32_t> ssrc;

44
media/engine/webrtc_video_engine.h
View File

@ -11,7 +11,10 @@
#ifndef MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
#define MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
#include <stddef.h>
#include <cstdint>
#include <functional>
#include <map>
#include <memory>
#include <set>
@ -19,26 +22,54 @@
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.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/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/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/recordable_encoded_frame.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video/video_stream_encoder_settings.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/call.h"
#include "call/flexfec_receive_stream.h"
#include "call/rtp_config.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.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_engine.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
#include "video/config/video_encoder_config.h"
namespace webrtc {
class VideoDecoderFactory;
@ -141,7 +172,6 @@ class WebRtcVideoChannel : public VideoMediaChannel,
bool RemoveRecvStream(uint32_t ssrc) override;
void ResetUnsignaledRecvStream() override;
absl::optional<uint32_t> GetUnsignaledSsrc() const override;
bool SetLocalSsrc(const StreamParams& sp) override;
void OnDemuxerCriteriaUpdatePending() override;
void OnDemuxerCriteriaUpdateComplete() override;
bool SetSink(uint32_t ssrc,
@ -191,6 +221,15 @@ class WebRtcVideoChannel : public VideoMediaChannel,
send_codec_changed_callback_ = std::move(callback);
}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
ssrc_list_changed_callback_ = std::move(callback);
}
// Choose one of the available SSRCs (or default if none) as the current
// receiver report SSRC.
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override;
// Implemented for VideoMediaChannelTest.
bool sending() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
@ -733,6 +772,9 @@ class WebRtcVideoChannel : public VideoMediaChannel,
// Callback invoked whenever the send codec changes.
// TODO(bugs.webrtc.org/13931): Remove again when coupling isn't needed.
absl::AnyInvocable<void()> send_codec_changed_callback_;
// Callback invoked whenever the list of SSRCs changes.
absl::AnyInvocable<void(const std::set<uint32_t>&)>
ssrc_list_changed_callback_;
};
} // namespace cricket

55
media/engine/webrtc_voice_engine.cc
View File

@ -12,20 +12,40 @@
#include <algorithm>
#include <atomic>
#include <cstdint>
#include <functional>
#include <initializer_list>
#include <iterator>
#include <memory>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/algorithm/algorithm.h"
#include "absl/algorithm/container.h"
#include "absl/functional/bind_front.h"
#include "absl/strings/match.h"
#include "api/audio/audio_frame.h"
#include "api/audio/audio_frame_processor.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/call/audio_sink.h"
#include "api/field_trials_view.h"
#include "api/make_ref_counted.h"
#include "api/media_types.h"
#include "api/priority.h"
#include "api/rtp_headers.h"
#include "api/rtp_transceiver_direction.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/bitrate_settings.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/packet_receiver.h"
#include "call/rtp_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "media/base/audio_source.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
@ -33,26 +53,25 @@
#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_device/audio_device_impl.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/include/audio_processing_statistics.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"
#include "rtc_base/arraysize.h"
#include "rtc_base/byte_order.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/field_trial_units.h"
#include "rtc_base/checks.h"
#include "rtc_base/dscp.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
#include "rtc_base/helpers.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/logging.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/string_encode.h"
#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/third_party/base64/base64.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
@ -1992,14 +2011,26 @@ absl::optional<uint32_t> WebRtcVoiceMediaChannel::GetUnsignaledSsrc() const {
return unsignaled_recv_ssrcs_.back();
}
bool WebRtcVoiceMediaChannel::SetLocalSsrc(const StreamParams& sp) {
RTC_DCHECK(role() == MediaChannel::Role::kReceive);
uint32_t ssrc = sp.first_ssrc();
void WebRtcVoiceMediaChannel::SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) {
ssrc_list_changed_callback_ = std::move(callback);
}
void WebRtcVoiceMediaChannel::ChooseReceiverReportSsrc(
const std::set<uint32_t>& choices) {
// Don't change SSRC if set is empty. Note that this differs from
// the behavior of video.
if (choices.empty()) {
return;
}
if (choices.find(receiver_reports_ssrc_) != choices.end()) {
return;
}
uint32_t ssrc = *(choices.begin());
receiver_reports_ssrc_ = ssrc;
for (auto& kv : recv_streams_) {
call_->OnLocalSsrcUpdated(kv.second->stream(), ssrc);
}
return true;
}
// Not implemented.

43
media/engine/webrtc_voice_engine.h
View File

@ -11,27 +11,60 @@
#ifndef MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
#define MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
#include <stddef.h>
#include <stdint.h>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/audio/audio_frame_processor.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder_factory.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/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/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/rtp/rtp_source.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/call.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_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/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/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/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/system/file_wrapper.h"
#include "rtc_base/task_queue.h"
namespace webrtc {
@ -183,7 +216,9 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
void ResetUnsignaledRecvStream() override;
absl::optional<uint32_t> GetUnsignaledSsrc() const override;
bool SetLocalSsrc(const StreamParams& sp) override;
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override;
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override;
void OnDemuxerCriteriaUpdatePending() override;
void OnDemuxerCriteriaUpdateComplete() override;
@ -334,11 +369,13 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
class WebRtcAudioSendStream;
std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
std::vector<webrtc::RtpExtension> send_rtp_extensions_;
std::string mid_;
class WebRtcAudioReceiveStream;
std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_;
@ -360,6 +397,10 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
void FillSendCodecStats(VoiceMediaSendInfo* voice_media_info);
void FillReceiveCodecStats(VoiceMediaReceiveInfo* voice_media_info);
// Callback invoked whenever the list of SSRCs changes.
absl::AnyInvocable<void(const std::set<uint32_t>&)>
ssrc_list_changed_callback_;
};
} // namespace cricket

5
pc/BUILD.gn
View File

@ -1593,6 +1593,7 @@ rtc_library("rtp_transceiver") {
":session_description",
"../api:array_view",
"../api:audio_options_api",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
@ -1600,13 +1601,17 @@ rtc_library("rtp_transceiver") {
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:options",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/video:video_bitrate_allocator_factory",
"../media:codec",
"../media:media_channel",
"../media:media_channel_impl",
"../media:media_constants",
"../media:rtc_media_base",
"../media:rtc_media_config",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",

12
pc/channel.cc
View File

@ -712,18 +712,6 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
if (media_send_channel()->AddSendStream(new_stream)) {
RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0]
<< " into " << ToString();
// Must also tell the corresponding receive stream to listen for
// RRs coming in on the new stream's SSRC
if (media_send_channel_impl_) {
if (all_streams.size() == 1) {
if (!media_receive_channel()->SetLocalSsrc(new_stream)) {
error_desc = StringFormat(
"Failed to set local ssrc: %u into m-section with mid='%s'",
new_stream.first_ssrc(), mid().c_str());
ret = false;
}
}
}
} else {
error_desc = StringFormat(
"Failed to add send stream ssrc: %u into m-section with mid='%s'",

22
pc/rtp_transceiver.cc
View File

@ -10,18 +10,22 @@
#include "pc/rtp_transceiver.h"
#include <algorithm>
#include <stdint.h>
#include <iterator>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "api/peer_connection_interface.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "media/base/codec.h"
#include "media/base/media_channel_impl.h"
#include "media/base/media_constants.h"
#include "media/base/media_engine.h"
#include "pc/channel.h"
@ -230,6 +234,13 @@ RTCError RtpTransceiver::CreateChannel(
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel = media_receive_channel.get()](
const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VoiceChannel>(
context()->worker_thread(), context()->network_thread(),
@ -278,6 +289,13 @@ RTCError RtpTransceiver::CreateChannel(
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel = media_receive_channel.get()](
const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VideoChannel>(
context()->worker_thread(), context()->network_thread(),

4
pc/rtp_transceiver.h
View File

@ -18,9 +18,11 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
@ -34,6 +36,8 @@
#include "api/task_queue/task_queue_base.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "media/base/media_channel.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "pc/channel_interface.h"
#include "pc/connection_context.h"
#include "pc/proxy.h"

10
pc/test/mock_voice_media_channel.h
View File

@ -11,6 +11,7 @@
#define PC_TEST_MOCK_VOICE_MEDIA_CHANNEL_H_
#include <memory>
#include <set>
#include <string>
#include <vector>
@ -62,7 +63,14 @@ class MockVoiceMediaChannel : public VoiceMediaChannel {
GetUnsignaledSsrc,
(),
(const, override));
MOCK_METHOD(bool, SetLocalSsrc, (const StreamParams& sp), (override));
MOCK_METHOD(void,
ChooseReceiverReportSsrc,
(const std::set<uint32_t>&),
(override));
MOCK_METHOD(void,
SetSsrcListChangedCallback,
(absl::AnyInvocable<void(const std::set<uint32_t>&)>),
(override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override));
MOCK_METHOD(int, GetRtpSendTimeExtnId, (), (const, override));