Rename Call::Config to CallConfig, keep old name as alias.
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.
Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.
Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.
Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}diff --git a/call/call.h b/call/call.h
index eb23e8b..884a21b 100644
--- a/call/call.h
+++ b/call/call.h
@@ -72,49 +72,51 @@
virtual ~PacketReceiver() {}
};
+struct CallConfig {
+ explicit CallConfig(RtcEventLog* event_log) : event_log(event_log) {
+ RTC_DCHECK(event_log);
+ }
+
+ static constexpr int kDefaultStartBitrateBps = 300000;
+
+ // Bitrate config used until valid bitrate estimates are calculated. Also
+ // used to cap total bitrate used. This comes from the remote connection.
+ struct BitrateConfig {
+ int min_bitrate_bps = 0;
+ int start_bitrate_bps = kDefaultStartBitrateBps;
+ int max_bitrate_bps = -1;
+ } bitrate_config;
+
+ // The local client's bitrate preferences. The actual configuration used
+ // is a combination of this and |bitrate_config|. The combination is
+ // currently more complicated than a simple mask operation (see
+ // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
+ // start <= max holds for set parameters.
+ struct BitrateConfigMask {
+ rtc::Optional<int> min_bitrate_bps;
+ rtc::Optional<int> start_bitrate_bps;
+ rtc::Optional<int> max_bitrate_bps;
+ };
+
+ // AudioState which is possibly shared between multiple calls.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ rtc::scoped_refptr<AudioState> audio_state;
+
+ // Audio Processing Module to be used in this call.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ AudioProcessing* audio_processing = nullptr;
+
+ // RtcEventLog to use for this call. Required.
+ // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
+ RtcEventLog* event_log = nullptr;
+};
+
// A Call instance can contain several send and/or receive streams. All streams
// are assumed to have the same remote endpoint and will share bitrate estimates
// etc.
class Call {
public:
- struct Config {
- explicit Config(RtcEventLog* event_log) : event_log(event_log) {
- RTC_DCHECK(event_log);
- }
-
- static constexpr int kDefaultStartBitrateBps = 300000;
-
- // Bitrate config used until valid bitrate estimates are calculated. Also
- // used to cap total bitrate used. This comes from the remote connection.
- struct BitrateConfig {
- int min_bitrate_bps = 0;
- int start_bitrate_bps = kDefaultStartBitrateBps;
- int max_bitrate_bps = -1;
- } bitrate_config;
-
- // The local client's bitrate preferences. The actual configuration used
- // is a combination of this and |bitrate_config|. The combination is
- // currently more complicated than a simple mask operation (see
- // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
- // start <= max holds for set parameters.
- struct BitrateConfigMask {
- rtc::Optional<int> min_bitrate_bps;
- rtc::Optional<int> start_bitrate_bps;
- rtc::Optional<int> max_bitrate_bps;
- };
-
- // AudioState which is possibly shared between multiple calls.
- // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
- rtc::scoped_refptr<AudioState> audio_state;
-
- // Audio Processing Module to be used in this call.
- // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
- AudioProcessing* audio_processing = nullptr;
-
- // RtcEventLog to use for this call. Required.
- // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
- RtcEventLog* event_log = nullptr;
- };
+ using Config = CallConfig;
struct Stats {
std::string ToString(int64_t time_ms) const;