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Move GetSendCodec() to MediaSendChannelInterface

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This allows the voice send channels to share the method definition.

Bug: webrtc:15214
Change-Id: Ie0cc23f3694eeb8322a9ea7328a8d56fa7571c95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40322}
This commit is contained in:
Florent Castelli 2023-06-20 17:12:57 +00:00 committed by WebRTC LUCI CQ
parent d1780836f4
commit ee97e6ad88
13 changed files with 56 additions and 19 deletions
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1
media/BUILD.gn
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@ -293,6 +293,7 @@ rtc_library("codec") {
":media_constants",
"../api:field_trials_view",
"../api:rtp_parameters",
"../api/audio_codecs:audio_codecs_api",
"../api/video_codecs:video_codecs_api",
"../rtc_base:checks",
"../rtc_base:logging",

10
media/base/codec.cc
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@ -12,6 +12,7 @@
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_format.h"
#include "api/video_codecs/av1_profile.h"
#include "api/video_codecs/h264_profile_level_id.h"
#include "api/video_codecs/vp9_profile.h"
@ -131,6 +132,11 @@ Codec::Codec(Type type,
Codec::Codec(Type type) : Codec(type, 0, "", 0) {}
Codec::Codec(const webrtc::SdpAudioFormat& c)
: Codec(Type::kAudio, 0, c.name, c.clockrate_hz, c.num_channels) {
params = c.parameters;
}
Codec::Codec(const webrtc::SdpVideoFormat& c)
: Codec(Type::kVideo, 0, c.name, kVideoCodecClockrate) {
params = c.parameters;
@ -442,6 +448,10 @@ Codec CreateAudioCodec(int id,
return Codec(Codec::Type::kAudio, id, name, clockrate, channels);
}
Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c) {
return Codec(c);
}
Codec CreateVideoCodec(const std::string& name) {
return CreateVideoCodec(0, name);
}

4
media/base/codec.h
View File

@ -19,6 +19,7 @@
#include "absl/container/inlined_vector.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
#include "api/video_codecs/sdp_video_format.h"
@ -166,12 +167,14 @@ struct RTC_EXPORT Codec {
int clockrate,
size_t channels);
explicit Codec(const webrtc::SdpAudioFormat& c);
explicit Codec(const webrtc::SdpVideoFormat& c);
friend Codec CreateAudioCodec(int id,
const std::string& name,
int clockrate,
size_t channels);
friend Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c);
friend Codec CreateAudioRtxCodec(int rtx_payload_type,
int associated_payload_type);
friend Codec CreateVideoCodec(int id, const std::string& name);
@ -188,6 +191,7 @@ Codec CreateAudioCodec(int id,
const std::string& name,
int clockrate,
size_t channels);
Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c);
Codec CreateAudioRtxCodec(int rtx_payload_type, int associated_payload_type);
Codec CreateVideoCodec(const std::string& name);
Codec CreateVideoCodec(int id, const std::string& name);

8
media/base/fake_media_engine.cc
View File

@ -74,6 +74,12 @@ const std::vector<AudioCodec>& FakeVoiceMediaChannel::send_codecs() const {
const std::vector<AudioCodec>& FakeVoiceMediaChannel::codecs() const {
return send_codecs();
}
absl::optional<Codec> FakeVoiceMediaChannel::GetSendCodec() const {
if (!send_codecs_.empty()) {
return send_codecs_.front();
}
return absl::nullopt;
}
const std::vector<FakeVoiceMediaChannel::DtmfInfo>&
FakeVoiceMediaChannel::dtmf_info_queue() const {
return dtmf_info_queue_;
@ -317,7 +323,7 @@ bool FakeVideoMediaChannel::AddSendStream(const StreamParams& sp) {
bool FakeVideoMediaChannel::RemoveSendStream(uint32_t ssrc) {
return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
}
absl::optional<VideoCodec> FakeVideoMediaChannel::GetSendCodec() {
absl::optional<Codec> FakeVideoMediaChannel::GetSendCodec() const {
if (send_codecs_.empty()) {
return absl::nullopt;
}

3
media/base/fake_media_engine.h
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@ -423,6 +423,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
bool SendCodecHasNack() const override { return false; }
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {}
absl::optional<Codec> GetSendCodec() const override;
private:
class VoiceChannelAudioSink : public AudioSource::Sink {
@ -489,7 +490,7 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
bool AddSendStream(const StreamParams& sp) override;
bool RemoveSendStream(uint32_t ssrc) override;
absl::optional<VideoCodec> GetSendCodec() override;
absl::optional<Codec> GetSendCodec() const override;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
void SetDefaultSink(

5
media/base/media_channel.h
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@ -193,6 +193,9 @@ class MediaSendChannelInterface {
virtual VoiceMediaSendChannelInterface* AsVoiceSendChannel() = 0;
virtual cricket::MediaType media_type() const = 0;
// Gets the currently set codecs/payload types to be used for outgoing media.
virtual absl::optional<Codec> GetSendCodec() const = 0;
// Creates a new outgoing media stream with SSRCs and CNAME as described
// by sp.
virtual bool AddSendStream(const StreamParams& sp) = 0;
@ -931,8 +934,6 @@ struct VideoRecvParameters : RtpParameters<VideoCodec> {};
class VideoMediaSendChannelInterface : public MediaSendChannelInterface {
public:
virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
// Gets the currently set codecs/payload types to be used for outgoing media.
virtual absl::optional<VideoCodec> GetSendCodec() = 0;
// Starts or stops transmission (and potentially capture) of local video.
virtual bool SetSend(bool send) = 0;
// Configure stream for sending and register a source.

5
media/base/media_channel_shim.h
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@ -87,6 +87,9 @@ class VoiceMediaShimChannel : public VoiceMediaChannel {
cricket::MediaType media_type() const override { return MEDIA_TYPE_AUDIO; }
// Implementation of MediaSendChannelInterface
absl::optional<Codec> GetSendCodec() const override {
return send_impl()->GetSendCodec();
}
void OnPacketSent(const rtc::SentPacket& sent_packet) override {
send_impl()->OnPacketSent(sent_packet);
}
@ -385,7 +388,7 @@ class VideoMediaShimChannel : public VideoMediaChannel {
bool SetSendParameters(const VideoSendParameters& params) override {
return send_impl()->SetSendParameters(params);
}
absl::optional<VideoCodec> GetSendCodec() override {
absl::optional<Codec> GetSendCodec() const override {
return send_impl()->GetSendCodec();
}
bool SetSend(bool send) override { return send_impl()->SetSend(send); }

2
media/engine/webrtc_video_engine.cc
View File

@ -1300,7 +1300,7 @@ webrtc::RTCError WebRtcVideoSendChannel::SetRtpSendParameters(
return it->second->SetRtpParameters(parameters, std::move(callback));
}
absl::optional<VideoCodec> WebRtcVideoSendChannel::GetSendCodec() {
absl::optional<Codec> WebRtcVideoSendChannel::GetSendCodec() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec()) {
RTC_LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";

2
media/engine/webrtc_video_engine.h
View File

@ -207,7 +207,7 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback) override;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
absl::optional<VideoCodec> GetSendCodec() override;
absl::optional<Codec> GetSendCodec() const override;
bool SetSend(bool send) override;
bool SetVideoSend(
uint32_t ssrc,

24
media/engine/webrtc_video_engine_unittest.cc
View File

@ -2539,7 +2539,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) {
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
absl::optional<VideoCodec> codec = channel_->GetSendCodec();
absl::optional<VideoCodec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP9", codec->name);
@ -2547,14 +2547,14 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) {
// the current thread), hence the ProcessMessages call.
SendImpl()->RequestEncoderFallback();
time_controller_.AdvanceTime(kFrameDuration);
codec = channel_->GetSendCodec();
codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
// No other codec to fall back to, keep using VP8.
SendImpl()->RequestEncoderFallback();
time_controller_.AdvanceTime(kFrameDuration);
codec = channel_->GetSendCodec();
codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
}
@ -2565,7 +2565,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchDefaultFallback) {
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
absl::optional<VideoCodec> codec = channel_->GetSendCodec();
absl::optional<VideoCodec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP9", codec->name);
@ -2577,7 +2577,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchDefaultFallback) {
// Requested encoder is not available. Default fallback is allowed. Switch to
// the next negotiated codec, VP8.
codec = channel_->GetSendCodec();
codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
}
@ -2591,7 +2591,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchStrictPreference) {
parameters.codecs.push_back(vp9);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
absl::optional<VideoCodec> codec = channel_->GetSendCodec();
absl::optional<VideoCodec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
@ -2602,7 +2602,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchStrictPreference) {
// VP9 profile_id=1 is not available. Default fallback is not allowed. Switch
// is not performed.
codec = channel_->GetSendCodec();
codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
@ -2612,7 +2612,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchStrictPreference) {
time_controller_.AdvanceTime(kFrameDuration);
// VP9 profile_id=0 is available. Switch encoder.
codec = channel_->GetSendCodec();
codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP9", codec->name);
}
@ -4281,7 +4281,7 @@ TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) {
AssignDefaultAptRtxTypes();
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
absl::optional<VideoCodec> codec = channel_->GetSendCodec();
absl::optional<VideoCodec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_TRUE(codec->Matches(engine_.send_codecs()[0], &field_trials_));
@ -5036,7 +5036,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithMaxQuantization) {
EXPECT_EQ(atoi(kMaxQuantization),
AddSendStream()->GetVideoStreams().back().max_qp);
absl::optional<VideoCodec> codec = channel_->GetSendCodec();
absl::optional<VideoCodec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ(kMaxQuantization, codec->params[kCodecParamMaxQuantization]);
}
@ -9751,7 +9751,7 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
EXPECT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(true);
absl::optional<VideoCodec> codec = channel_->GetSendCodec();
absl::optional<VideoCodec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
@ -9762,7 +9762,7 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
channel_->SetEncoderSelector(kSsrc, &encoder_selector);
time_controller_.AdvanceTime(kFrameDuration);
codec = channel_->GetSendCodec();
codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP9", codec->name);

8
media/engine/webrtc_voice_engine.cc
View File

@ -49,6 +49,7 @@
#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_constants.h"
#include "media/base/stream_params.h"
#include "media/engine/adm_helpers.h"
@ -1330,6 +1331,13 @@ bool WebRtcVoiceSendChannel::SetSendParameters(
return SetOptions(params.options);
}
absl::optional<Codec> WebRtcVoiceSendChannel::GetSendCodec() const {
if (send_codec_spec_) {
return CreateAudioCodec(send_codec_spec_->format);
}
return absl::nullopt;
}
// Utility function called from SetSendParameters() to extract current send
// codec settings from the given list of codecs (originally from SDP). Both send
// and receive streams may be reconfigured based on the new settings.

2
media/engine/webrtc_voice_engine.h
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@ -218,6 +218,8 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
}
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
absl::optional<Codec> GetSendCodec() const override;
// Functions imported from MediaChannelUtil
void SetInterface(MediaChannelNetworkInterface* iface) override {
MediaChannelUtil::SetInterface(iface);

1
pc/test/mock_voice_media_channel.h
View File

@ -67,6 +67,7 @@ class MockVoiceMediaChannel : public VoiceMediaChannel {
ChooseReceiverReportSsrc,
(const std::set<uint32_t>&),
(override));
MOCK_METHOD(absl::optional<Codec>, GetSendCodec, (), (const, override));
MOCK_METHOD(bool, SendCodecHasNack, (), (const, override));
MOCK_METHOD(void,
SetSsrcListChangedCallback,