Commit 4b266222 authored by Byron Campen's avatar Byron Campen
Browse files

Bug 1822194 - Vendor libwebrtc from 514dff834b 

Upstream commit: https://webrtc.googlesource.com/src/+/514dff834b9ed89aac7a8fcdcca752a468c75c94
    Fix crash when setting RTCRtpEncodingParameter.maxBitrate to a non-positive value.

    This will result in using a default value instead.

    The crash is caused by configuring audio send stream with a non-positive max_bitrate_bps and it expects a positive value (or special value -1).

    Also remove support for setting min bitrate since it is not in the spec and there are no tests.

    Bug: chromium:1377286
    Change-Id: I19af29bff79eda5c0fbb1e528fced7976f5c2511
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284640


    Reviewed-by: default avatarHarald Alvestrand <hta@webrtc.org>
    Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#38719}
parent 6b01cb09

third_party/libwebrtc/README.moz-ff-commit

+3 −0
Original line number Diff line number Diff line
@@ -19758,3 +19758,6 @@ bbc8fc165c 
# MOZ_LIBWEBRTC_SRC=/home/bcampen/checkouts/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh

# base of lastest vendoring

4bee365c82

# MOZ_LIBWEBRTC_SRC=/home/bcampen/checkouts/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh

# base of lastest vendoring

514dff834b

third_party/libwebrtc/README.mozilla

+2 −0
Original line number Diff line number Diff line
@@ -13194,3 +13194,5 @@ libwebrtc updated from /home/bcampen/checkouts/moz-libwebrtc commit mozpatches o 
libwebrtc updated from /home/bcampen/checkouts/moz-libwebrtc commit mozpatches on 2023-03-31T23:53:24.052266.

# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/bcampen/checkouts/moz-libwebrtc --commit mozpatches libwebrtc

libwebrtc updated from /home/bcampen/checkouts/moz-libwebrtc commit mozpatches on 2023-03-31T23:54:45.893780.

# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/bcampen/checkouts/moz-libwebrtc --commit mozpatches libwebrtc

libwebrtc updated from /home/bcampen/checkouts/moz-libwebrtc commit mozpatches on 2023-03-31T23:56:01.420669.

third_party/libwebrtc/media/BUILD.gn

+1 −0
Original line number Diff line number Diff line
@@ -653,6 +653,7 @@ if (rtc_include_tests) { 
        "../call:call_interfaces",

        "../common_video",

        "../modules/audio_device:mock_audio_device",

        "../modules/audio_mixer:audio_mixer_impl",

        "../modules/audio_processing",

        "../modules/audio_processing:api",

        "../modules/audio_processing:mocks",

third_party/libwebrtc/media/engine/webrtc_voice_engine.cc

+0 −8
Original line number Diff line number Diff line
@@ -1041,7 +1041,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream 
    // - a reasonable default of 32kbps min/max

    // - fixed target bitrate from codec spec

    // - lower min bitrate if adaptive ptime is enabled

    // - bitrate configured in the rtp_parameter encodings settings

    const int kDefaultBitrateBps = 32000;

    config_.min_bitrate_bps = kDefaultBitrateBps;

    config_.max_bitrate_bps = kDefaultBitrateBps;
@@ -1057,13 +1056,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream 
          config_.min_bitrate_bps,

          static_cast<int>(adaptive_ptime_config_.min_encoder_bitrate.bps()));

    }



    if (rtp_parameters_.encodings[0].min_bitrate_bps) {

      config_.min_bitrate_bps = *rtp_parameters_.encodings[0].min_bitrate_bps;

    }

    if (rtp_parameters_.encodings[0].max_bitrate_bps) {

      config_.max_bitrate_bps = *rtp_parameters_.encodings[0].max_bitrate_bps;

    }

  }



  void UpdateSendCodecSpec(

third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc

+57 −0
Original line number Diff line number Diff line
@@ -29,6 +29,7 @@ 
#include "media/base/media_constants.h"

#include "media/engine/fake_webrtc_call.h"

#include "modules/audio_device/include/mock_audio_device.h"

#include "modules/audio_mixer/audio_mixer_impl.h"

#include "modules/audio_processing/include/mock_audio_processing.h"

#include "rtc_base/arraysize.h"

#include "rtc_base/byte_order.h"
@@ -444,6 +445,10 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> { 
    return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps;

  }



  int GetMaxBitrate(int32_t ssrc) {

    return GetSendStreamConfig(ssrc).max_bitrate_bps;

  }



  const absl::optional<std::string>& GetAudioNetworkAdaptorConfig(

      int32_t ssrc) {

    return GetSendStreamConfig(ssrc).audio_network_adaptor_config;
@@ -471,6 +476,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> { 


    // Verify that the codec settings have the expected bitrate.

    EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrcX));

    EXPECT_EQ(expected_codec_bitrate, GetMaxBitrate(kSsrcX));

  }



  void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps,
@@ -3741,6 +3747,57 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) { 
  }

}



TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) {

  rtc::AutoThread main_thread;

  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =

      webrtc::CreateDefaultTaskQueueFactory();

  rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =

      webrtc::test::MockAudioDeviceModule::CreateNice();

  webrtc::FieldTrialBasedConfig field_trials;

  FakeAudioSource source;

  cricket::WebRtcVoiceEngine engine(task_queue_factory.get(), adm.get(),

                                    webrtc::CreateBuiltinAudioEncoderFactory(),

                                    webrtc::CreateBuiltinAudioDecoderFactory(),

                                    nullptr, nullptr, nullptr, field_trials);

  engine.Init();

  webrtc::RtcEventLogNull event_log;

  webrtc::Call::Config call_config(&event_log);

  call_config.trials = &field_trials;

  call_config.task_queue_factory = task_queue_factory.get();

  {

    webrtc::AudioState::Config config;

    config.audio_mixer = webrtc::AudioMixerImpl::Create();

    config.audio_device_module =

        webrtc::test::MockAudioDeviceModule::CreateNice();

    call_config.audio_state = webrtc::AudioState::Create(config);

  }

  auto call = absl::WrapUnique(webrtc::Call::Create(call_config));

  cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),

                                           cricket::AudioOptions(),

                                           webrtc::CryptoOptions(), call.get());

  {

    cricket::AudioSendParameters params;

    params.codecs.push_back(cricket::AudioCodec(1, "opus", 48000, 32000, 2));

    params.extensions.push_back(webrtc::RtpExtension(

        webrtc::RtpExtension::kTransportSequenceNumberUri, 1));

    EXPECT_TRUE(channel.SetSendParameters(params));

  }

  constexpr int kSsrc = 1234;

  {

    cricket::StreamParams params;

    params.add_ssrc(kSsrc);

    channel.AddSendStream(params);

  }

  channel.SetAudioSend(kSsrc, true, nullptr, &source);

  channel.SetSend(true);

  webrtc::RtpParameters params = channel.GetRtpSendParameters(kSsrc);

  for (int max_bitrate : {-10, -1, 0, 10000}) {

    params.encodings[0].max_bitrate_bps = max_bitrate;

    channel.SetRtpSendParameters(

        kSsrc, params, [](webrtc::RTCError error) { EXPECT_TRUE(error.ok()); });

  }

}



TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {

  for (bool use_null_apm : {false, true}) {

    std::vector<webrtc::AudioCodecSpec> specs;