Backed out 4 changesets (bug 1437366) for mochitest failures on /test_peerConnection_basicAudio_forced_higher_rate.html

Backed out changeset 57826a5b03e8 (bug 1437366) Backed out changeset a26852df062b (bug 1437366) Backed out changeset abca0eb36d33 (bug 1437366) Backed out changeset 41bb4a676575 (bug 1437366) --HG-- extra : rebase_source : d81a0833db5d331fe928100cf88d1b9cc9253a1b
2018-03-08 19:27:25 +02:00 · 2018-03-08 19:27:25 +02:00 · e73ecbc91f
--- a/dom/media/CubebUtils.cpp
+++ b/dom/media/CubebUtils.cpp
@ -33,9 +33,6 @@
 #define PREF_CUBEB_BACKEND "media.cubeb.backend"
 #define PREF_CUBEB_LATENCY_PLAYBACK "media.cubeb_latency_playback_ms"
 #define PREF_CUBEB_LATENCY_MSG "media.cubeb_latency_msg_frames"
-// Allows to get something non-default for the preferred sample-rate, to allow
-// troubleshooting in the field and testing.
-#define PREF_CUBEB_FORCE_SAMPLE_RATE "media.cubeb.force_sample_rate"
 #define PREF_CUBEB_LOGGING_LEVEL "media.cubeb.logging_level"
 #define PREF_CUBEB_SANDBOX "media.cubeb.sandbox"

@ -125,10 +122,6 @@ cubeb* sCubebContext;
 double sVolumeScale = 1.0;
 uint32_t sCubebPlaybackLatencyInMilliseconds = 100;
 uint32_t sCubebMSGLatencyInFrames = 512;
-// If sCubebForcedSampleRate is zero, PreferredSampleRate will return the
-// preferred sample-rate for the audio backend in use. Otherwise, it will be
-// used as the preferred sample-rate.
-uint32_t sCubebForcedSampleRate = 0;
 bool sCubebPlaybackLatencyPrefSet = false;
 bool sCubebMSGLatencyPrefSet = false;
 bool sAudioStreamInitEverSucceeded = false;
@ -244,9 +237,6 @@ void PrefChanged(const char* aPref, void* aClosure)
    // We don't want to limit the upper limit too much, so that people can
    // experiment.
    sCubebMSGLatencyInFrames = std::min<uint32_t>(std::max<uint32_t>(value, 128), 1e6);
-  } else if (strcmp(aPref, PREF_CUBEB_FORCE_SAMPLE_RATE) == 0) {
-    StaticMutexAutoLock lock(sMutex);
-    sCubebForcedSampleRate = Preferences::GetUint(aPref);
  } else if (strcmp(aPref, PREF_CUBEB_LOGGING_LEVEL) == 0) {
    nsAutoCString value;
    Preferences::GetCString(aPref, value);
@ -333,9 +323,6 @@ bool InitPreferredSampleRate()

 uint32_t PreferredSampleRate()
 {
-  if (sCubebForcedSampleRate) {
-    return sCubebForcedSampleRate;
-  }
  if (!InitPreferredSampleRate()) {
    return 44100;
  }
@ -576,7 +563,6 @@ void InitLibrary()
  Preferences::RegisterCallbackAndCall(PrefChanged, PREF_VOLUME_SCALE);
  Preferences::RegisterCallbackAndCall(PrefChanged, PREF_CUBEB_LATENCY_PLAYBACK);
  Preferences::RegisterCallbackAndCall(PrefChanged, PREF_CUBEB_LATENCY_MSG);
-  Preferences::RegisterCallback(PrefChanged, PREF_CUBEB_FORCE_SAMPLE_RATE);
  Preferences::RegisterCallbackAndCall(PrefChanged, PREF_CUBEB_BACKEND);
  Preferences::RegisterCallbackAndCall(PrefChanged, PREF_CUBEB_SANDBOX);
  if (MOZ_LOG_TEST(gCubebLog, LogLevel::Verbose)) {
@ -606,7 +592,6 @@ void ShutdownLibrary()
  Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_SANDBOX);
  Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_BACKEND);
  Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LATENCY_PLAYBACK);
-  Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_FORCE_SAMPLE_RATE);
  Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LATENCY_MSG);
  Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LOGGING_LEVEL);

--- a/dom/media/tests/mochitest/mochitest.ini
+++ b/dom/media/tests/mochitest/mochitest.ini
@ -15,7 +15,6 @@ support-files =
  sdpUtils.js
  addTurnsSelfsignedCert.js
  parser_rtp.js
-  peerconnection_audio_forced_sample_rate.js
  !/dom/canvas/test/captureStream_common.js
  !/dom/canvas/test/webgl-mochitest/webgl-util.js
  !/dom/media/test/manifest.js
@ -105,8 +104,6 @@ skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emula
 skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emulator)
 [test_peerConnection_basicAudio.html]
 skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emulator)
-[test_peerConnection_basicAudio_forced_lower_rate.html]
-[test_peerConnection_basicAudio_forced_higher_rate.html]
 [test_peerConnection_audioSynchronizationSources.html]
 skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emulator)
 [test_peerConnection_audioSynchronizationSourcesUnidirectional.html]
@ -328,4 +325,3 @@ skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emula
 [test_peerConnection_verifyDescriptions.html]
 skip-if = (android_version == '18')
 [test_fingerprinting_resistance.html]
-[test_forceSampleRate.html]
--- a/dom/media/tests/mochitest/peerconnection_audio_forced_sample_rate.js
+++ b/dom/media/tests/mochitest/peerconnection_audio_forced_sample_rate.js
@ -1,38 +0,0 @@
-// This function takes a sample-rate, and tests that audio flows correctly when
-// the sampling-rate at which the MSG runs is not one of the sampling-rates that
-// the MediaPipeline can work with.
-// It is in a separate file because we have an MSG per document, and we want to
-// test multiple sample-rates, so we include it in multiple HTML mochitest
-// files.
-function test_peerconnection_audio_forced_sample_rate(forcedSampleRate) {
-  scriptsReady.then(function () {
-    pushPrefs(
-      ["media.cubeb.force_sample_rate", forcedSampleRate]
-    ).then(function () {
-      runNetworkTest(function (options) {
-        let test = new PeerConnectionTest(options);
-        let ac = new AudioContext();
-        test.setMediaConstraints([{
-          audio: true
-        }], []);
-        test.chain.replace("PC_LOCAL_GUM", [
-          function PC_LOCAL_WEBAUDIO_SOURCE(test) {
-            let oscillator = ac.createOscillator();
-            oscillator.type = 'sine';
-            oscillator.frequency.value = 700;
-            oscillator.start();
-            let dest = ac.createMediaStreamDestination();
-            oscillator.connect(dest);
-            test.pcLocal.attachLocalStream(dest.stream);
-          }
-        ]);
-        test.chain.append([
-          function CHECK_REMOTE_AUDIO_FLOW(test) {
-            return test.pcRemote.checkReceivingToneFrom(ac, test.pcLocal);
-          }
-        ]);
-        test.run();
-      });
-    });
-  })
-}
--- a/dom/media/tests/mochitest/test_forceSampleRate.html
+++ b/dom/media/tests/mochitest/test_forceSampleRate.html
@ -1,23 +0,0 @@
-<!DOCTYPE HTML>
-<html>
-<head>
-  <title>Test the pref media.cubeb.force_sample_rate</title>
-  <script type="text/javascript" src="/MochiKit/MochiKit.js"></script>
-  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
-  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
-</head>
-<body>
-<script class="testbody" type="text/javascript">
-const WEIRD_SAMPLE_RATE = 44101;
-
-SimpleTest.waitForExplicitFinish();
-SpecialPowers.pushPrefEnv({"set": [
-  ["media.cubeb.force_sample_rate", WEIRD_SAMPLE_RATE]
-]}).then(function() {
-  var ac = new AudioContext();
-  is(ac.sampleRate, WEIRD_SAMPLE_RATE, "Forced sample-rate set successfully.");
-  SimpleTest.finish();
-});
-</script>
-</pre>
-</body>
--- a/dom/media/tests/mochitest/test_peerConnection_basicAudio_forced_higher_rate.html
+++ b/dom/media/tests/mochitest/test_peerConnection_basicAudio_forced_higher_rate.html
@ -1,20 +0,0 @@
-<!DOCTYPE HTML>
-<html>
-<head>
-  <script type="application/javascript" src="pc.js"></script>
-  <script type="application/javascript" src="peerconnection_audio_forced_sample_rate.js"></script>
-</head>
-<body>
-<pre id="test">
-<script type="application/javascript">
-  createHTML({
-    bug: "1437366",
-    title: "Basic audio-only peer connection, with the MSG running at a rate not supported by the MediaPipeline (96000Hz)"
-  });
-
-  test_peerconnection_audio_forced_sample_rate(96000);
-
-</script>
-</pre>
-</body>
-</html>
--- a/dom/media/tests/mochitest/test_peerConnection_basicAudio_forced_lower_rate.html
+++ b/dom/media/tests/mochitest/test_peerConnection_basicAudio_forced_lower_rate.html
@ -1,20 +0,0 @@
-<!DOCTYPE HTML>
-<html>
-<head>
-  <script type="application/javascript" src="pc.js"></script>
-  <script type="application/javascript" src="peerconnection_audio_forced_sample_rate.js"></script>
-</head>
-<body>
-<pre id="test">
-<script type="application/javascript">
-  createHTML({
-    bug: "1437366",
-    title: "Basic audio-only peer connection, with the MSG running at a rate not supported by the MediaPipeline (24000Hz)"
-  });
-
-test_peerconnection_audio_forced_sample_rate(24000);
-
-</script>
-</pre>
-</body>
-</html>
--- a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
@ -11,7 +11,6 @@
 #include <math.h>

 #include "AudioSegment.h"
-#include "AudioConverter.h"
 #include "AutoTaskQueue.h"
 #include "CSFLog.h"
 #include "DOMMediaStream.h"
@ -488,156 +487,76 @@ public:
    , mTaskQueue(
        new AutoTaskQueue(GetMediaThreadPool(MediaThreadType::WEBRTC_DECODER),
                          "AudioProxy"))
-    , mAudioConverter(nullptr)
  {
    MOZ_ASSERT(mConduit);
    MOZ_COUNT_CTOR(AudioProxyThread);
  }

-  // This function is the identity if aInputRate is supported.
-  // Else, it returns a rate that is supported, that ensure no loss in audio
-  // quality: the sampling rate returned is always greater to the inputed
-  // sampling-rate, if they differ..
-  uint32_t AppropriateSendingRateForInputRate(uint32_t aInputRate)
-  {
-    AudioSessionConduit* conduit =
-      static_cast<AudioSessionConduit*>(mConduit.get());
-    if (conduit->IsSamplingFreqSupported(aInputRate)) {
-      return aInputRate;
-    }
-    if (aInputRate < 16000) {
-      return 16000;
-    } else if (aInputRate < 32000) {
-      return 32000;
-    } else if (aInputRate < 44100) {
-      return 44100;
-    } else {
-      return 48000;
-    }
-  }
-
-  // From an arbitrary AudioChunk at sampling-rate aRate, process the audio into
-  // something the conduit can work with (or send silence if the track is not
-  // enabled), and send the audio in 10ms chunks to the conduit.
-  void InternalProcessAudioChunk(TrackRate aRate,
-                                 const AudioChunk& aChunk,
-                                 bool aEnabled)
+  void InternalProcessAudioChunk(TrackRate rate,
+                                 const AudioChunk& chunk,
+                                 bool enabled)
  {
    MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());

-    // Convert to interleaved 16-bits integer audio, with a maximum of two
+    // Convert to interleaved, 16-bits integer audio, with a maximum of two
    // channels (since the WebRTC.org code below makes the assumption that the
-    // input audio is either mono or stereo), with a sample-rate rate that is
-    // 16, 32, 44.1, or 48kHz.
-    uint32_t outputChannels = aChunk.ChannelCount() == 1 ? 1 : 2;
-    int32_t transmissionRate = AppropriateSendingRateForInputRate(aRate);
+    // input audio is either mono or stereo).
+    uint32_t outputChannels = chunk.ChannelCount() == 1 ? 1 : 2;
+    const int16_t* samples = nullptr;
+    UniquePtr<int16_t[]> convertedSamples;

    // We take advantage of the fact that the common case (microphone directly
    // to PeerConnection, that is, a normal call), the samples are already
    // 16-bits mono, so the representation in interleaved and planar is the
    // same, and we can just use that.
-    if (aEnabled &&
-        outputChannels == 1 &&
-        aChunk.mBufferFormat == AUDIO_FORMAT_S16 &&
-        transmissionRate == aRate) {
-      const int16_t* samples = aChunk.ChannelData<int16_t>().Elements()[0];
-      PacketizeAndSend(samples,
-                       transmissionRate,
-                       outputChannels,
-                       aChunk.mDuration);
-      return;
-    }
-
-    uint32_t sampleCount = aChunk.mDuration * outputChannels;
-    if (mInterleavedAudio.Length() < sampleCount) {
-      mInterleavedAudio.SetLength(sampleCount);
-    }
-
-    if (!aEnabled || aChunk.mBufferFormat == AUDIO_FORMAT_SILENCE) {
-      PodZero(mInterleavedAudio.Elements(), sampleCount);
-    } else if (aChunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
-      DownmixAndInterleave(aChunk.ChannelData<float>(),
-                           aChunk.mDuration,
-                           aChunk.mVolume,
-                           outputChannels,
-                           mInterleavedAudio.Elements());
-    } else if (aChunk.mBufferFormat == AUDIO_FORMAT_S16) {
-      DownmixAndInterleave(aChunk.ChannelData<int16_t>(),
-                           aChunk.mDuration,
-                           aChunk.mVolume,
-                           outputChannels,
-                           mInterleavedAudio.Elements());
-    }
-    int16_t* inputAudio = mInterleavedAudio.Elements();
-    size_t inputAudioFrameCount = aChunk.mDuration;
-
-    AudioConfig inputConfig(AudioConfig::ChannelLayout(outputChannels),
-                            aRate,
-                            AudioConfig::FORMAT_S16);
-    AudioConfig outputConfig(AudioConfig::ChannelLayout(outputChannels),
-                             transmissionRate,
-                             AudioConfig::FORMAT_S16);
-    // Resample to an acceptable sample-rate for the sending side
-    if (!mAudioConverter ||
-        mAudioConverter->InputConfig() != inputConfig ||
-        mAudioConverter->OutputConfig() != outputConfig) {
-      mAudioConverter = MakeUnique<AudioConverter>(inputConfig, outputConfig);
-    }
-
-    int16_t* processedAudio = nullptr;
-    size_t framesProcessed =
-      mAudioConverter->Process(inputAudio, inputAudioFrameCount);
-
-    if (framesProcessed == 0) {
-      // In place conversion not possible, use a buffer.
-      framesProcessed =
-        mAudioConverter->Process(mOutputAudio,
-                                 inputAudio,
-                                 inputAudioFrameCount);
-      processedAudio = mOutputAudio.Data();
+    if (enabled && outputChannels == 1 &&
+        chunk.mBufferFormat == AUDIO_FORMAT_S16) {
+      samples = chunk.ChannelData<int16_t>().Elements()[0];
    } else {
-      processedAudio = inputAudio;
+      convertedSamples =
+        MakeUnique<int16_t[]>(chunk.mDuration * outputChannels);
+
+      if (!enabled || chunk.mBufferFormat == AUDIO_FORMAT_SILENCE) {
+        PodZero(convertedSamples.get(), chunk.mDuration * outputChannels);
+      } else if (chunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
+        DownmixAndInterleave(chunk.ChannelData<float>(),
+                             chunk.mDuration,
+                             chunk.mVolume,
+                             outputChannels,
+                             convertedSamples.get());
+      } else if (chunk.mBufferFormat == AUDIO_FORMAT_S16) {
+        DownmixAndInterleave(chunk.ChannelData<int16_t>(),
+                             chunk.mDuration,
+                             chunk.mVolume,
+                             outputChannels,
+                             convertedSamples.get());
+      }
+      samples = convertedSamples.get();
    }

-    PacketizeAndSend(processedAudio,
-                     transmissionRate,
-                     outputChannels,
-                     framesProcessed);
-  }
+    MOZ_ASSERT(!(rate % 100)); // rate should be a multiple of 100

-  // This packetizes aAudioData in 10ms chunks and sends it.
-  // aAudioData is interleaved audio data at a rate and with a channel count
-  // that is appropriate to send with the conduit.
-  void PacketizeAndSend(const int16_t* aAudioData,
-                        uint32_t aRate,
-                        uint32_t aChannels,
-                        uint32_t aFrameCount)
-  {
-    MOZ_ASSERT(AppropriateSendingRateForInputRate(aRate) == aRate);
-    MOZ_ASSERT(aChannels == 1 || aChannels == 2);
-    MOZ_ASSERT(aAudioData);
+    // Check if the rate or the number of channels has changed since the last
+    // time we came through. I realize it may be overkill to check if the rate
+    // has changed, but I believe it is possible (e.g. if we change sources) and
+    // it costs us very little to handle this case.

-    uint32_t audio_10ms = aRate / 100;
+    uint32_t audio_10ms = rate / 100;

    if (!mPacketizer || mPacketizer->PacketSize() != audio_10ms ||
-        mPacketizer->Channels() != aChannels) {
-      // It's the right thing to drop the bit of audio still in the packetizer:
-      // we don't want to send to the conduit audio that has two different
-      // rates while telling it that it has a constante rate.
+        mPacketizer->Channels() != outputChannels) {
+      // It's ok to drop the audio still in the packetizer here.
      mPacketizer = MakeUnique<AudioPacketizer<int16_t, int16_t>>(
-        audio_10ms, aChannels);
-      mPacket = MakeUnique<int16_t[]>(audio_10ms * aChannels);
+        audio_10ms, outputChannels);
+      mPacket = MakeUnique<int16_t[]>(audio_10ms * outputChannels);
    }

-    mPacketizer->Input(aAudioData, aFrameCount);
+    mPacketizer->Input(samples, chunk.mDuration);

    while (mPacketizer->PacketsAvailable()) {
      mPacketizer->Output(mPacket.get());
-      mConduit->SendAudioFrame(mPacket.get(),
-                               mPacketizer->PacketSize(),
-                               aRate,
-                               mPacketizer->Channels(),
-                               0);
+      mConduit->SendAudioFrame(
+        mPacket.get(), mPacketizer->PacketSize(), rate, mPacketizer->Channels(), 0);
    }
  }

@ -669,9 +588,6 @@ protected:
  UniquePtr<AudioPacketizer<int16_t, int16_t>> mPacketizer;
  // A buffer to hold a single packet of audio.
  UniquePtr<int16_t[]> mPacket;
-  nsTArray<int16_t> mInterleavedAudio;
-  AlignedShortBuffer mOutputAudio;
-  UniquePtr<AudioConverter> mAudioConverter;
 };

 static char kDTLSExporterLabel[] = "EXTRACTOR-dtls_srtp";
@ -2049,24 +1965,14 @@ public:
    , mMaybeTrackNeedsUnmute(true)
  {
    MOZ_RELEASE_ASSERT(mSource, "Must be used with a SourceMediaStream");
-  }
-
-  virtual ~GenericReceiveListener()
-  {
-    NS_ReleaseOnMainThreadSystemGroup(
-      "GenericReceiveListener::track_", mTrack.forget());
-  }
-
-  void AddTrackToSource(uint32_t aRate = 0)
-  {
-    MOZ_ASSERT((aRate != 0 && mTrack->AsAudioStreamTrack()) ||
-               mTrack->AsVideoStreamTrack());

    if (mTrack->AsAudioStreamTrack()) {
      mSource->AddAudioTrack(
-          mTrackId, aRate, 0, new AudioSegment());
+          mTrackId, mSource->GraphRate(), 0, new AudioSegment());
    } else if (mTrack->AsVideoStreamTrack()) {
      mSource->AddTrack(mTrackId, 0, new VideoSegment());
+    } else {
+      MOZ_ASSERT_UNREACHABLE("Unknown track type");
    }
    CSFLogDebug(
      LOGTAG,
@ -2080,6 +1986,12 @@ public:
    mSource->AddListener(this);
  }

+  virtual ~GenericReceiveListener()
+  {
+    NS_ReleaseOnMainThreadSystemGroup(
+      "GenericReceiveListener::track_", mTrack.forget());
+  }
+
  void AddSelf()
  {
    if (!mListening) {
@ -2203,7 +2115,6 @@ public:
                          "AudioPipelineListener"))
    , mLastLog(0)
  {
-    AddTrackToSource(mRate);
  }

  // Implement MediaStreamListener
@ -2235,13 +2146,10 @@ private:
  void NotifyPullImpl(StreamTime aDesiredTime)
  {
    uint32_t samplesPer10ms = mRate / 100;
-
-    // mSource's rate is not necessarily the same as the graph rate, since there
-    // are sample-rate constraints on the inbound audio: only 16, 32, 44.1 and
-    // 48kHz are supported. The audio frames we get here is going to be
-    // resampled when inserted into the graph.
-    TrackTicks desired = mSource->TimeToTicksRoundUp(mRate, aDesiredTime);
-    TrackTicks framesNeeded = desired - mPlayedTicks;
+    // Determine how many frames we need.
+    // As we get frames from conduit_ at the same rate as the graph's rate,
+    // the number of frames needed straightfully determined.
+    TrackTicks framesNeeded = aDesiredTime - mPlayedTicks;

    while (framesNeeded >= 0) {
      const int scratchBufferLength =
@ -2405,7 +2313,6 @@ public:
        LayerManager::CreateImageContainer(ImageContainer::ASYNCHRONOUS))
    , mMutex("Video PipelineListener")
  {
-    AddTrackToSource();
  }

  // Implement MediaStreamListener