Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/audio/audio_send_stream.h"
 
+#include <algorithm>
 #include <string>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
+
+namespace {
+
+constexpr char kOpusCodecName[] = "opus";
+
+bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
+  return (_stricmp(codec.plname, ref_name) == 0);
+}
+
+template <typename T>
+static T MinPositive(T a, T b) {
+  if (a <= 0) {
+    return b;
+  }
+  if (b <= 0) {
+    return a;
+  }
+  return std::min(a, b);
+}
+
+}  // namespace
+
 std::string AudioSendStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{ssrc: " << ssrc;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
     if (i != extensions.size() - 1) {
       ss << ", ";
     }
   }
@@ skipping 52 matching lines @@
     if (extension.uri == RtpExtension::kAbsSendTimeUri) {
       channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kAudioLevelUri) {
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
+  SetSendCodecs();
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
 }
 
@@ skipping 163 matching lines @@
   return config_;
 }
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
+
+// Apply current codec settings to a single voe::Channel used for sending.
+bool AudioSendStream::SetSendCodecs() {
+  ScopedVoEInterface<VoECodec> codec(voice_engine());
+  const int channel = config_.voe_channel_id;
+
+  // Disable VAD and FEC unless we know the other side wants them.
+  codec->SetVADStatus(channel, false);
+  codec->SetFECStatus(channel, false);
+
+  // Set the codec immediately, since SetVADStatus() depends on whether
+  // the current codec is mono or stereo.
+  if (!SetSendCodec(config_.send_codec_spec.codec_inst)) {
+    return false;
+  }
+
+  // FEC should be enabled after SetSendCodec.
+  if (config_.send_codec_spec.enable_codec_fec) {
+    LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
+                 << channel;
+    if (codec->SetFECStatus(channel, true) == -1) {
+      // Enable codec internal FEC. Treat any failure as fatal internal error.
+      // TODO(minyue): use normal logging.
+      // LOG_RTCERR2(SetFECStatus, channel, true);
+      return false;
+    }
+  }
+
+  if (IsCodec(config_.send_codec_spec.codec_inst, kOpusCodecName)) {
+    // DTX and maxplaybackrate should be set after SetSendCodec. Because current
+    // send codec has to be Opus.
+
+    // Set Opus internal DTX.
+    LOG(LS_INFO) << "Attempt to "
+                 << (config_.send_codec_spec.enable_opus_dtx ? "enable"
+                                                             : "disable")
+                 << " Opus DTX on channel " << channel;
+    if (codec->SetOpusDtx(channel, config_.send_codec_spec.enable_opus_dtx)) {
+      // TODO(minyue): use normal logging.
+      // LOG_RTCERR2(SetOpusDtx, channel,
+      // config_.send_codec_spec.enable_opus_dtx);
+      return false;
+    }
+
+    // If opus_max_playback_rate <= 0, the default maximum playback rate
+    // (48 kHz) will be used.
+    if (config_.send_codec_spec.opus_max_playback_rate > 0) {
+      LOG(LS_INFO) << "Attempt to set maximum playback rate to "
+                   << config_.send_codec_spec.opus_max_playback_rate
+                   << " Hz on channel " << channel;
+      if (codec->SetOpusMaxPlaybackRate(
+              channel, config_.send_codec_spec.opus_max_playback_rate) == -1) {
+        // TODO(minyue): use normal logging.
+        // LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
+        //             config_.send_codec_spec.opus_max_playback_rate);
+        return false;
+      }
+    }
+  }
+  // TODO(solenberg): ApplyMaxSendBitrate() yields another call to
+  // SetSendCodec(). Check if it is possible to fuse with the previous call
+  // in this function.
+  ApplyMaxSendBitrate();

[minyue-webrtc, 2016/10/13 12:48:08]

The only reason to place this here is, as far as I can see, the SetSendCodec
inside ApplyMaxSendBitrate can fail. But, SetSendCodec on line 313 can also
fail. I think it is totally fine to do clamping on
|config_.send_codec_spec.codec_inst.rate| before ctor.

+
+  // Set the CN payloadtype and the VAD status.
+  if (config_.send_codec_spec.cng_payload_type != -1) {
+    // The CN payload type for 8000 Hz clockrate is fixed at 13.
+    if (config_.send_codec_spec.cng_plfreq != 8000) {
+      webrtc::PayloadFrequencies cn_freq;
+      switch (config_.send_codec_spec.cng_plfreq) {
+        case 16000:
+          cn_freq = webrtc::kFreq16000Hz;
+          break;
+        case 32000:
+          cn_freq = webrtc::kFreq32000Hz;
+          break;
+        default:
+          RTC_NOTREACHED();
+          return false;
+      }
+      if (codec->SetSendCNPayloadType(channel,
+                                      config_.send_codec_spec.cng_payload_type,
+                                      cn_freq) == -1) {
+        // TODO(minyue): use normal logging.
+        // LOG_RTCERR3(SetSendCNPayloadType, channel,
+        //            config_.send_codec_spec.cng_payload_type, cn_freq);
+
+        // TODO(ajm): This failure condition will be removed from VoE.
+        // Restore the return here when we update to a new enough webrtc.
+        //
+        // Not returning false because the SetSendCNPayloadType will fail if
+        // the channel is already sending.
+        // This can happen if the remote description is applied twice, for
+        // example in the case of ROAP on top of JSEP, where both side will
+        // send the offer.
+      }
+    }
+
+    // Only turn on VAD if we have a CN payload type that matches the
+    // clockrate for the codec we are going to use.
+    if (config_.send_codec_spec.cng_plfreq ==
+            config_.send_codec_spec.codec_inst.plfreq &&
+        config_.send_codec_spec.codec_inst.channels == 1) {
+      // TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
+      // interaction between VAD and Opus FEC.
+      LOG(LS_INFO) << "Enabling VAD";
+      if (codec->SetVADStatus(channel, true) == -1) {
+        // TODO(minyue): use normal logging.
+        // LOG_RTCERR2(SetVADStatus, channel, true);
+        return false;
+      }
+    }
+  }
+  return true;
+}
+
+bool AudioSendStream::SetSendCodec(const webrtc::CodecInst& send_codec) {
+  // TODO(minyue): avoid ToString
+  // LOG(LS_INFO) << "Send channel " << channel <<  " selected voice codec "
+  //              << ToString(send_codec) << ", bitrate=" << send_codec.rate;
+
+  ScopedVoEInterface<VoECodec> codec(voice_engine());
+  int channel = config_.voe_channel_id;
+
+  webrtc::CodecInst current_codec = {0};
+  if (codec->GetSendCodec(channel, current_codec) == 0 &&
+      (send_codec == current_codec)) {
+    // Codec is already configured, we can return without setting it again.
+    return true;
+  }
+
+  if (codec->SetSendCodec(channel, send_codec) == -1) {
+    // TODO(minyue): use normal logging.
+    // LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec));
+    return false;
+  }
+  return true;
+}
+
+bool AudioSendStream::ApplyMaxSendBitrate() {
+  int bps = config_.max_send_bitrate_bps;
+
+  // Bitrate is auto by default.
+  // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
+  // SetMaxSendBandwith(0), the second call removes the previous limit.
+  if (bps <= 0) {
+    return true;
+  }
+
+  if (config_.send_codec_spec.codec_inst.pltype == -1) {
+    LOG(LS_INFO) << "The send codec has not been set up yet. "
+                 << "The send bitrate setting will be applied later.";
+    return true;
+  }
+
+  webrtc::CodecInst codec = config_.send_codec_spec.codec_inst;
+  bool is_multi_rate = IsCodecMultiRate(codec);
+
+  if (is_multi_rate) {
+    // If codec is multi-rate then just set the bitrate.
+    int max_bitrate_bps = MaxBitrateBps(codec);
+    codec.rate = std::min(bps, max_bitrate_bps);
+    LOG(LS_INFO) << "Setting codec " << codec.plname << " to bitrate " << bps
+                 << " bps.";
+    if (!SetSendCodec(codec)) {
+      LOG(LS_ERROR) << "Failed to set codec " << codec.plname << " to bitrate "
+                    << bps << " bps.";
+      return false;
+    }
+    return true;
+  } else {
+    // If codec is not multi-rate and |bps| is less than the fixed bitrate
+    // then fail. If codec is not multi-rate and |bps| exceeds or equal the
+    // fixed bitrate then ignore.
+    if (bps < codec.rate) {
+      LOG(LS_ERROR) << "Failed to set codec " << codec.plname << " to bitrate "
+                    << bps << " bps"
+                    << ", requires at least " << codec.rate << " bps.";
+      return false;
+    }
+    return true;
+  }
+}
+
+bool AudioSendStream::IsCodecMultiRate(const webrtc::CodecInst& codec) const {
+  for (size_t i = 0; i < arraysize(config_.codec_prefs); ++i) {
+    if (IsCodec(codec, config_.codec_prefs[i].name) &&
+        config_.codec_prefs[i].clockrate == codec.plfreq) {
+      return config_.codec_prefs[i].is_multi_rate;
+    }
+  }
+  return false;
+}
+
+int AudioSendStream::MaxBitrateBps(const webrtc::CodecInst& codec) const {
+  for (size_t i = 0; i < arraysize(config_.codec_prefs); ++i) {
+    if (IsCodec(codec, config_.codec_prefs[i].name) &&
+        config_.codec_prefs[i].clockrate == codec.plfreq) {
+      return config_.codec_prefs[i].max_bitrate_bps;
+    }
+  }
+  return 0;
+}
+
 }  // namespace internal
 }  // namespace webrtc