Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 446 matching lines @@
     // G722 should be advertised as 8000 Hz because of the RFC "bug".
     {kG722CodecName, 8000, 1, 9, false, {10, 20, 30, 40, 50, 60}},
     {kIlbcCodecName, 8000, 1, 102, false, {20, 30, 40, 60}},
     {kPcmuCodecName, 8000, 1, 0, false, {10, 20, 30, 40, 50, 60}},
     {kPcmaCodecName, 8000, 1, 8, false, {10, 20, 30, 40, 50, 60}},
     {kCnCodecName, 32000, 1, 106, false, {}},
     {kCnCodecName, 16000, 1, 105, false, {}},
     {kCnCodecName, 8000, 1, 13, false, {}},
     {kDtmfCodecName, 8000, 1, 126, false, {}}
 };
-} // namespace {
 
-bool SendCodecSpec::operator==(const SendCodecSpec& rhs) const {
-  if (nack_enabled != rhs.nack_enabled) {
-    return false;
-  }
-  if (transport_cc_enabled != rhs.transport_cc_enabled) {
-    return false;
-  }
-  if (enable_codec_fec != rhs.enable_codec_fec) {
-    return false;
-  }
-  if (enable_opus_dtx != rhs.enable_opus_dtx) {
-    return false;
-  }
-  if (opus_max_playback_rate != rhs.opus_max_playback_rate) {
-    return false;
-  }
-  if (red_payload_type != rhs.red_payload_type) {
-    return false;
-  }
-  if (cng_payload_type != rhs.cng_payload_type) {
-    return false;
-  }
-  if (cng_plfreq != rhs.cng_plfreq) {
-    return false;
-  }
-  if (codec_inst != rhs.codec_inst) {
-    return false;
-  }
-  return true;
-}
-
-bool SendCodecSpec::operator!=(const SendCodecSpec& rhs) const {
-  return !(*this == rhs);
-}
+}  // namespace {
 
 bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
                                     webrtc::CodecInst* out) {
   return WebRtcVoiceCodecs::ToCodecInst(in, out);
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
     : WebRtcVoiceEngine(adm, decoder_factory, new VoEWrapper()) {
@@ skipping 620 matching lines @@
 
   // Add telephone-event codec last
   map_format({kDtmfCodecName, 8000, 1});
 
   return out;
 }
 
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioSource::Sink {
  public:
-  WebRtcAudioSendStream(int ch,
-                        webrtc::AudioTransport* voe_audio_transport,
-                        uint32_t ssrc,
-                        const std::string& c_name,
-                        const SendCodecSpec& send_codec_spec,
-                        const std::vector<webrtc::RtpExtension>& extensions,
-                        webrtc::Call* call,
-                        webrtc::Transport* send_transport)
+  WebRtcAudioSendStream(
+      int ch,
+      webrtc::AudioTransport* voe_audio_transport,
+      uint32_t ssrc,
+      const std::string& c_name,
+      const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec,
+      const std::vector<webrtc::RtpExtension>& extensions,
+      int max_send_bitrate_bps,
+      webrtc::Call* call,
+      webrtc::Transport* send_transport)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
         config_(send_transport),
+        max_send_bitrate_bps_(max_send_bitrate_bps),
         rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
     RTC_DCHECK_GE(ch, 0);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
     config_.rtp.ssrc = ssrc;
     config_.rtp.c_name = c_name;
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
     RecreateAudioSendStream(send_codec_spec);
   }
 
   ~WebRtcAudioSendStream() override {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     ClearSource();
     call_->DestroyAudioSendStream(stream_);
   }
 
-  void RecreateAudioSendStream(const SendCodecSpec& send_codec_spec) {
+  void RecreateAudioSendStream(
+      const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioSendStream(stream_);
       stream_ = nullptr;
     }
     config_.rtp.nack.rtp_history_ms =
         send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0;
+    config_.send_codec_spec = send_codec_spec;
+    config_.send_codec_spec.codec_inst.rate = DecideSendBitrate();
     RTC_DCHECK(!stream_);
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);
     UpdateSendState();
   }
 
   void RecreateAudioSendStream(
       const std::vector<webrtc::RtpExtension>& extensions) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioSendStream(stream_);
       stream_ = nullptr;
     }
     config_.rtp.extensions = extensions;
     if (webrtc::field_trial::FindFullName("WebRTC-AdaptAudioBitrate") ==
         "Enabled") {
       // TODO(mflodman): Keep testing this and set proper values.
       // Note: This is an early experiment currently only supported by Opus.
       config_.min_bitrate_kbps = kOpusMinBitrate;
       config_.max_bitrate_kbps = kOpusBitrateFb;
     }
 
     RTC_DCHECK(!stream_);
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);
     UpdateSendState();
   }
 
+  bool SetMaxSendBitrate(int bps) {
+    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+    if (!IsMaxSendBitrateValid(
+            MinPositive(bps, rtp_parameters_.encodings[0].max_bitrate_bps))) {
+      return false;
+    }
+    max_send_bitrate_bps_ = bps;
+
+    int new_sent_bitrate_bps = DecideSendBitrate();
+    if (config_.send_codec_spec.codec_inst.rate == new_sent_bitrate_bps)
+      return true;
+
+    // Recreate AudioSendStream with new bit rate.
+    config_.send_codec_spec.codec_inst.rate = new_sent_bitrate_bps;
+    if (stream_) {
+      call_->DestroyAudioSendStream(stream_);
+      stream_ = nullptr;
+    }
+    RTC_DCHECK(!stream_);
+    stream_ = call_->CreateAudioSendStream(config_);
+    RTC_CHECK(stream_);
+    UpdateSendState();
+    return true;
+  }
+
   bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
   }
 
   void SetSend(bool send) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     send_ = send;
     UpdateSendState();
@@ skipping 72 matching lines @@
   // Accessor to the VoE channel ID.
   int channel() const {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     return config_.voe_channel_id;
   }
 
   const webrtc::RtpParameters& rtp_parameters() const {
     return rtp_parameters_;
   }
 
-  void SetRtpParameters(const webrtc::RtpParameters& parameters) {
+  bool SetRtpParameters(const webrtc::RtpParameters& parameters) {
     RTC_CHECK_EQ(1UL, parameters.encodings.size());
+
+    if (!IsMaxSendBitrateValid(MinPositive(
+            max_send_bitrate_bps_, parameters.encodings[0].max_bitrate_bps))) {
+      return false;
+    }
+
     rtp_parameters_ = parameters;
+
+    // parameters.encodings[0].encodings[0].max_bitrate_bps could have changed.
+    SetMaxSendBitrate(max_send_bitrate_bps_);
+
     // parameters.encodings[0].active could have changed.
     UpdateSendState();
+
+    return true;
   }
 
  private:
   void UpdateSendState() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size());
     if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
       stream_->Start();
     } else {  // !send || source_ = nullptr
       stream_->Stop();
     }
   }
 
+  bool IsMaxSendBitrateValid(int bps) const {
+    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;
+    }
+
+    if (!WebRtcVoiceCodecs::IsCodecMultiRate(
+            config_.send_codec_spec.codec_inst) &&
+        bps < config_.send_codec_spec.codec_inst.rate) {
+      // If codec is not multi-rate and |max_send_bit_rate_| is less than the
+      // fixed bitrate then fail. If codec is not multi-rate and |bps| exceeds
+      // or
+      // equal the fixed bitrate then ignore.
+      LOG(LS_ERROR) << "Failed to set codec "
+                    << config_.send_codec_spec.codec_inst.plname
+                    << " to bitrate " << bps << " bps"
+                    << ", requires at least "
+                    << config_.send_codec_spec.codec_inst.rate << " bps.";
+      return false;
+    }
+
+    return true;
+  }
+
+  int DecideSendBitrate() const {

[minyue-webrtc, 2016/10/14 13:32:56]

Needed to split this into DecideSendBitrate and IsMaxSendBitrateValid because we
need to decide if a new max_send_bitrate_ or a new
rtp_parameter.max_bitrate_bps_ is valid, at various place.

+    const int bps = MinPositive(max_send_bitrate_bps_,
+                                rtp_parameters_.encodings[0].max_bitrate_bps);
+    const int current_rate = config_.send_codec_spec.codec_inst.rate;
+
+    // Bitrate is auto by default.
+    // TODO(bemasc): Fix this so that if SetMaxSendBitrate(50) is followed by
+    // SetMaxSendBitrate(0), the second call removes the previous limit.
+    if (bps <= 0) {
+      return current_rate;
+    }
+
+    if (config_.send_codec_spec.codec_inst.pltype == -1) {
+      return current_rate;
+    }
+
+    if (!WebRtcVoiceCodecs::IsCodecMultiRate(
+            config_.send_codec_spec.codec_inst)) {
+      return current_rate;
+    }
+
+    // If codec is multi-rate then just set the bitrate.
+    return std::min(bps, WebRtcVoiceCodecs::MaxBitrateBps(
+                             config_.send_codec_spec.codec_inst));
+  }
+
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
   webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
   webrtc::Call* call_ = nullptr;
   webrtc::AudioSendStream::Config config_;
   // The stream is owned by WebRtcAudioSendStream and may be reallocated if
   // configuration changes.
   webrtc::AudioSendStream* stream_ = nullptr;
 
   // Raw pointer to AudioSource owned by LocalAudioTrackHandler.
   // PeerConnection will make sure invalidating the pointer before the object
   // goes away.
   AudioSource* source_ = nullptr;
   bool send_ = false;
   bool muted_ = false;
+  int max_send_bitrate_bps_;
   webrtc::RtpParameters rtp_parameters_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
 };
 
 class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
  public:
   WebRtcAudioReceiveStream(
       int ch,
       uint32_t remote_ssrc,
@@ skipping 240 matching lines @@
 
   // TODO(deadbeef): Handle setting parameters with a list of codecs in a
   // different order (which should change the send codec).
   webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
   if (current_parameters.codecs != parameters.codecs) {
     LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
                   << "is not currently supported.";
     return false;
   }
 
-  if (!SetChannelSendParameters(it->second->channel(), parameters)) {
-    LOG(LS_WARNING) << "Failed to set send RtpParameters.";
-    return false;
-  }
+  // TODO(minyue): The following legacy actions go into
+  // |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end,
+  // though there are two difference:
+  // 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls
+  // |SetSendCodec| while |WebRtcAudioSendStream::SetRtpParameters()| calls
+  // |SetSendCodecs|. The outcome should be the same.
+  // 2. AudioSendStream can be recreated.
+
   // Codecs are handled at the WebRtcVoiceMediaChannel level.
   webrtc::RtpParameters reduced_params = parameters;
   reduced_params.codecs.clear();
-  it->second->SetRtpParameters(reduced_params);
-  return true;
+  return it->second->SetRtpParameters(reduced_params);

[minyue-webrtc, 2016/10/14 13:32:56]

possibility of returning false for this function is captured by
it->second->SetRtpParameters()

 }
 
 webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
     uint32_t ssrc) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
     LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
                     << "with ssrc " << ssrc << " which doesn't exist.";
     return webrtc::RtpParameters();
@@ skipping 148 matching lines @@
       }
       dtmf_payload_type_ = rtc::Optional<int>(codec.id);
       break;
     }
   }
 
   // Scan through the list to figure out the codec to use for sending, along
   // with the proper configuration for VAD, CNG, NACK and Opus-specific
   // parameters.
   // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
-  SendCodecSpec send_codec_spec;
+  webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
   {
     send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
 
     // Find send codec (the first non-telephone-event/CN codec).
     const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
         codecs, &send_codec_spec.codec_inst);
     if (!codec) {
       LOG(LS_WARNING) << "Received empty list of codecs.";
       return false;
     }
@@ skipping 53 matching lines @@
         break;
       }
     }
   }
 
   // Apply new settings to all streams.
   if (send_codec_spec_ != send_codec_spec) {
     send_codec_spec_ = std::move(send_codec_spec);
     for (const auto& kv : send_streams_) {
       kv.second->RecreateAudioSendStream(send_codec_spec_);
-      if (!SetSendCodecs(kv.second->channel(), kv.second->rtp_parameters())) {
-        return false;
-      }
     }
   }
 
   // Check if the transport cc feedback or NACK status has changed on the
   // preferred send codec, and in that case reconfigure all receive streams.
   if (recv_transport_cc_enabled_ != send_codec_spec_.transport_cc_enabled ||
       recv_nack_enabled_ != send_codec_spec_.nack_enabled) {
     LOG(LS_INFO) << "Recreate all the receive streams because the send "
                     "codec has changed.";
     recv_transport_cc_enabled_ = send_codec_spec_.transport_cc_enabled;
     recv_nack_enabled_ = send_codec_spec_.nack_enabled;
     for (auto& kv : recv_streams_) {
       kv.second->RecreateAudioReceiveStream(recv_transport_cc_enabled_,
                                             recv_nack_enabled_);
     }
   }
 
   send_codecs_ = codecs;
   return true;
 }
 
-// Apply current codec settings to a single voe::Channel used for sending.
-bool WebRtcVoiceMediaChannel::SetSendCodecs(
-    int channel,
-    const webrtc::RtpParameters& rtp_parameters) {
-  // Disable VAD and FEC unless we know the other side wants them.
-  engine()->voe()->codec()->SetVADStatus(channel, false);
-  engine()->voe()->codec()->SetFECStatus(channel, false);
-
-  // Set the codec immediately, since SetVADStatus() depends on whether
-  // the current codec is mono or stereo.
-  if (!SetSendCodec(channel, send_codec_spec_.codec_inst)) {
-    return false;
-  }
-
-  // FEC should be enabled after SetSendCodec.
-  if (send_codec_spec_.enable_codec_fec) {
-    LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
-                 << channel;
-    if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) {
-      // Enable codec internal FEC. Treat any failure as fatal internal error.
-      LOG_RTCERR2(SetFECStatus, channel, true);
-      return false;
-    }
-  }
-
-  if (IsCodec(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 "
-                 << (send_codec_spec_.enable_opus_dtx ? "enable" : "disable")
-                 << " Opus DTX on channel "
-                 << channel;
-    if (engine()->voe()->codec()->SetOpusDtx(channel,
-        send_codec_spec_.enable_opus_dtx)) {
-      LOG_RTCERR2(SetOpusDtx, channel, 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 (send_codec_spec_.opus_max_playback_rate > 0) {
-      LOG(LS_INFO) << "Attempt to set maximum playback rate to "
-                   << send_codec_spec_.opus_max_playback_rate
-                   << " Hz on channel "
-                   << channel;
-      if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
-          channel, send_codec_spec_.opus_max_playback_rate) == -1) {
-        LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
-                    send_codec_spec_.opus_max_playback_rate);
-        return false;
-      }
-    }
-  }
-  // TODO(solenberg): SetMaxSendBitrate() yields another call to SetSendCodec().
-  // Check if it is possible to fuse with the previous call in this function.
-  SetChannelSendParameters(channel, rtp_parameters);
-
-  // Set the CN payloadtype and the VAD status.
-  if (send_codec_spec_.cng_payload_type != -1) {
-    // The CN payload type for 8000 Hz clockrate is fixed at 13.
-    if (send_codec_spec_.cng_plfreq != 8000) {
-      webrtc::PayloadFrequencies cn_freq;
-      switch (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 (engine()->voe()->codec()->SetSendCNPayloadType(
-              channel, send_codec_spec_.cng_payload_type, cn_freq) == -1) {
-        LOG_RTCERR3(SetSendCNPayloadType, channel,
-                    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 (send_codec_spec_.cng_plfreq == send_codec_spec_.codec_inst.plfreq &&
-        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 (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
-        LOG_RTCERR2(SetVADStatus, channel, true);
-        return false;
-      }
-    }
-  }
-  return true;
-}
-
-bool WebRtcVoiceMediaChannel::SetSendCodec(
-    int channel, const webrtc::CodecInst& send_codec) {
-  LOG(LS_INFO) << "Send channel " << channel <<  " selected voice codec "
-               << ToString(send_codec) << ", bitrate=" << send_codec.rate;
-
-  webrtc::CodecInst current_codec = {0};
-  if (engine()->voe()->codec()->GetSendCodec(channel, current_codec) == 0 &&
-      (send_codec == current_codec)) {
-    // Codec is already configured, we can return without setting it again.
-    return true;
-  }
-
-  if (engine()->voe()->codec()->SetSendCodec(channel, send_codec) == -1) {
-    LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec));
-    return false;
-  }
-  return true;
-}
-
 void WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
   desired_playout_ = playout;
   return ChangePlayout(desired_playout_);
 }
 
 void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (playout_ == playout) {
     return;
@@ skipping 89 matching lines @@
     return false;
   }
 
   // Save the channel to send_streams_, so that RemoveSendStream() can still
   // delete the channel in case failure happens below.
   webrtc::AudioTransport* audio_transport =
       engine()->voe()->base()->audio_transport();
 
   WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
       channel, audio_transport, ssrc, sp.cname, send_codec_spec_,
-      send_rtp_extensions_, call_, this);
+      send_rtp_extensions_, max_send_bitrate_bps_, call_, this);
   send_streams_.insert(std::make_pair(ssrc, stream));
 
-  // Set the current codecs to be used for the new channel. We need to do this
-  // after adding the channel to send_channels_, because of how max bitrate is
-  // currently being configured by SetSendCodec().
-  if (HasSendCodec() && !SetSendCodecs(channel, stream->rtp_parameters())) {
-    RemoveSendStream(ssrc);
-    return false;
-  }
-
   // At this point the stream's local SSRC has been updated. If it is the first
   // send stream, make sure that all the receive streams are updated with the
   // same SSRC in order to send receiver reports.
   if (send_streams_.size() == 1) {
     receiver_reports_ssrc_ = ssrc;
     for (const auto& kv : recv_streams_) {
       // TODO(solenberg): Allow applications to set the RTCP SSRC of receive
       // streams instead, so we can avoid recreating the streams here.
       kv.second->RecreateAudioReceiveStream(ssrc);
       int recv_channel = kv.second->channel();
@@ skipping 353 matching lines @@
   webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
   if (ap) {
     ap->set_output_will_be_muted(all_muted);
   }
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) {
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBitrate.";
   max_send_bitrate_bps_ = bps;
-
+  bool success = true;

[minyue-webrtc, 2016/10/14 13:32:56]

this is now possible to return false as the old code. But different from the old
one, we do not break when on the first failure, which is poor behavior IMO.

   for (const auto& kv : send_streams_) {
-    if (!SetChannelSendParameters(kv.second->channel(),
-                                  kv.second->rtp_parameters())) {
-      return false;
+    if (!kv.second->SetMaxSendBitrate(max_send_bitrate_bps_)) {
+      success = false;
     }
   }
-  return true;
-}
-
-bool WebRtcVoiceMediaChannel::SetChannelSendParameters(
-    int channel,
-    const webrtc::RtpParameters& parameters) {
-  RTC_CHECK_EQ(1UL, parameters.encodings.size());
-  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
-  // different order (which should change the send codec).
-  return SetMaxSendBitrate(
-      channel, MinPositive(max_send_bitrate_bps_,
-                           parameters.encodings[0].max_bitrate_bps));
-}
-
-bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int channel, int 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 (!HasSendCodec()) {
-    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 = send_codec_spec_.codec_inst;
-  bool is_multi_rate = WebRtcVoiceCodecs::IsCodecMultiRate(codec);
-
-  if (is_multi_rate) {
-    // If codec is multi-rate then just set the bitrate.
-    int max_bitrate_bps = WebRtcVoiceCodecs::MaxBitrateBps(codec);
-    codec.rate = std::min(bps, max_bitrate_bps);
-    LOG(LS_INFO) << "Setting codec " << codec.plname << " to bitrate " << bps
-                 << " bps.";
-    if (!SetSendCodec(channel, 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;
-  }
+  return success;
 }
 
 void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
   call_->SignalChannelNetworkState(
       webrtc::MediaType::AUDIO,
       ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
@@ skipping 105 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE