Extracted hardcoded constants from cast_rtp_stream.

chrome/renderer/media/cast_rtp_stream.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "chrome/renderer/media/cast_rtp_stream.h"
 
 #include <algorithm>
 
 #include "base/bind.h"
 #include "base/callback_helpers.h"
@@ skipping 163 matching lines @@
           params.payload.max_latency_ms);
   config->max_playout_delay =
       base::TimeDelta::FromMilliseconds(params.payload.max_latency_ms);
   if (config->min_playout_delay <= base::TimeDelta())
     return false;
   if (config->min_playout_delay > config->max_playout_delay)
     return false;
   config->rtp_payload_type = params.payload.payload_type;
   config->use_external_encoder = false;
   config->frequency = params.payload.clock_rate;
-  if (config->frequency < 8000)
+  if (config->frequency < media::cast::kMinUnackedFrequency)

[miu, 2015/09/14 17:55:07]

8000 is the largest of the minimum sampling rate required by all codecs (e.g.,
Opus).  It has nothing to do with unacked frames.

[msu.koo, 2015/09/15 00:05:06]

Changed to use media::limits::kMinSampleRate instead.
And if the minimum audio sample rate is 8000 for all codecs,
I think we can also change the value of kMinSamleRate to 8000.(currently 3000)

[miu, 2015/09/15 04:16:57]

There's a difference: 3000 is an operational lower-bound for the entire audio
stack.  However, 8000 is the lower-bound for the audio codecs that process the
audio here.  So, this constant really does need to be 8000.

Can you add a kMinSampleRateForEncoding constant (set to 8000) here:
https://code.google.com/p/chromium/codesearch#chromium/src/media/cast/cast_de...

     return false;
   config->channels = params.payload.channels;
   if (config->channels < 1)
     return false;
   config->bitrate = params.payload.max_bitrate * kBitrateMultiplier;
   if (params.payload.codec_name == kCodecNameOpus)
     config->codec = media::cast::CODEC_AUDIO_OPUS;
   else
     return false;
   config->aes_key = params.payload.aes_key;
@@ skipping 20 matching lines @@
     return false;
   config->rtp_payload_type = params.payload.payload_type;
   config->min_bitrate = config->start_bitrate =
       params.payload.min_bitrate * kBitrateMultiplier;
   config->max_bitrate = params.payload.max_bitrate * kBitrateMultiplier;
   if (config->min_bitrate > config->max_bitrate)
     return false;
   config->start_bitrate = config->min_bitrate;
   config->max_frame_rate = static_cast<int>(
       std::max(1.0, params.payload.max_frame_rate) + 0.5);
-  if (config->max_frame_rate > 120)
+  if (config->max_frame_rate > media::cast::kMaxUnackedFrames)

[miu, 2015/09/14 17:55:07]

120 was chosen arbitrarily in the past.  Instead, I think
media::limits::kMaxFramesPerSecond is more appropriate.  You'll need to #include
"media/base/limits.h"

[msu.koo, 2015/09/15 00:05:06]

Done.

And IMHO, kMaxFramesPerSecond 1000 is too large for the MaxFPS. It was
originally introduced for video capture implementations, so I think 120 is also
OK for this value.

     return false;
   if (params.payload.codec_name == kCodecNameVp8) {
     config->use_external_encoder = IsHardwareVP8EncodingSupported();
     config->codec = media::cast::CODEC_VIDEO_VP8;
   } else if (params.payload.codec_name == kCodecNameH264) {
     config->use_external_encoder = IsHardwareH264EncodingSupported();
     config->codec = media::cast::CODEC_VIDEO_H264;
   } else {
     return false;
   }
@@ skipping 339 matching lines @@
   DCHECK(content::RenderThread::Get());
   DVLOG(1) << "CastRtpStream::DidEncounterError(" << message << ") = "
            << (IsAudio() ? "audio" : "video");
   // Save the WeakPtr first because the error callback might delete this object.
   base::WeakPtr<CastRtpStream> ptr = weak_factory_.GetWeakPtr();
   error_callback_.Run(message);
   base::ThreadTaskRunnerHandle::Get()->PostTask(
       FROM_HERE,
       base::Bind(&CastRtpStream::Stop, ptr));
 }