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Bug 1253499 - Break out VideoStreamFactory to separate file. r=dminor 

Differential Revision: https://phabricator.services.mozilla.com/D6263

--HG--
extra : moz-landing-system : lando
This commit is contained in:
Andreas Pehrson 2018-09-19 15:51:46 +00:00
Родитель 6f26c10cda
Коммит ac369ba83b
5 изменённых файлов: 349 добавлений и 298 удалений
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230
media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
Просмотреть файл

@ -8,6 +8,7 @@
#include "AudioConduit.h"
#include "VideoConduit.h"
#include "VideoStreamFactory.h"
#include "YuvStamper.h"
#include "mozilla/TemplateLib.h"
#include "mozilla/media/MediaUtils.h"
@ -88,12 +89,8 @@ static const char* kRedPayloadName = "red";
// is active and one or more layers are being scaled.
#define SIMULCAST_RESOLUTION_ALIGNMENT 16
// Convert (SI) kilobits/sec to (SI) bits/sec
#define KBPS(kbps) kbps * 1000
// 32 bytes is what WebRTC CodecInst expects
const unsigned int WebrtcVideoConduit::CODEC_PLNAME_SIZE = 32;
static const int kViEMinCodecBitrate_bps = KBPS(30);
template<typename T>
T MinIgnoreZero(const T& a, const T& b)
@ -173,100 +170,6 @@ SelectSendFrameRate(const VideoCodecConfig* codecConfig,
return new_framerate;
}
#define MB_OF(w,h) ((unsigned int)((((w+15)>>4))*((unsigned int)((h+15)>>4))))
// For now, try to set the max rates well above the knee in the curve.
// Chosen somewhat arbitrarily; it's hard to find good data oriented for
// realtime interactive/talking-head recording. These rates assume
// 30fps.
// XXX Populate this based on a pref (which we should consider sorting because
// people won't assume they need to).
static WebrtcVideoConduit::ResolutionAndBitrateLimits kResolutionAndBitrateLimits[] = {
{MB_OF(1920, 1200), KBPS(1500), KBPS(2000), KBPS(10000)}, // >HD (3K, 4K, etc)
{MB_OF(1280, 720), KBPS(1200), KBPS(1500), KBPS(5000)}, // HD ~1080-1200
{MB_OF(800, 480), KBPS(600), KBPS(800), KBPS(2500)}, // HD ~720
{MB_OF(480, 270), KBPS(150), KBPS(500), KBPS(2000)}, // WVGA
{tl::Max<MB_OF(400, 240), MB_OF(352, 288)>::value, KBPS(125), KBPS(300), KBPS(1300)}, // VGA
{MB_OF(176, 144), KBPS(100), KBPS(150), KBPS(500)}, // WQVGA, CIF
{0 , KBPS(40), KBPS(80), KBPS(250)} // QCIF and below
};
static WebrtcVideoConduit::ResolutionAndBitrateLimits
GetLimitsFor(unsigned int aWidth, unsigned int aHeight, int aCapBps = 0)
{
// max bandwidth should be proportional (not linearly!) to resolution, and
// proportional (perhaps linearly, or close) to current frame rate.
int fs = MB_OF(aWidth, aHeight);
for (const auto& resAndLimits : kResolutionAndBitrateLimits) {
if (fs > resAndLimits.resolution_in_mb &&
// pick the highest range where at least start rate is within cap
// (or if we're at the end of the array).
(aCapBps == 0 ||
resAndLimits.start_bitrate_bps <= aCapBps ||
resAndLimits.resolution_in_mb == 0)) {
return resAndLimits;
}
}
MOZ_CRASH("Loop should have handled fallback");
}
/**
* Function to set the encoding bitrate limits based on incoming frame size and rate
* @param width, height: dimensions of the frame
* @param min: minimum bitrate in bps
* @param start: bitrate in bps that the encoder should start with
* @param cap: user-enforced max bitrate, or 0
* @param pref_cap: cap enforced by prefs
* @param negotiated_cap: cap negotiated through SDP
* @param aVideoStream stream to apply bitrates to
*/
static void
SelectBitrates(
unsigned short width, unsigned short height,
int min, int start,
int cap, int pref_cap, int negotiated_cap,
webrtc::VideoStream& aVideoStream)
{
int& out_min = aVideoStream.min_bitrate_bps;
int& out_start = aVideoStream.target_bitrate_bps;
int& out_max = aVideoStream.max_bitrate_bps;
WebrtcVideoConduit::ResolutionAndBitrateLimits resAndLimits =
GetLimitsFor(width, height);
out_min = MinIgnoreZero(resAndLimits.min_bitrate_bps, cap);
out_start = MinIgnoreZero(resAndLimits.start_bitrate_bps, cap);
out_max = MinIgnoreZero(resAndLimits.max_bitrate_bps, cap);
// Note: negotiated_cap is the max transport bitrate - it applies to
// a single codec encoding, but should also apply to the sum of all
// simulcast layers in this encoding! So sum(layers.maxBitrate) <=
// negotiated_cap
// Note that out_max already has had pref_cap applied to it
out_max = MinIgnoreZero(negotiated_cap, out_max);
out_min = std::min(out_min, out_max);
out_start = std::min(out_start, out_max);
if (min && min > out_min) {
out_min = min;
}
// If we try to set a minimum bitrate that is too low, ViE will reject it.
out_min = std::max(kViEMinCodecBitrate_bps, out_min);
out_max = std::max(kViEMinCodecBitrate_bps, out_max);
if (start && start > out_start) {
out_start = start;
}
// Ensure that min <= start <= max
if (out_min > out_max) {
out_min = out_max;
}
out_start = std::min(out_max, std::max(out_start, out_min));
MOZ_ASSERT(pref_cap == 0 || out_max <= pref_cap);
}
/**
* Perform validation on the codecConfig to be applied
*/
@ -898,135 +801,6 @@ CodecsDifferent(const nsTArray<UniquePtr<VideoCodecConfig>>& a,
return false;
}
std::vector<webrtc::VideoStream>
WebrtcVideoConduit::VideoStreamFactory::CreateEncoderStreams(
int width, int height, const webrtc::VideoEncoderConfig& config)
{
size_t streamCount = config.number_of_streams;
// We only allow one layer when screensharing
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
streamCount = 1;
}
std::vector<webrtc::VideoStream> streams;
streams.reserve(streamCount);
// We assume that the first stream is the full-resolution stream.
// This ensures all simulcast layers will be of the same aspect ratio as the input.
mSimulcastAdapter->OnOutputFormatRequest(
cricket::VideoFormat(width, height, 0, 0));
for (size_t idx = streamCount - 1; streamCount > 0; idx--, streamCount--) {
webrtc::VideoStream video_stream;
auto& simulcastEncoding = mCodecConfig.mSimulcastEncodings[idx];
MOZ_ASSERT(simulcastEncoding.constraints.scaleDownBy >= 1.0);
// All streams' dimensions must retain the aspect ratio of the input stream.
// Note that the first stream might already have been scaled by us.
// Webrtc.org doesn't know this, so we have to adjust lower layers manually.
int unusedCropWidth, unusedCropHeight, outWidth, outHeight;
if (idx == 0) {
// This is the highest-resolution stream. We avoid calling
// AdaptFrameResolution on this because precision errors in VideoAdapter
// can cause the out-resolution to be an odd pixel smaller than the
// source (1920x1419 has caused this). We shortcut this instead.
outWidth = width;
outHeight = height;
} else {
float effectiveScaleDownBy =
simulcastEncoding.constraints.scaleDownBy /
mCodecConfig.mSimulcastEncodings[0].constraints.scaleDownBy;
MOZ_ASSERT(effectiveScaleDownBy >= 1.0);
mSimulcastAdapter->OnScaleResolutionBy(
effectiveScaleDownBy > 1.0 ?
rtc::Optional<float>(effectiveScaleDownBy) :
rtc::Optional<float>());
bool rv = mSimulcastAdapter->AdaptFrameResolution(
width,
height,
0, // Ok, since we don't request an output format with an interval
&unusedCropWidth,
&unusedCropHeight,
&outWidth,
&outHeight);
if (!rv) {
// The only thing that can make AdaptFrameResolution fail in this case
// is if this layer is scaled so far down that it has less than one pixel.
outWidth = 0;
outHeight = 0;
}
}
if (outWidth == 0 || outHeight == 0) {
CSFLogInfo(LOGTAG,
"%s Stream with RID %s ignored because of no resolution.",
__FUNCTION__, simulcastEncoding.rid.c_str());
continue;
}
MOZ_ASSERT(outWidth > 0);
MOZ_ASSERT(outHeight > 0);
video_stream.width = outWidth;
video_stream.height = outHeight;
CSFLogInfo(LOGTAG, "%s Input frame %ux%u, RID %s scaling to %zux%zu",
__FUNCTION__, width, height, simulcastEncoding.rid.c_str(),
video_stream.width, video_stream.height);
if (video_stream.width * height != width * video_stream.height) {
CSFLogInfo(LOGTAG,
"%s Stream with RID %s ignored because of bad aspect ratio.",
__FUNCTION__, simulcastEncoding.rid.c_str());
continue;
}
// We want to ensure this picks up the current framerate, so indirect
video_stream.max_framerate = mSendingFramerate;
SelectBitrates(
video_stream.width, video_stream.height,
mMinBitrate, mStartBitrate, simulcastEncoding.constraints.maxBr,
mPrefMaxBitrate, mNegotiatedMaxBitrate, video_stream);
video_stream.max_qp = kQpMax;
video_stream.SetRid(simulcastEncoding.rid);
// leave vector temporal_layer_thresholds_bps empty for non-simulcast
video_stream.temporal_layer_thresholds_bps.clear();
if (config.number_of_streams > 1) {
// XXX Note: in simulcast.cc in upstream code, the array value is
// 3(-1) for all streams, though it's in an array, except for screencasts,
// which use 1 (i.e 2 layers).
// Oddly, though this is a 'bps' array, nothing really looks at the
// values for normal video, just the size of the array to know the
// number of temporal layers.
// For VideoEncoderConfig::ContentType::kScreen, though, in
// video_codec_initializer.cc it uses [0] to set the target bitrate
// for the screenshare.
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
video_stream.temporal_layer_thresholds_bps.push_back(video_stream.target_bitrate_bps);
} else {
video_stream.temporal_layer_thresholds_bps.resize(2);
}
// XXX Bug 1390215 investigate using more of
// simulcast.cc:GetSimulcastConfig() or our own algorithm to replace it
}
if (mCodecConfig.mName == "H264") {
if (mCodecConfig.mEncodingConstraints.maxMbps > 0) {
// Not supported yet!
CSFLogError(LOGTAG, "%s H.264 max_mbps not supported yet", __FUNCTION__);
}
}
streams.push_back(video_stream);
}
return streams;
}
/**
* Note: Setting the send-codec on the Video Engine will restart the encoder,
* sets up new SSRC and reset RTP_RTCP module with the new codec setting.
@ -2654,7 +2428,7 @@ WebrtcVideoConduit::VideoEncoderConfigBuilder::SetEncoderSpecificSettings(
}
void
WebrtcVideoConduit::VideoEncoderConfigBuilder::SetVideoStreamFactory(rtc::scoped_refptr<WebrtcVideoConduit::VideoStreamFactory> aFactory)
WebrtcVideoConduit::VideoEncoderConfigBuilder::SetVideoStreamFactory(rtc::scoped_refptr<VideoStreamFactory> aFactory)
{
MOZ_ASSERT(NS_IsMainThread());

79
media/webrtc/signaling/src/media-conduit/VideoConduit.h
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@ -40,9 +40,17 @@
namespace mozilla {
// Convert (SI) kilobits/sec to (SI) bits/sec
#define KBPS(kbps) kbps * 1000
const int kViEMinCodecBitrate_bps = KBPS(30);
const unsigned int kVideoMtu = 1200;
const int kQpMax = 56;
template<typename T>
T MinIgnoreZero(const T& a, const T& b);
class VideoStreamFactory;
class WebrtcAudioConduit;
class nsThread;
@ -68,14 +76,6 @@ class WebrtcVideoConduit : public VideoSessionConduit
, public rtc::VideoSourceInterface<webrtc::VideoFrame>
{
public:
struct ResolutionAndBitrateLimits
{
int resolution_in_mb;
int min_bitrate_bps;
int start_bitrate_bps;
int max_bitrate_bps;
};
//VoiceEngine defined constant for Payload Name Size.
static const unsigned int CODEC_PLNAME_SIZE;
@ -421,8 +421,6 @@ private:
* Stores encoder configuration information and produces
* a VideoEncoderConfig from it.
*/
class VideoStreamFactory;
class VideoEncoderConfigBuilder {
public:
/**
@ -434,7 +432,7 @@ private:
double jsScaleDownBy=1.0; // user-controlled downscale
};
void SetEncoderSpecificSettings(rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings> aSettings);
void SetVideoStreamFactory(rtc::scoped_refptr<WebrtcVideoConduit::VideoStreamFactory> aFactory);
void SetVideoStreamFactory(rtc::scoped_refptr<VideoStreamFactory> aFactory);
void SetMinTransmitBitrateBps(int aXmitMinBps);
void SetContentType(webrtc::VideoEncoderConfig::ContentType aContentType);
void SetResolutionDivisor(unsigned char aDivisor);
@ -456,65 +454,6 @@ private:
// Utility function to dump recv codec database
void DumpCodecDB() const;
// Factory class for VideoStreams... vie_encoder.cc will call this to reconfigure.
class VideoStreamFactory : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface
{
public:
VideoStreamFactory(VideoCodecConfig aConfig,
webrtc::VideoCodecMode aCodecMode,
int aMinBitrate, int aStartBitrate,
int aPrefMaxBitrate, int aNegotiatedMaxBitrate,
unsigned int aSendingFramerate)
: mCodecMode(aCodecMode)
, mSendingFramerate(aSendingFramerate)
, mCodecConfig(std::forward<VideoCodecConfig>(aConfig))
, mMinBitrate(aMinBitrate)
, mStartBitrate(aStartBitrate)
, mPrefMaxBitrate(aPrefMaxBitrate)
, mNegotiatedMaxBitrate(aNegotiatedMaxBitrate)
, mSimulcastAdapter(MakeUnique<cricket::VideoAdapter>())
{}
void SetCodecMode(webrtc::VideoCodecMode aCodecMode)
{
MOZ_ASSERT(NS_IsMainThread());
mCodecMode = aCodecMode;
}
void SetSendingFramerate(unsigned int aSendingFramerate)
{
MOZ_ASSERT(NS_IsMainThread());
mSendingFramerate = aSendingFramerate;
}
private:
// This gets called off-main thread and may hold internal webrtc.org
// locks. May *NOT* lock the conduit's mutex, to avoid deadlocks.
std::vector<webrtc::VideoStream>
CreateEncoderStreams(int width, int height,
const webrtc::VideoEncoderConfig& config) override;
// Used to limit number of streams for screensharing.
Atomic<webrtc::VideoCodecMode> mCodecMode;
// The framerate we're currently sending at.
Atomic<unsigned int> mSendingFramerate;
// The current send codec config, containing simulcast layer configs.
const VideoCodecConfig mCodecConfig;
// Bitrate limits in bps.
const int mMinBitrate = 0;
const int mStartBitrate = 0;
const int mPrefMaxBitrate = 0;
const int mNegotiatedMaxBitrate = 0;
// Adapter for simulcast layers. We use this to handle scaleResolutionDownBy
// for layers. It's separate from the conduit's mVideoAdapter to not affect
// scaling settings for incoming frames.
UniquePtr<cricket::VideoAdapter> mSimulcastAdapter;
};
// Video Latency Test averaging filter
void VideoLatencyUpdate(uint64_t new_sample);

258
media/webrtc/signaling/src/media-conduit/VideoStreamFactory.cpp Normal file
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@ -0,0 +1,258 @@
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at https://mozilla.org/MPL/2.0/. */
#include "VideoStreamFactory.h"
#include "CSFLog.h"
#include "nsThreadUtils.h"
#include "VideoConduit.h"
namespace mozilla {
#ifdef LOGTAG
#undef LOGTAG
#endif
#define LOGTAG "WebrtcVideoSessionConduit"
#define MB_OF(w,h) ((unsigned int)((((w+15)>>4))*((unsigned int)((h+15)>>4))))
// For now, try to set the max rates well above the knee in the curve.
// Chosen somewhat arbitrarily; it's hard to find good data oriented for
// realtime interactive/talking-head recording. These rates assume
// 30fps.
// XXX Populate this based on a pref (which we should consider sorting because
// people won't assume they need to).
static VideoStreamFactory::ResolutionAndBitrateLimits kResolutionAndBitrateLimits[] = {
{MB_OF(1920, 1200), KBPS(1500), KBPS(2000), KBPS(10000)}, // >HD (3K, 4K, etc)
{MB_OF(1280, 720), KBPS(1200), KBPS(1500), KBPS(5000)}, // HD ~1080-1200
{MB_OF(800, 480), KBPS(600), KBPS(800), KBPS(2500)}, // HD ~720
{MB_OF(480, 270), KBPS(150), KBPS(500), KBPS(2000)}, // WVGA
{tl::Max<MB_OF(400, 240), MB_OF(352, 288)>::value, KBPS(125), KBPS(300), KBPS(1300)}, // VGA
{MB_OF(176, 144), KBPS(100), KBPS(150), KBPS(500)}, // WQVGA, CIF
{0 , KBPS(40), KBPS(80), KBPS(250)} // QCIF and below
};
static VideoStreamFactory::ResolutionAndBitrateLimits
GetLimitsFor(unsigned int aWidth, unsigned int aHeight, int aCapBps = 0)
{
// max bandwidth should be proportional (not linearly!) to resolution, and
// proportional (perhaps linearly, or close) to current frame rate.
int fs = MB_OF(aWidth, aHeight);
for (const auto& resAndLimits : kResolutionAndBitrateLimits) {
if (fs > resAndLimits.resolution_in_mb &&
// pick the highest range where at least start rate is within cap
// (or if we're at the end of the array).
(aCapBps == 0 ||
resAndLimits.start_bitrate_bps <= aCapBps ||
resAndLimits.resolution_in_mb == 0)) {
return resAndLimits;
}
}
MOZ_CRASH("Loop should have handled fallback");
}
/**
* Function to set the encoding bitrate limits based on incoming frame size and rate
* @param width, height: dimensions of the frame
* @param min: minimum bitrate in bps
* @param start: bitrate in bps that the encoder should start with
* @param cap: user-enforced max bitrate, or 0
* @param pref_cap: cap enforced by prefs
* @param negotiated_cap: cap negotiated through SDP
* @param aVideoStream stream to apply bitrates to
*/
static void
SelectBitrates(
unsigned short width, unsigned short height,
int min, int start,
int cap, int pref_cap, int negotiated_cap,
webrtc::VideoStream& aVideoStream)
{
int& out_min = aVideoStream.min_bitrate_bps;
int& out_start = aVideoStream.target_bitrate_bps;
int& out_max = aVideoStream.max_bitrate_bps;
VideoStreamFactory::ResolutionAndBitrateLimits resAndLimits =
GetLimitsFor(width, height);
out_min = MinIgnoreZero(resAndLimits.min_bitrate_bps, cap);
out_start = MinIgnoreZero(resAndLimits.start_bitrate_bps, cap);
out_max = MinIgnoreZero(resAndLimits.max_bitrate_bps, cap);
// Note: negotiated_cap is the max transport bitrate - it applies to
// a single codec encoding, but should also apply to the sum of all
// simulcast layers in this encoding! So sum(layers.maxBitrate) <=
// negotiated_cap
// Note that out_max already has had pref_cap applied to it
out_max = MinIgnoreZero(negotiated_cap, out_max);
out_min = std::min(out_min, out_max);
out_start = std::min(out_start, out_max);
if (min && min > out_min) {
out_min = min;
}
// If we try to set a minimum bitrate that is too low, ViE will reject it.
out_min = std::max(kViEMinCodecBitrate_bps, out_min);
out_max = std::max(kViEMinCodecBitrate_bps, out_max);
if (start && start > out_start) {
out_start = start;
}
// Ensure that min <= start <= max
if (out_min > out_max) {
out_min = out_max;
}
out_start = std::min(out_max, std::max(out_start, out_min));
MOZ_ASSERT(pref_cap == 0 || out_max <= pref_cap);
}
void
VideoStreamFactory::SetCodecMode(webrtc::VideoCodecMode aCodecMode)
{
MOZ_ASSERT(NS_IsMainThread());
mCodecMode = aCodecMode;
}
void
VideoStreamFactory::SetSendingFramerate(unsigned int aSendingFramerate)
{
MOZ_ASSERT(NS_IsMainThread());
mSendingFramerate = aSendingFramerate;
}
std::vector<webrtc::VideoStream>
VideoStreamFactory::CreateEncoderStreams(
int width, int height, const webrtc::VideoEncoderConfig& config)
{
size_t streamCount = config.number_of_streams;
// We only allow one layer when screensharing
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
streamCount = 1;
}
std::vector<webrtc::VideoStream> streams;
streams.reserve(streamCount);
// We assume that the first stream is the full-resolution stream.
// This ensures all simulcast layers will be of the same aspect ratio as the input.
mSimulcastAdapter->OnOutputFormatRequest(
cricket::VideoFormat(width, height, 0, 0));
for (size_t idx = streamCount - 1; streamCount > 0; idx--, streamCount--) {
webrtc::VideoStream video_stream;
auto& simulcastEncoding = mCodecConfig.mSimulcastEncodings[idx];
MOZ_ASSERT(simulcastEncoding.constraints.scaleDownBy >= 1.0);
// All streams' dimensions must retain the aspect ratio of the input stream.
// Note that the first stream might already have been scaled by us.
// Webrtc.org doesn't know this, so we have to adjust lower layers manually.
int unusedCropWidth, unusedCropHeight, outWidth, outHeight;
if (idx == 0) {
// This is the highest-resolution stream. We avoid calling
// AdaptFrameResolution on this because precision errors in VideoAdapter
// can cause the out-resolution to be an odd pixel smaller than the
// source (1920x1419 has caused this). We shortcut this instead.
outWidth = width;
outHeight = height;
} else {
float effectiveScaleDownBy =
simulcastEncoding.constraints.scaleDownBy /
mCodecConfig.mSimulcastEncodings[0].constraints.scaleDownBy;
MOZ_ASSERT(effectiveScaleDownBy >= 1.0);
mSimulcastAdapter->OnScaleResolutionBy(
effectiveScaleDownBy > 1.0 ?
rtc::Optional<float>(effectiveScaleDownBy) :
rtc::Optional<float>());
bool rv = mSimulcastAdapter->AdaptFrameResolution(
width,
height,
0, // Ok, since we don't request an output format with an interval
&unusedCropWidth,
&unusedCropHeight,
&outWidth,
&outHeight);
if (!rv) {
// The only thing that can make AdaptFrameResolution fail in this case
// is if this layer is scaled so far down that it has less than one pixel.
outWidth = 0;
outHeight = 0;
}
}
if (outWidth == 0 || outHeight == 0) {
CSFLogInfo(LOGTAG,
"%s Stream with RID %s ignored because of no resolution.",
__FUNCTION__, simulcastEncoding.rid.c_str());
continue;
}
MOZ_ASSERT(outWidth > 0);
MOZ_ASSERT(outHeight > 0);
video_stream.width = outWidth;
video_stream.height = outHeight;
CSFLogInfo(LOGTAG, "%s Input frame %ux%u, RID %s scaling to %zux%zu",
__FUNCTION__, width, height, simulcastEncoding.rid.c_str(),
video_stream.width, video_stream.height);
if (video_stream.width * height != width * video_stream.height) {
CSFLogInfo(LOGTAG,
"%s Stream with RID %s ignored because of bad aspect ratio.",
__FUNCTION__, simulcastEncoding.rid.c_str());
continue;
}
// We want to ensure this picks up the current framerate, so indirect
video_stream.max_framerate = mSendingFramerate;
SelectBitrates(
video_stream.width, video_stream.height,
mMinBitrate, mStartBitrate, simulcastEncoding.constraints.maxBr,
mPrefMaxBitrate, mNegotiatedMaxBitrate, video_stream);
video_stream.max_qp = kQpMax;
video_stream.SetRid(simulcastEncoding.rid);
// leave vector temporal_layer_thresholds_bps empty for non-simulcast
video_stream.temporal_layer_thresholds_bps.clear();
if (config.number_of_streams > 1) {
// XXX Note: in simulcast.cc in upstream code, the array value is
// 3(-1) for all streams, though it's in an array, except for screencasts,
// which use 1 (i.e 2 layers).
// Oddly, though this is a 'bps' array, nothing really looks at the
// values for normal video, just the size of the array to know the
// number of temporal layers.
// For VideoEncoderConfig::ContentType::kScreen, though, in
// video_codec_initializer.cc it uses [0] to set the target bitrate
// for the screenshare.
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
video_stream.temporal_layer_thresholds_bps.push_back(video_stream.target_bitrate_bps);
} else {
video_stream.temporal_layer_thresholds_bps.resize(2);
}
// XXX Bug 1390215 investigate using more of
// simulcast.cc:GetSimulcastConfig() or our own algorithm to replace it
}
if (mCodecConfig.mName == "H264") {
if (mCodecConfig.mEncodingConstraints.maxMbps > 0) {
// Not supported yet!
CSFLogError(LOGTAG, "%s H.264 max_mbps not supported yet", __FUNCTION__);
}
}
streams.push_back(video_stream);
}
return streams;
}
} // namespace mozilla

79
media/webrtc/signaling/src/media-conduit/VideoStreamFactory.h Normal file
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@ -0,0 +1,79 @@
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at https://mozilla.org/MPL/2.0/. */
#ifndef VideoStreamFactory_h
#define VideoStreamFactory_h
#include "CodecConfig.h"
#include "mozilla/Atomics.h"
#include "mozilla/UniquePtr.h"
#include "webrtc/config.h"
#include "webrtc/media/base/videoadapter.h"
namespace mozilla {
// Factory class for VideoStreams... vie_encoder.cc will call this to reconfigure.
class VideoStreamFactory : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface
{
public:
struct ResolutionAndBitrateLimits
{
int resolution_in_mb;
int min_bitrate_bps;
int start_bitrate_bps;
int max_bitrate_bps;
};
VideoStreamFactory(VideoCodecConfig aConfig,
webrtc::VideoCodecMode aCodecMode,
int aMinBitrate, int aStartBitrate,
int aPrefMaxBitrate, int aNegotiatedMaxBitrate,
unsigned int aSendingFramerate)
: mCodecMode(aCodecMode)
, mSendingFramerate(aSendingFramerate)
, mCodecConfig(std::forward<VideoCodecConfig>(aConfig))
, mMinBitrate(aMinBitrate)
, mStartBitrate(aStartBitrate)
, mPrefMaxBitrate(aPrefMaxBitrate)
, mNegotiatedMaxBitrate(aNegotiatedMaxBitrate)
, mSimulcastAdapter(MakeUnique<cricket::VideoAdapter>())
{}
void SetCodecMode(webrtc::VideoCodecMode aCodecMode);
void SetSendingFramerate(unsigned int aSendingFramerate);
// This gets called off-main thread and may hold internal webrtc.org
// locks. May *NOT* lock the conduit's mutex, to avoid deadlocks.
std::vector<webrtc::VideoStream>
CreateEncoderStreams(int width, int height,
const webrtc::VideoEncoderConfig& config) override;
private:
// Used to limit number of streams for screensharing.
Atomic<webrtc::VideoCodecMode> mCodecMode;
// The framerate we're currently sending at.
Atomic<unsigned int> mSendingFramerate;
// The current send codec config, containing simulcast layer configs.
const VideoCodecConfig mCodecConfig;
// Bitrate limits in bps.
const int mMinBitrate = 0;
const int mStartBitrate = 0;
const int mPrefMaxBitrate = 0;
const int mNegotiatedMaxBitrate = 0;
// Adapter for simulcast layers. We use this to handle scaleResolutionDownBy
// for layers. It's separate from the conduit's mVideoAdapter to not affect
// scaling settings for incoming frames.
UniquePtr<cricket::VideoAdapter> mSimulcastAdapter;
};
} // namespace mozilla
#endif

1
media/webrtc/signaling/src/media-conduit/moz.build
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@ -24,6 +24,7 @@ UNIFIED_SOURCES += [
'MediaDataDecoderCodec.cpp',
'RtpSourceObserver.cpp',
'VideoConduit.cpp',
'VideoStreamFactory.cpp',
'WebrtcGmpVideoCodec.cpp',
'WebrtcMediaDataDecoderCodec.cpp',
]