webrtc_m130/webrtc/test/encoder_settings.cc
hbos bab934bffe H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
It works on all platforms except Android and iOS (FFmpeg limitation).

Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.

Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)

Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)

NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424

Review URL: https://codereview.webrtc.org/1306813009

Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 09:36:07 +00:00

74 lines
2.5 KiB
C++

/*
* Copyright (c) 2014 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/test/encoder_settings.h"
#include <assert.h>
#include <string.h>
#include "webrtc/test/fake_decoder.h"
#include "webrtc/video_decoder.h"
namespace webrtc {
namespace test {
std::vector<VideoStream> CreateVideoStreams(size_t num_streams) {
assert(num_streams > 0);
// Add more streams to the settings above with reasonable values if required.
static const size_t kNumSettings = 3;
assert(num_streams <= kNumSettings);
std::vector<VideoStream> stream_settings(kNumSettings);
stream_settings[0].width = 320;
stream_settings[0].height = 180;
stream_settings[0].max_framerate = 30;
stream_settings[0].min_bitrate_bps = 50000;
stream_settings[0].target_bitrate_bps = stream_settings[0].max_bitrate_bps =
150000;
stream_settings[0].max_qp = 56;
stream_settings[1].width = 640;
stream_settings[1].height = 360;
stream_settings[1].max_framerate = 30;
stream_settings[1].min_bitrate_bps = 200000;
stream_settings[1].target_bitrate_bps = stream_settings[1].max_bitrate_bps =
450000;
stream_settings[1].max_qp = 56;
stream_settings[2].width = 1280;
stream_settings[2].height = 720;
stream_settings[2].max_framerate = 30;
stream_settings[2].min_bitrate_bps = 700000;
stream_settings[2].target_bitrate_bps = stream_settings[2].max_bitrate_bps =
1500000;
stream_settings[2].max_qp = 56;
stream_settings.resize(num_streams);
return stream_settings;
}
VideoReceiveStream::Decoder CreateMatchingDecoder(
const VideoSendStream::Config::EncoderSettings& encoder_settings) {
VideoReceiveStream::Decoder decoder;
decoder.payload_type = encoder_settings.payload_type;
decoder.payload_name = encoder_settings.payload_name;
if (encoder_settings.payload_name == "H264") {
decoder.decoder = VideoDecoder::Create(VideoDecoder::kH264);
} else if (encoder_settings.payload_name == "VP8") {
decoder.decoder = VideoDecoder::Create(VideoDecoder::kVp8);
} else if (encoder_settings.payload_name == "VP9") {
decoder.decoder = VideoDecoder::Create(VideoDecoder::kVp9);
} else {
decoder.decoder = new FakeDecoder();
}
return decoder;
}
} // namespace test
} // namespace webrtc