709ed67c38
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE). BUG=webrtc:4690 R=pbos@webrtc.org, pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1269863005 . Cr-Commit-Position: refs/heads/master@{#9939}
3635 lines
123 KiB
C++
3635 lines
123 KiB
C++
/*
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* libjingle
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* Copyright 2004 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#ifdef HAVE_WEBRTC_VOICE
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#include "talk/media/webrtc/webrtcvoiceengine.h"
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#include <algorithm>
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#include <cstdio>
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#include <string>
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#include <vector>
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#include "talk/media/base/audioframe.h"
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#include "talk/media/base/audiorenderer.h"
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#include "talk/media/base/constants.h"
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#include "talk/media/base/streamparams.h"
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#include "talk/media/base/voiceprocessor.h"
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#include "talk/media/webrtc/webrtcvoe.h"
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#include "webrtc/base/base64.h"
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#include "webrtc/base/byteorder.h"
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#include "webrtc/base/common.h"
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#include "webrtc/base/helpers.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/stringencode.h"
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#include "webrtc/base/stringutils.h"
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#include "webrtc/common.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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namespace cricket {
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static const int kMaxNumPacketSize = 6;
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struct CodecPref {
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const char* name;
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int clockrate;
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int channels;
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int payload_type;
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bool is_multi_rate;
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int packet_sizes_ms[kMaxNumPacketSize];
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};
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// Note: keep the supported packet sizes in ascending order.
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static const CodecPref kCodecPrefs[] = {
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{ kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
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{ kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
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{ kIsacCodecName, 32000, 1, 104, true, { 30 } },
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// G722 should be advertised as 8000 Hz because of the RFC "bug".
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{ kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } },
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{ kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } },
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{ kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } },
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{ kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } },
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{ kCnCodecName, 32000, 1, 106, false, { } },
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{ kCnCodecName, 16000, 1, 105, false, { } },
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{ kCnCodecName, 8000, 1, 13, false, { } },
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{ kRedCodecName, 8000, 1, 127, false, { } },
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{ kDtmfCodecName, 8000, 1, 126, false, { } },
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};
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// For Linux/Mac, using the default device is done by specifying index 0 for
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// VoE 4.0 and not -1 (which was the case for VoE 3.5).
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//
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// On Windows Vista and newer, Microsoft introduced the concept of "Default
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// Communications Device". This means that there are two types of default
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// devices (old Wave Audio style default and Default Communications Device).
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//
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// On Windows systems which only support Wave Audio style default, uses either
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// -1 or 0 to select the default device.
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//
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// On Windows systems which support both "Default Communication Device" and
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// old Wave Audio style default, use -1 for Default Communications Device and
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// -2 for Wave Audio style default, which is what we want to use for clips.
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// It's not clear yet whether the -2 index is handled properly on other OSes.
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#ifdef WIN32
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static const int kDefaultAudioDeviceId = -1;
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#else
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static const int kDefaultAudioDeviceId = 0;
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#endif
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// Parameter used for NACK.
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// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
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static const int kNackMaxPackets = 250;
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// Codec parameters for Opus.
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// draft-spittka-payload-rtp-opus-03
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// Recommended bitrates:
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// 8-12 kb/s for NB speech,
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// 16-20 kb/s for WB speech,
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// 28-40 kb/s for FB speech,
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// 48-64 kb/s for FB mono music, and
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// 64-128 kb/s for FB stereo music.
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// The current implementation applies the following values to mono signals,
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// and multiplies them by 2 for stereo.
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static const int kOpusBitrateNb = 12000;
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static const int kOpusBitrateWb = 20000;
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static const int kOpusBitrateFb = 32000;
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// Opus bitrate should be in the range between 6000 and 510000.
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static const int kOpusMinBitrate = 6000;
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static const int kOpusMaxBitrate = 510000;
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// Default audio dscp value.
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// See http://tools.ietf.org/html/rfc2474 for details.
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// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
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static const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
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// Ensure we open the file in a writeable path on ChromeOS and Android. This
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// workaround can be removed when it's possible to specify a filename for audio
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// option based AEC dumps.
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//
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// TODO(grunell): Use a string in the options instead of hardcoding it here
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// and let the embedder choose the filename (crbug.com/264223).
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//
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// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
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// below.
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#if defined(CHROMEOS)
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static const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
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#elif defined(ANDROID)
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static const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
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#else
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static const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
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#endif
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// Dumps an AudioCodec in RFC 2327-ish format.
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static std::string ToString(const AudioCodec& codec) {
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std::stringstream ss;
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ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
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<< " (" << codec.id << ")";
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return ss.str();
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}
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static std::string ToString(const webrtc::CodecInst& codec) {
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std::stringstream ss;
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ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
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<< " (" << codec.pltype << ")";
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return ss.str();
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}
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static void LogMultiline(rtc::LoggingSeverity sev, char* text) {
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const char* delim = "\r\n";
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for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
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LOG_V(sev) << tok;
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}
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}
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// Severity is an integer because it comes is assumed to be from command line.
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static int SeverityToFilter(int severity) {
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int filter = webrtc::kTraceNone;
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switch (severity) {
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case rtc::LS_VERBOSE:
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filter |= webrtc::kTraceAll;
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FALLTHROUGH();
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case rtc::LS_INFO:
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filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo);
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FALLTHROUGH();
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case rtc::LS_WARNING:
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filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning);
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FALLTHROUGH();
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case rtc::LS_ERROR:
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filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
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}
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return filter;
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}
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static bool IsCodec(const AudioCodec& codec, const char* ref_name) {
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return (_stricmp(codec.name.c_str(), ref_name) == 0);
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}
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static bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
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return (_stricmp(codec.plname, ref_name) == 0);
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}
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static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
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for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
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if (IsCodec(codec, kCodecPrefs[i].name) &&
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kCodecPrefs[i].clockrate == codec.plfreq) {
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return kCodecPrefs[i].is_multi_rate;
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}
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}
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return false;
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}
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static bool FindCodec(const std::vector<AudioCodec>& codecs,
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const AudioCodec& codec,
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AudioCodec* found_codec) {
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for (const AudioCodec& c : codecs) {
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if (c.Matches(codec)) {
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if (found_codec != NULL) {
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*found_codec = c;
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}
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return true;
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}
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}
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return false;
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}
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static bool IsNackEnabled(const AudioCodec& codec) {
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return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
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kParamValueEmpty));
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}
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static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
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int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
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for (int packet_size_ms : codec_pref.packet_sizes_ms) {
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if (packet_size_ms && packet_size_ms <= ptime_ms) {
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selected_packet_size_ms = packet_size_ms;
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}
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}
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return selected_packet_size_ms;
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}
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// If the AudioCodec param kCodecParamPTime is set, then we will set it to codec
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// pacsize if it's valid, or we will pick the next smallest value we support.
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// TODO(Brave): Query supported packet sizes from ACM when the API is ready.
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static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
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for (const CodecPref& codec_pref : kCodecPrefs) {
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if ((IsCodec(*codec, codec_pref.name) &&
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codec_pref.clockrate == codec->plfreq) ||
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IsCodec(*codec, kG722CodecName)) {
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int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
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if (packet_size_ms) {
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// Convert unit from milli-seconds to samples.
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codec->pacsize = (codec->plfreq / 1000) * packet_size_ms;
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return true;
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}
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}
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}
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return false;
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}
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// Return true if codec.params[feature] == "1", false otherwise.
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static bool IsCodecFeatureEnabled(const AudioCodec& codec,
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const char* feature) {
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int value;
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return codec.GetParam(feature, &value) && value == 1;
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}
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// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
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// otherwise. If the value (either from params or codec.bitrate) <=0, use the
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// default configuration. If the value is beyond feasible bit rate of Opus,
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// clamp it. Returns the Opus bit rate for operation.
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static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
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int bitrate = 0;
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bool use_param = true;
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if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
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bitrate = codec.bitrate;
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use_param = false;
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}
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if (bitrate <= 0) {
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if (max_playback_rate <= 8000) {
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bitrate = kOpusBitrateNb;
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} else if (max_playback_rate <= 16000) {
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bitrate = kOpusBitrateWb;
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} else {
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bitrate = kOpusBitrateFb;
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}
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if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
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bitrate *= 2;
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}
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} else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
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bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
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std::string rate_source =
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use_param ? "Codec parameter \"maxaveragebitrate\"" :
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"Supplied Opus bitrate";
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LOG(LS_WARNING) << rate_source
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<< " is invalid and is replaced by: "
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<< bitrate;
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}
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return bitrate;
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}
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// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
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// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
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static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
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int value;
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if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
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return value;
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}
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return kOpusDefaultMaxPlaybackRate;
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}
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static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
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bool* enable_codec_fec, int* max_playback_rate,
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bool* enable_codec_dtx) {
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*enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
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*enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
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*max_playback_rate = GetOpusMaxPlaybackRate(codec);
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// If OPUS, change what we send according to the "stereo" codec
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// parameter, and not the "channels" parameter. We set
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// voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
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// the bitrate is not specified, i.e. is <= zero, we set it to the
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// appropriate default value for mono or stereo Opus.
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voe_codec->channels = IsCodecFeatureEnabled(codec, kCodecParamStereo) ? 2 : 1;
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voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
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}
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// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
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// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
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// codec.
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static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
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if (IsCodec(*voe_codec, kG722CodecName)) {
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// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
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// has changed, and this special case is no longer needed.
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DCHECK(voe_codec->plfreq != new_plfreq);
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voe_codec->plfreq = new_plfreq;
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}
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}
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// Gets the default set of options applied to the engine. Historically, these
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// were supplied as a combination of flags from the channel manager (ec, agc,
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// ns, and highpass) and the rest hardcoded in InitInternal.
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static AudioOptions GetDefaultEngineOptions() {
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AudioOptions options;
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options.echo_cancellation.Set(true);
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options.auto_gain_control.Set(true);
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options.noise_suppression.Set(true);
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options.highpass_filter.Set(true);
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options.stereo_swapping.Set(false);
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options.audio_jitter_buffer_max_packets.Set(50);
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options.audio_jitter_buffer_fast_accelerate.Set(false);
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options.typing_detection.Set(true);
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options.conference_mode.Set(false);
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options.adjust_agc_delta.Set(0);
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options.experimental_agc.Set(false);
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options.extended_filter_aec.Set(false);
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options.delay_agnostic_aec.Set(false);
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options.experimental_ns.Set(false);
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options.aec_dump.Set(false);
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return options;
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}
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|
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static std::string GetEnableString(bool enable) {
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return enable ? "enable" : "disable";
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}
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WebRtcVoiceEngine::WebRtcVoiceEngine()
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: voe_wrapper_(new VoEWrapper()),
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tracing_(new VoETraceWrapper()),
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adm_(NULL),
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log_filter_(SeverityToFilter(kDefaultLogSeverity)),
|
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is_dumping_aec_(false),
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desired_local_monitor_enable_(false),
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tx_processor_ssrc_(0),
|
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rx_processor_ssrc_(0) {
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Construct();
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}
|
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|
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WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
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VoETraceWrapper* tracing)
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: voe_wrapper_(voe_wrapper),
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tracing_(tracing),
|
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adm_(NULL),
|
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log_filter_(SeverityToFilter(kDefaultLogSeverity)),
|
|
is_dumping_aec_(false),
|
|
desired_local_monitor_enable_(false),
|
|
tx_processor_ssrc_(0),
|
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rx_processor_ssrc_(0) {
|
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Construct();
|
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}
|
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|
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void WebRtcVoiceEngine::Construct() {
|
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SetTraceFilter(log_filter_);
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initialized_ = false;
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LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
|
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SetTraceOptions("");
|
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if (tracing_->SetTraceCallback(this) == -1) {
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LOG_RTCERR0(SetTraceCallback);
|
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}
|
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if (voe_wrapper_->base()->RegisterVoiceEngineObserver(*this) == -1) {
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LOG_RTCERR0(RegisterVoiceEngineObserver);
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}
|
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// Clear the default agc state.
|
|
memset(&default_agc_config_, 0, sizeof(default_agc_config_));
|
|
|
|
// Load our audio codec list.
|
|
ConstructCodecs();
|
|
|
|
// Load our RTP Header extensions.
|
|
rtp_header_extensions_.push_back(
|
|
RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
|
|
kRtpAudioLevelHeaderExtensionDefaultId));
|
|
rtp_header_extensions_.push_back(
|
|
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
|
|
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
|
|
options_ = GetDefaultEngineOptions();
|
|
}
|
|
|
|
void WebRtcVoiceEngine::ConstructCodecs() {
|
|
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
|
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int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
|
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for (int i = 0; i < ncodecs; ++i) {
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webrtc::CodecInst voe_codec;
|
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if (GetVoeCodec(i, &voe_codec)) {
|
|
// Skip uncompressed formats.
|
|
if (IsCodec(voe_codec, kL16CodecName)) {
|
|
continue;
|
|
}
|
|
|
|
const CodecPref* pref = NULL;
|
|
for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) {
|
|
if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
|
|
kCodecPrefs[j].clockrate == voe_codec.plfreq &&
|
|
kCodecPrefs[j].channels == voe_codec.channels) {
|
|
pref = &kCodecPrefs[j];
|
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break;
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}
|
|
}
|
|
|
|
if (pref) {
|
|
// Use the payload type that we've configured in our pref table;
|
|
// use the offset in our pref table to determine the sort order.
|
|
AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
|
|
voe_codec.rate, voe_codec.channels,
|
|
ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
|
|
LOG(LS_INFO) << ToString(codec);
|
|
if (IsCodec(codec, kIsacCodecName)) {
|
|
// Indicate auto-bitrate in signaling.
|
|
codec.bitrate = 0;
|
|
}
|
|
if (IsCodec(codec, kOpusCodecName)) {
|
|
// Only add fmtp parameters that differ from the spec.
|
|
if (kPreferredMinPTime != kOpusDefaultMinPTime) {
|
|
codec.params[kCodecParamMinPTime] =
|
|
rtc::ToString(kPreferredMinPTime);
|
|
}
|
|
if (kPreferredMaxPTime != kOpusDefaultMaxPTime) {
|
|
codec.params[kCodecParamMaxPTime] =
|
|
rtc::ToString(kPreferredMaxPTime);
|
|
}
|
|
codec.SetParam(kCodecParamUseInbandFec, 1);
|
|
|
|
// TODO(hellner): Add ptime, sprop-stereo, and stereo
|
|
// when they can be set to values other than the default.
|
|
}
|
|
codecs_.push_back(codec);
|
|
} else {
|
|
LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec);
|
|
}
|
|
}
|
|
}
|
|
// Make sure they are in local preference order.
|
|
std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable);
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst* codec) {
|
|
if (voe_wrapper_->codec()->GetCodec(index, *codec) == -1) {
|
|
return false;
|
|
}
|
|
// Change the sample rate of G722 to 8000 to match SDP.
|
|
MaybeFixupG722(codec, 8000);
|
|
return true;
|
|
}
|
|
|
|
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
|
|
LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
|
|
if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) {
|
|
LOG_RTCERR0(DeRegisterVoiceEngineObserver);
|
|
}
|
|
if (adm_) {
|
|
voe_wrapper_.reset();
|
|
adm_->Release();
|
|
adm_ = NULL;
|
|
}
|
|
|
|
// Test to see if the media processor was deregistered properly
|
|
DCHECK(SignalRxMediaFrame.is_empty());
|
|
DCHECK(SignalTxMediaFrame.is_empty());
|
|
|
|
tracing_->SetTraceCallback(NULL);
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
|
|
DCHECK(worker_thread == rtc::Thread::Current());
|
|
LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
|
|
bool res = InitInternal();
|
|
if (res) {
|
|
LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
|
|
} else {
|
|
LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
|
|
Terminate();
|
|
}
|
|
return res;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::InitInternal() {
|
|
// Temporarily turn logging level up for the Init call
|
|
int old_filter = log_filter_;
|
|
int extended_filter = log_filter_ | SeverityToFilter(rtc::LS_INFO);
|
|
SetTraceFilter(extended_filter);
|
|
SetTraceOptions("");
|
|
|
|
// Init WebRtc VoiceEngine.
|
|
if (voe_wrapper_->base()->Init(adm_) == -1) {
|
|
LOG_RTCERR0_EX(Init, voe_wrapper_->error());
|
|
SetTraceFilter(old_filter);
|
|
return false;
|
|
}
|
|
|
|
SetTraceFilter(old_filter);
|
|
SetTraceOptions(log_options_);
|
|
|
|
// Log the VoiceEngine version info
|
|
char buffer[1024] = "";
|
|
voe_wrapper_->base()->GetVersion(buffer);
|
|
LOG(LS_INFO) << "WebRtc VoiceEngine Version:";
|
|
LogMultiline(rtc::LS_INFO, buffer);
|
|
|
|
// Save the default AGC configuration settings. This must happen before
|
|
// calling SetOptions or the default will be overwritten.
|
|
if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
|
|
LOG_RTCERR0(GetAgcConfig);
|
|
return false;
|
|
}
|
|
|
|
// Set defaults for options, so that ApplyOptions applies them explicitly
|
|
// when we clear option (channel) overrides. External clients can still
|
|
// modify the defaults via SetOptions (on the media engine).
|
|
if (!SetOptions(GetDefaultEngineOptions())) {
|
|
return false;
|
|
}
|
|
|
|
// Print our codec list again for the call diagnostic log
|
|
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
|
|
for (const AudioCodec& codec : codecs_) {
|
|
LOG(LS_INFO) << ToString(codec);
|
|
}
|
|
|
|
// Disable the DTMF playout when a tone is sent.
|
|
// PlayDtmfTone will be used if local playout is needed.
|
|
if (voe_wrapper_->dtmf()->SetDtmfFeedbackStatus(false) == -1) {
|
|
LOG_RTCERR1(SetDtmfFeedbackStatus, false);
|
|
}
|
|
|
|
initialized_ = true;
|
|
return true;
|
|
}
|
|
|
|
void WebRtcVoiceEngine::Terminate() {
|
|
LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
|
|
initialized_ = false;
|
|
|
|
StopAecDump();
|
|
|
|
voe_wrapper_->base()->Terminate();
|
|
desired_local_monitor_enable_ = false;
|
|
}
|
|
|
|
int WebRtcVoiceEngine::GetCapabilities() {
|
|
return AUDIO_SEND | AUDIO_RECV;
|
|
}
|
|
|
|
VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
|
|
const AudioOptions& options) {
|
|
WebRtcVoiceMediaChannel* ch = new WebRtcVoiceMediaChannel(this, call);
|
|
if (!ch->valid()) {
|
|
delete ch;
|
|
return nullptr;
|
|
}
|
|
if (!ch->SetOptions(options)) {
|
|
LOG(LS_WARNING) << "Failed to set options while creating channel.";
|
|
}
|
|
return ch;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
|
|
if (!ApplyOptions(options)) {
|
|
return false;
|
|
}
|
|
options_ = options;
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::SetOptionOverrides(const AudioOptions& overrides) {
|
|
LOG(LS_INFO) << "Setting option overrides: " << overrides.ToString();
|
|
if (!ApplyOptions(overrides)) {
|
|
return false;
|
|
}
|
|
option_overrides_ = overrides;
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::ClearOptionOverrides() {
|
|
LOG(LS_INFO) << "Clearing option overrides.";
|
|
AudioOptions options = options_;
|
|
// Only call ApplyOptions if |options_overrides_| contains overrided options.
|
|
// ApplyOptions affects NS, AGC other options that is shared between
|
|
// all WebRtcVoiceEngineChannels.
|
|
if (option_overrides_ == AudioOptions()) {
|
|
return true;
|
|
}
|
|
|
|
if (!ApplyOptions(options)) {
|
|
return false;
|
|
}
|
|
option_overrides_ = AudioOptions();
|
|
return true;
|
|
}
|
|
|
|
// AudioOptions defaults are set in InitInternal (for options with corresponding
|
|
// MediaEngineInterface flags) and in SetOptions(int) for flagless options.
|
|
bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
|
|
AudioOptions options = options_in; // The options are modified below.
|
|
// kEcConference is AEC with high suppression.
|
|
webrtc::EcModes ec_mode = webrtc::kEcConference;
|
|
webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
|
|
webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
|
|
webrtc::NsModes ns_mode = webrtc::kNsHighSuppression;
|
|
bool aecm_comfort_noise = false;
|
|
if (options.aecm_generate_comfort_noise.Get(&aecm_comfort_noise)) {
|
|
LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
|
|
<< aecm_comfort_noise << " (default is false).";
|
|
}
|
|
|
|
#if defined(IOS)
|
|
// On iOS, VPIO provides built-in EC and AGC.
|
|
options.echo_cancellation.Set(false);
|
|
options.auto_gain_control.Set(false);
|
|
LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead.";
|
|
#elif defined(ANDROID)
|
|
ec_mode = webrtc::kEcAecm;
|
|
#endif
|
|
|
|
#if defined(IOS) || defined(ANDROID)
|
|
// Set the AGC mode for iOS as well despite disabling it above, to avoid
|
|
// unsupported configuration errors from webrtc.
|
|
agc_mode = webrtc::kAgcFixedDigital;
|
|
options.typing_detection.Set(false);
|
|
options.experimental_agc.Set(false);
|
|
options.extended_filter_aec.Set(false);
|
|
options.experimental_ns.Set(false);
|
|
#endif
|
|
|
|
// Delay Agnostic AEC automatically turns on EC if not set except on iOS
|
|
// where the feature is not supported.
|
|
bool use_delay_agnostic_aec = false;
|
|
#if !defined(IOS)
|
|
if (options.delay_agnostic_aec.Get(&use_delay_agnostic_aec)) {
|
|
if (use_delay_agnostic_aec) {
|
|
options.echo_cancellation.Set(true);
|
|
options.extended_filter_aec.Set(true);
|
|
ec_mode = webrtc::kEcConference;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
LOG(LS_INFO) << "Applying audio options: " << options.ToString();
|
|
|
|
webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
|
|
|
|
bool echo_cancellation = false;
|
|
if (options.echo_cancellation.Get(&echo_cancellation)) {
|
|
// Check if platform supports built-in EC. Currently only supported on
|
|
// Android and in combination with Java based audio layer.
|
|
// TODO(henrika): investigate possibility to support built-in EC also
|
|
// in combination with Open SL ES audio.
|
|
const bool built_in_aec = voe_wrapper_->hw()->BuiltInAECIsAvailable();
|
|
if (built_in_aec) {
|
|
// Built-in EC exists on this device and use_delay_agnostic_aec is not
|
|
// overriding it. Enable/Disable it according to the echo_cancellation
|
|
// audio option.
|
|
const bool enable_built_in_aec =
|
|
echo_cancellation && !use_delay_agnostic_aec;
|
|
if (voe_wrapper_->hw()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
|
|
enable_built_in_aec) {
|
|
// Disable internal software EC if built-in EC is enabled,
|
|
// i.e., replace the software EC with the built-in EC.
|
|
options.echo_cancellation.Set(false);
|
|
echo_cancellation = false;
|
|
LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
|
|
}
|
|
}
|
|
if (voep->SetEcStatus(echo_cancellation, ec_mode) == -1) {
|
|
LOG_RTCERR2(SetEcStatus, echo_cancellation, ec_mode);
|
|
return false;
|
|
} else {
|
|
LOG(LS_INFO) << "Echo control set to " << echo_cancellation
|
|
<< " with mode " << ec_mode;
|
|
}
|
|
#if !defined(ANDROID)
|
|
// TODO(ajm): Remove the error return on Android from webrtc.
|
|
if (voep->SetEcMetricsStatus(echo_cancellation) == -1) {
|
|
LOG_RTCERR1(SetEcMetricsStatus, echo_cancellation);
|
|
return false;
|
|
}
|
|
#endif
|
|
if (ec_mode == webrtc::kEcAecm) {
|
|
if (voep->SetAecmMode(aecm_mode, aecm_comfort_noise) != 0) {
|
|
LOG_RTCERR2(SetAecmMode, aecm_mode, aecm_comfort_noise);
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
|
|
bool auto_gain_control;
|
|
if (options.auto_gain_control.Get(&auto_gain_control)) {
|
|
if (voep->SetAgcStatus(auto_gain_control, agc_mode) == -1) {
|
|
LOG_RTCERR2(SetAgcStatus, auto_gain_control, agc_mode);
|
|
return false;
|
|
} else {
|
|
LOG(LS_INFO) << "Auto gain set to " << auto_gain_control << " with mode "
|
|
<< agc_mode;
|
|
}
|
|
}
|
|
|
|
if (options.tx_agc_target_dbov.IsSet() ||
|
|
options.tx_agc_digital_compression_gain.IsSet() ||
|
|
options.tx_agc_limiter.IsSet()) {
|
|
// Override default_agc_config_. Generally, an unset option means "leave
|
|
// the VoE bits alone" in this function, so we want whatever is set to be
|
|
// stored as the new "default". If we didn't, then setting e.g.
|
|
// tx_agc_target_dbov would reset digital compression gain and limiter
|
|
// settings.
|
|
// Also, if we don't update default_agc_config_, then adjust_agc_delta
|
|
// would be an offset from the original values, and not whatever was set
|
|
// explicitly.
|
|
default_agc_config_.targetLeveldBOv =
|
|
options.tx_agc_target_dbov.GetWithDefaultIfUnset(
|
|
default_agc_config_.targetLeveldBOv);
|
|
default_agc_config_.digitalCompressionGaindB =
|
|
options.tx_agc_digital_compression_gain.GetWithDefaultIfUnset(
|
|
default_agc_config_.digitalCompressionGaindB);
|
|
default_agc_config_.limiterEnable =
|
|
options.tx_agc_limiter.GetWithDefaultIfUnset(
|
|
default_agc_config_.limiterEnable);
|
|
if (voe_wrapper_->processing()->SetAgcConfig(default_agc_config_) == -1) {
|
|
LOG_RTCERR3(SetAgcConfig,
|
|
default_agc_config_.targetLeveldBOv,
|
|
default_agc_config_.digitalCompressionGaindB,
|
|
default_agc_config_.limiterEnable);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool noise_suppression;
|
|
if (options.noise_suppression.Get(&noise_suppression)) {
|
|
if (voep->SetNsStatus(noise_suppression, ns_mode) == -1) {
|
|
LOG_RTCERR2(SetNsStatus, noise_suppression, ns_mode);
|
|
return false;
|
|
} else {
|
|
LOG(LS_VERBOSE) << "Noise suppression set to " << noise_suppression
|
|
<< " with mode " << ns_mode;
|
|
}
|
|
}
|
|
|
|
bool highpass_filter;
|
|
if (options.highpass_filter.Get(&highpass_filter)) {
|
|
LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter;
|
|
if (voep->EnableHighPassFilter(highpass_filter) == -1) {
|
|
LOG_RTCERR1(SetHighpassFilterStatus, highpass_filter);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool stereo_swapping;
|
|
if (options.stereo_swapping.Get(&stereo_swapping)) {
|
|
LOG(LS_INFO) << "Stereo swapping enabled? " << stereo_swapping;
|
|
voep->EnableStereoChannelSwapping(stereo_swapping);
|
|
if (voep->IsStereoChannelSwappingEnabled() != stereo_swapping) {
|
|
LOG_RTCERR1(EnableStereoChannelSwapping, stereo_swapping);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
int audio_jitter_buffer_max_packets;
|
|
if (options.audio_jitter_buffer_max_packets.Get(
|
|
&audio_jitter_buffer_max_packets)) {
|
|
LOG(LS_INFO) << "NetEq capacity is " << audio_jitter_buffer_max_packets;
|
|
voe_config_.Set<webrtc::NetEqCapacityConfig>(
|
|
new webrtc::NetEqCapacityConfig(audio_jitter_buffer_max_packets));
|
|
}
|
|
|
|
bool audio_jitter_buffer_fast_accelerate;
|
|
if (options.audio_jitter_buffer_fast_accelerate.Get(
|
|
&audio_jitter_buffer_fast_accelerate)) {
|
|
LOG(LS_INFO) << "NetEq fast mode? " << audio_jitter_buffer_fast_accelerate;
|
|
voe_config_.Set<webrtc::NetEqFastAccelerate>(
|
|
new webrtc::NetEqFastAccelerate(audio_jitter_buffer_fast_accelerate));
|
|
}
|
|
|
|
bool typing_detection;
|
|
if (options.typing_detection.Get(&typing_detection)) {
|
|
LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection;
|
|
if (voep->SetTypingDetectionStatus(typing_detection) == -1) {
|
|
// In case of error, log the info and continue
|
|
LOG_RTCERR1(SetTypingDetectionStatus, typing_detection);
|
|
}
|
|
}
|
|
|
|
int adjust_agc_delta;
|
|
if (options.adjust_agc_delta.Get(&adjust_agc_delta)) {
|
|
LOG(LS_INFO) << "Adjust agc delta is " << adjust_agc_delta;
|
|
if (!AdjustAgcLevel(adjust_agc_delta)) {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool aec_dump;
|
|
if (options.aec_dump.Get(&aec_dump)) {
|
|
LOG(LS_INFO) << "Aec dump is enabled? " << aec_dump;
|
|
if (aec_dump)
|
|
StartAecDump(kAecDumpByAudioOptionFilename);
|
|
else
|
|
StopAecDump();
|
|
}
|
|
|
|
webrtc::Config config;
|
|
|
|
delay_agnostic_aec_.SetFrom(options.delay_agnostic_aec);
|
|
bool delay_agnostic_aec;
|
|
if (delay_agnostic_aec_.Get(&delay_agnostic_aec)) {
|
|
LOG(LS_INFO) << "Delay agnostic aec is enabled? " << delay_agnostic_aec;
|
|
config.Set<webrtc::DelayAgnostic>(
|
|
new webrtc::DelayAgnostic(delay_agnostic_aec));
|
|
}
|
|
|
|
extended_filter_aec_.SetFrom(options.extended_filter_aec);
|
|
bool extended_filter;
|
|
if (extended_filter_aec_.Get(&extended_filter)) {
|
|
LOG(LS_INFO) << "Extended filter aec is enabled? " << extended_filter;
|
|
config.Set<webrtc::ExtendedFilter>(
|
|
new webrtc::ExtendedFilter(extended_filter));
|
|
}
|
|
|
|
experimental_ns_.SetFrom(options.experimental_ns);
|
|
bool experimental_ns;
|
|
if (experimental_ns_.Get(&experimental_ns)) {
|
|
LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
|
|
config.Set<webrtc::ExperimentalNs>(
|
|
new webrtc::ExperimentalNs(experimental_ns));
|
|
}
|
|
|
|
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
|
|
// returns NULL on audio_processing().
|
|
webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
|
|
if (audioproc) {
|
|
audioproc->SetExtraOptions(config);
|
|
}
|
|
|
|
uint32 recording_sample_rate;
|
|
if (options.recording_sample_rate.Get(&recording_sample_rate)) {
|
|
LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate;
|
|
if (voe_wrapper_->hw()->SetRecordingSampleRate(recording_sample_rate)) {
|
|
LOG_RTCERR1(SetRecordingSampleRate, recording_sample_rate);
|
|
}
|
|
}
|
|
|
|
uint32 playout_sample_rate;
|
|
if (options.playout_sample_rate.Get(&playout_sample_rate)) {
|
|
LOG(LS_INFO) << "Playout sample rate is " << playout_sample_rate;
|
|
if (voe_wrapper_->hw()->SetPlayoutSampleRate(playout_sample_rate)) {
|
|
LOG_RTCERR1(SetPlayoutSampleRate, playout_sample_rate);
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::SetDelayOffset(int offset) {
|
|
voe_wrapper_->processing()->SetDelayOffsetMs(offset);
|
|
if (voe_wrapper_->processing()->DelayOffsetMs() != offset) {
|
|
LOG_RTCERR1(SetDelayOffsetMs, offset);
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
struct ResumeEntry {
|
|
ResumeEntry(WebRtcVoiceMediaChannel *c, bool p, SendFlags s)
|
|
: channel(c),
|
|
playout(p),
|
|
send(s) {
|
|
}
|
|
|
|
WebRtcVoiceMediaChannel *channel;
|
|
bool playout;
|
|
SendFlags send;
|
|
};
|
|
|
|
// TODO(juberti): Refactor this so that the core logic can be used to set the
|
|
// soundclip device. At that time, reinstate the soundclip pause/resume code.
|
|
bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
|
|
const Device* out_device) {
|
|
#if !defined(IOS)
|
|
int in_id = in_device ? rtc::FromString<int>(in_device->id) :
|
|
kDefaultAudioDeviceId;
|
|
int out_id = out_device ? rtc::FromString<int>(out_device->id) :
|
|
kDefaultAudioDeviceId;
|
|
// The device manager uses -1 as the default device, which was the case for
|
|
// VoE 3.5. VoE 4.0, however, uses 0 as the default in Linux and Mac.
|
|
#ifndef WIN32
|
|
if (-1 == in_id) {
|
|
in_id = kDefaultAudioDeviceId;
|
|
}
|
|
if (-1 == out_id) {
|
|
out_id = kDefaultAudioDeviceId;
|
|
}
|
|
#endif
|
|
|
|
std::string in_name = (in_id != kDefaultAudioDeviceId) ?
|
|
in_device->name : "Default device";
|
|
std::string out_name = (out_id != kDefaultAudioDeviceId) ?
|
|
out_device->name : "Default device";
|
|
LOG(LS_INFO) << "Setting microphone to (id=" << in_id << ", name=" << in_name
|
|
<< ") and speaker to (id=" << out_id << ", name=" << out_name
|
|
<< ")";
|
|
|
|
// If we're running the local monitor, we need to stop it first.
|
|
bool ret = true;
|
|
if (!PauseLocalMonitor()) {
|
|
LOG(LS_WARNING) << "Failed to pause local monitor";
|
|
ret = false;
|
|
}
|
|
|
|
// Must also pause all audio playback and capture.
|
|
for (WebRtcVoiceMediaChannel* channel : channels_) {
|
|
if (!channel->PausePlayout()) {
|
|
LOG(LS_WARNING) << "Failed to pause playout";
|
|
ret = false;
|
|
}
|
|
if (!channel->PauseSend()) {
|
|
LOG(LS_WARNING) << "Failed to pause send";
|
|
ret = false;
|
|
}
|
|
}
|
|
|
|
// Find the recording device id in VoiceEngine and set recording device.
|
|
if (!FindWebRtcAudioDeviceId(true, in_name, in_id, &in_id)) {
|
|
ret = false;
|
|
}
|
|
if (ret) {
|
|
if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
|
|
LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
|
|
ret = false;
|
|
}
|
|
webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
|
|
if (ap)
|
|
ap->Initialize();
|
|
}
|
|
|
|
// Find the playout device id in VoiceEngine and set playout device.
|
|
if (!FindWebRtcAudioDeviceId(false, out_name, out_id, &out_id)) {
|
|
LOG(LS_WARNING) << "Failed to find VoiceEngine device id for " << out_name;
|
|
ret = false;
|
|
}
|
|
if (ret) {
|
|
if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
|
|
LOG_RTCERR2(SetPlayoutDevice, out_name, out_id);
|
|
ret = false;
|
|
}
|
|
}
|
|
|
|
// Resume all audio playback and capture.
|
|
for (WebRtcVoiceMediaChannel* channel : channels_) {
|
|
if (!channel->ResumePlayout()) {
|
|
LOG(LS_WARNING) << "Failed to resume playout";
|
|
ret = false;
|
|
}
|
|
if (!channel->ResumeSend()) {
|
|
LOG(LS_WARNING) << "Failed to resume send";
|
|
ret = false;
|
|
}
|
|
}
|
|
|
|
// Resume local monitor.
|
|
if (!ResumeLocalMonitor()) {
|
|
LOG(LS_WARNING) << "Failed to resume local monitor";
|
|
ret = false;
|
|
}
|
|
|
|
if (ret) {
|
|
LOG(LS_INFO) << "Set microphone to (id=" << in_id <<" name=" << in_name
|
|
<< ") and speaker to (id="<< out_id << " name=" << out_name
|
|
<< ")";
|
|
}
|
|
|
|
return ret;
|
|
#else
|
|
return true;
|
|
#endif // !IOS
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::FindWebRtcAudioDeviceId(
|
|
bool is_input, const std::string& dev_name, int dev_id, int* rtc_id) {
|
|
// In Linux, VoiceEngine uses the same device dev_id as the device manager.
|
|
#if defined(LINUX) || defined(ANDROID)
|
|
*rtc_id = dev_id;
|
|
return true;
|
|
#else
|
|
// In Windows and Mac, we need to find the VoiceEngine device id by name
|
|
// unless the input dev_id is the default device id.
|
|
if (kDefaultAudioDeviceId == dev_id) {
|
|
*rtc_id = dev_id;
|
|
return true;
|
|
}
|
|
|
|
// Get the number of VoiceEngine audio devices.
|
|
int count = 0;
|
|
if (is_input) {
|
|
if (-1 == voe_wrapper_->hw()->GetNumOfRecordingDevices(count)) {
|
|
LOG_RTCERR0(GetNumOfRecordingDevices);
|
|
return false;
|
|
}
|
|
} else {
|
|
if (-1 == voe_wrapper_->hw()->GetNumOfPlayoutDevices(count)) {
|
|
LOG_RTCERR0(GetNumOfPlayoutDevices);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
for (int i = 0; i < count; ++i) {
|
|
char name[128];
|
|
char guid[128];
|
|
if (is_input) {
|
|
voe_wrapper_->hw()->GetRecordingDeviceName(i, name, guid);
|
|
LOG(LS_VERBOSE) << "VoiceEngine microphone " << i << ": " << name;
|
|
} else {
|
|
voe_wrapper_->hw()->GetPlayoutDeviceName(i, name, guid);
|
|
LOG(LS_VERBOSE) << "VoiceEngine speaker " << i << ": " << name;
|
|
}
|
|
|
|
std::string webrtc_name(name);
|
|
if (dev_name.compare(0, webrtc_name.size(), webrtc_name) == 0) {
|
|
*rtc_id = i;
|
|
return true;
|
|
}
|
|
}
|
|
LOG(LS_WARNING) << "VoiceEngine cannot find device: " << dev_name;
|
|
return false;
|
|
#endif
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
|
|
unsigned int ulevel;
|
|
if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) {
|
|
LOG_RTCERR1(GetSpeakerVolume, level);
|
|
return false;
|
|
}
|
|
*level = ulevel;
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::SetOutputVolume(int level) {
|
|
DCHECK(level >= 0 && level <= 255);
|
|
if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
|
|
LOG_RTCERR1(SetSpeakerVolume, level);
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
int WebRtcVoiceEngine::GetInputLevel() {
|
|
unsigned int ulevel;
|
|
return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
|
|
static_cast<int>(ulevel) : -1;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::SetLocalMonitor(bool enable) {
|
|
desired_local_monitor_enable_ = enable;
|
|
return ChangeLocalMonitor(desired_local_monitor_enable_);
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::ChangeLocalMonitor(bool enable) {
|
|
// The voe file api is not available in chrome.
|
|
if (!voe_wrapper_->file()) {
|
|
return false;
|
|
}
|
|
if (enable && !monitor_) {
|
|
monitor_.reset(new WebRtcMonitorStream);
|
|
if (voe_wrapper_->file()->StartRecordingMicrophone(monitor_.get()) == -1) {
|
|
LOG_RTCERR1(StartRecordingMicrophone, monitor_.get());
|
|
// Must call Stop() because there are some cases where Start will report
|
|
// failure but still change the state, and if we leave VE in the on state
|
|
// then it could crash later when trying to invoke methods on our monitor.
|
|
voe_wrapper_->file()->StopRecordingMicrophone();
|
|
monitor_.reset();
|
|
return false;
|
|
}
|
|
} else if (!enable && monitor_) {
|
|
voe_wrapper_->file()->StopRecordingMicrophone();
|
|
monitor_.reset();
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::PauseLocalMonitor() {
|
|
return ChangeLocalMonitor(false);
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::ResumeLocalMonitor() {
|
|
return ChangeLocalMonitor(desired_local_monitor_enable_);
|
|
}
|
|
|
|
const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
|
|
return codecs_;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::FindCodec(const AudioCodec& in) {
|
|
return FindWebRtcCodec(in, NULL);
|
|
}
|
|
|
|
// Get the VoiceEngine codec that matches |in|, with the supplied settings.
|
|
bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
|
|
webrtc::CodecInst* out) {
|
|
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
|
|
for (int i = 0; i < ncodecs; ++i) {
|
|
webrtc::CodecInst voe_codec;
|
|
if (GetVoeCodec(i, &voe_codec)) {
|
|
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
|
|
voe_codec.rate, voe_codec.channels, 0);
|
|
bool multi_rate = IsCodecMultiRate(voe_codec);
|
|
// Allow arbitrary rates for ISAC to be specified.
|
|
if (multi_rate) {
|
|
// Set codec.bitrate to 0 so the check for codec.Matches() passes.
|
|
codec.bitrate = 0;
|
|
}
|
|
if (codec.Matches(in)) {
|
|
if (out) {
|
|
// Fixup the payload type.
|
|
voe_codec.pltype = in.id;
|
|
|
|
// Set bitrate if specified.
|
|
if (multi_rate && in.bitrate != 0) {
|
|
voe_codec.rate = in.bitrate;
|
|
}
|
|
|
|
// Reset G722 sample rate to 16000 to match WebRTC.
|
|
MaybeFixupG722(&voe_codec, 16000);
|
|
|
|
// Apply codec-specific settings.
|
|
if (IsCodec(codec, kIsacCodecName)) {
|
|
// If ISAC and an explicit bitrate is not specified,
|
|
// enable auto bitrate adjustment.
|
|
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
|
|
}
|
|
*out = voe_codec;
|
|
}
|
|
return true;
|
|
}
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
const std::vector<RtpHeaderExtension>&
|
|
WebRtcVoiceEngine::rtp_header_extensions() const {
|
|
return rtp_header_extensions_;
|
|
}
|
|
|
|
void WebRtcVoiceEngine::SetLogging(int min_sev, const char* filter) {
|
|
// if min_sev == -1, we keep the current log level.
|
|
if (min_sev >= 0) {
|
|
SetTraceFilter(SeverityToFilter(min_sev));
|
|
}
|
|
log_options_ = filter;
|
|
SetTraceOptions(initialized_ ? log_options_ : "");
|
|
}
|
|
|
|
int WebRtcVoiceEngine::GetLastEngineError() {
|
|
return voe_wrapper_->error();
|
|
}
|
|
|
|
void WebRtcVoiceEngine::SetTraceFilter(int filter) {
|
|
log_filter_ = filter;
|
|
tracing_->SetTraceFilter(filter);
|
|
}
|
|
|
|
// We suppport three different logging settings for VoiceEngine:
|
|
// 1. Observer callback that goes into talk diagnostic logfile.
|
|
// Use --logfile and --loglevel
|
|
//
|
|
// 2. Encrypted VoiceEngine log for debugging VoiceEngine.
|
|
// Use --voice_loglevel --voice_logfilter "tracefile file_name"
|
|
//
|
|
// 3. EC log and dump for debugging QualityEngine.
|
|
// Use --voice_loglevel --voice_logfilter "recordEC file_name"
|
|
//
|
|
// For more details see: "https://sites.google.com/a/google.com/wavelet/Home/
|
|
// Magic-Flute--RTC-Engine-/Magic-Flute-Command-Line-Parameters"
|
|
void WebRtcVoiceEngine::SetTraceOptions(const std::string& options) {
|
|
// Set encrypted trace file.
|
|
std::vector<std::string> opts;
|
|
rtc::tokenize(options, ' ', '"', '"', &opts);
|
|
std::vector<std::string>::iterator tracefile =
|
|
std::find(opts.begin(), opts.end(), "tracefile");
|
|
if (tracefile != opts.end() && ++tracefile != opts.end()) {
|
|
// Write encrypted debug output (at same loglevel) to file
|
|
// EncryptedTraceFile no longer supported.
|
|
if (tracing_->SetTraceFile(tracefile->c_str()) == -1) {
|
|
LOG_RTCERR1(SetTraceFile, *tracefile);
|
|
}
|
|
}
|
|
|
|
// Allow trace options to override the trace filter. We default
|
|
// it to log_filter_ (as a translation of libjingle log levels)
|
|
// elsewhere, but this allows clients to explicitly set webrtc
|
|
// log levels.
|
|
std::vector<std::string>::iterator tracefilter =
|
|
std::find(opts.begin(), opts.end(), "tracefilter");
|
|
if (tracefilter != opts.end() && ++tracefilter != opts.end()) {
|
|
if (!tracing_->SetTraceFilter(rtc::FromString<int>(*tracefilter))) {
|
|
LOG_RTCERR1(SetTraceFilter, *tracefilter);
|
|
}
|
|
}
|
|
|
|
// Set AEC dump file
|
|
std::vector<std::string>::iterator recordEC =
|
|
std::find(opts.begin(), opts.end(), "recordEC");
|
|
if (recordEC != opts.end()) {
|
|
++recordEC;
|
|
if (recordEC != opts.end())
|
|
StartAecDump(recordEC->c_str());
|
|
else
|
|
StopAecDump();
|
|
}
|
|
}
|
|
|
|
// Ignore spammy trace messages, mostly from the stats API when we haven't
|
|
// gotten RTCP info yet from the remote side.
|
|
bool WebRtcVoiceEngine::ShouldIgnoreTrace(const std::string& trace) {
|
|
static const char* kTracesToIgnore[] = {
|
|
"\tfailed to GetReportBlockInformation",
|
|
"GetRecCodec() failed to get received codec",
|
|
"GetReceivedRtcpStatistics: Could not get received RTP statistics",
|
|
"GetRemoteRTCPData() failed to measure statistics due to lack of received RTP and/or RTCP packets", // NOLINT
|
|
"GetRemoteRTCPData() failed to retrieve sender info for remote side",
|
|
"GetRTPStatistics() failed to measure RTT since no RTP packets have been received yet", // NOLINT
|
|
"GetRTPStatistics() failed to read RTP statistics from the RTP/RTCP module",
|
|
"GetRTPStatistics() failed to retrieve RTT from the RTP/RTCP module",
|
|
"SenderInfoReceived No received SR",
|
|
"StatisticsRTP() no statistics available",
|
|
"TransmitMixer::TypingDetection() VE_TYPING_NOISE_WARNING message has been posted", // NOLINT
|
|
"TransmitMixer::TypingDetection() pending noise-saturation warning exists", // NOLINT
|
|
"GetRecPayloadType() failed to retrieve RX payload type (error=10026)", // NOLINT
|
|
"StopPlayingFileAsMicrophone() isnot playing (error=8088)",
|
|
NULL
|
|
};
|
|
for (const char* const* p = kTracesToIgnore; *p; ++p) {
|
|
if (trace.find(*p) != std::string::npos) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
|
|
int length) {
|
|
rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
|
|
if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
|
|
sev = rtc::LS_ERROR;
|
|
else if (level == webrtc::kTraceWarning)
|
|
sev = rtc::LS_WARNING;
|
|
else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
|
|
sev = rtc::LS_INFO;
|
|
else if (level == webrtc::kTraceTerseInfo)
|
|
sev = rtc::LS_INFO;
|
|
|
|
// Skip past boilerplate prefix text
|
|
if (length < 72) {
|
|
std::string msg(trace, length);
|
|
LOG(LS_ERROR) << "Malformed webrtc log message: ";
|
|
LOG_V(sev) << msg;
|
|
} else {
|
|
std::string msg(trace + 71, length - 72);
|
|
if (!ShouldIgnoreTrace(msg)) {
|
|
LOG_V(sev) << "webrtc: " << msg;
|
|
}
|
|
}
|
|
}
|
|
|
|
void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) {
|
|
rtc::CritScope lock(&channels_cs_);
|
|
WebRtcVoiceMediaChannel* channel = NULL;
|
|
uint32 ssrc = 0;
|
|
LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
|
|
<< channel_num << ".";
|
|
if (FindChannelAndSsrc(channel_num, &channel, &ssrc)) {
|
|
DCHECK(channel != NULL);
|
|
channel->OnError(ssrc, err_code);
|
|
} else {
|
|
LOG(LS_ERROR) << "VoiceEngine channel " << channel_num
|
|
<< " could not be found in channel list when error reported.";
|
|
}
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::FindChannelAndSsrc(
|
|
int channel_num, WebRtcVoiceMediaChannel** channel, uint32* ssrc) const {
|
|
DCHECK(channel != NULL && ssrc != NULL);
|
|
|
|
*channel = NULL;
|
|
*ssrc = 0;
|
|
// Find corresponding channel and ssrc
|
|
for (WebRtcVoiceMediaChannel* ch : channels_) {
|
|
DCHECK(ch != NULL);
|
|
if (ch->FindSsrc(channel_num, ssrc)) {
|
|
*channel = ch;
|
|
return true;
|
|
}
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
// This method will search through the WebRtcVoiceMediaChannels and
|
|
// obtain the voice engine's channel number.
|
|
bool WebRtcVoiceEngine::FindChannelNumFromSsrc(
|
|
uint32 ssrc, MediaProcessorDirection direction, int* channel_num) {
|
|
DCHECK(channel_num != NULL);
|
|
DCHECK(direction == MPD_RX || direction == MPD_TX);
|
|
|
|
*channel_num = -1;
|
|
// Find corresponding channel for ssrc.
|
|
for (const WebRtcVoiceMediaChannel* ch : channels_) {
|
|
DCHECK(ch != NULL);
|
|
if (direction & MPD_RX) {
|
|
*channel_num = ch->GetReceiveChannelNum(ssrc);
|
|
}
|
|
if (*channel_num == -1 && (direction & MPD_TX)) {
|
|
*channel_num = ch->GetSendChannelNum(ssrc);
|
|
}
|
|
if (*channel_num != -1) {
|
|
return true;
|
|
}
|
|
}
|
|
LOG(LS_WARNING) << "FindChannelFromSsrc. No Channel Found for Ssrc: " << ssrc;
|
|
return false;
|
|
}
|
|
|
|
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel *channel) {
|
|
rtc::CritScope lock(&channels_cs_);
|
|
channels_.push_back(channel);
|
|
}
|
|
|
|
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel *channel) {
|
|
rtc::CritScope lock(&channels_cs_);
|
|
ChannelList::iterator i = std::find(channels_.begin(),
|
|
channels_.end(),
|
|
channel);
|
|
if (i != channels_.end()) {
|
|
channels_.erase(i);
|
|
}
|
|
}
|
|
|
|
// Adjusts the default AGC target level by the specified delta.
|
|
// NB: If we start messing with other config fields, we'll want
|
|
// to save the current webrtc::AgcConfig as well.
|
|
bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) {
|
|
webrtc::AgcConfig config = default_agc_config_;
|
|
config.targetLeveldBOv -= delta;
|
|
|
|
LOG(LS_INFO) << "Adjusting AGC level from default -"
|
|
<< default_agc_config_.targetLeveldBOv << "dB to -"
|
|
<< config.targetLeveldBOv << "dB";
|
|
|
|
if (voe_wrapper_->processing()->SetAgcConfig(config) == -1) {
|
|
LOG_RTCERR1(SetAgcConfig, config.targetLeveldBOv);
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
|
|
if (initialized_) {
|
|
LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
|
|
return false;
|
|
}
|
|
if (adm_) {
|
|
adm_->Release();
|
|
adm_ = NULL;
|
|
}
|
|
if (adm) {
|
|
adm_ = adm;
|
|
adm_->AddRef();
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
|
|
FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
|
|
if (!aec_dump_file_stream) {
|
|
LOG(LS_ERROR) << "Could not open AEC dump file stream.";
|
|
if (!rtc::ClosePlatformFile(file))
|
|
LOG(LS_WARNING) << "Could not close file.";
|
|
return false;
|
|
}
|
|
StopAecDump();
|
|
if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
|
|
webrtc::AudioProcessing::kNoError) {
|
|
LOG_RTCERR0(StartDebugRecording);
|
|
fclose(aec_dump_file_stream);
|
|
return false;
|
|
}
|
|
is_dumping_aec_ = true;
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::RegisterProcessor(
|
|
uint32 ssrc,
|
|
VoiceProcessor* voice_processor,
|
|
MediaProcessorDirection direction) {
|
|
bool register_with_webrtc = false;
|
|
int channel_id = -1;
|
|
bool success = false;
|
|
uint32* processor_ssrc = NULL;
|
|
bool found_channel = FindChannelNumFromSsrc(ssrc, direction, &channel_id);
|
|
if (voice_processor == NULL || !found_channel) {
|
|
LOG(LS_WARNING) << "Media Processing Registration Failed. ssrc: " << ssrc
|
|
<< " foundChannel: " << found_channel;
|
|
return false;
|
|
}
|
|
|
|
webrtc::ProcessingTypes processing_type;
|
|
{
|
|
rtc::CritScope cs(&signal_media_critical_);
|
|
if (direction == MPD_RX) {
|
|
processing_type = webrtc::kPlaybackAllChannelsMixed;
|
|
if (SignalRxMediaFrame.is_empty()) {
|
|
register_with_webrtc = true;
|
|
processor_ssrc = &rx_processor_ssrc_;
|
|
}
|
|
SignalRxMediaFrame.connect(voice_processor,
|
|
&VoiceProcessor::OnFrame);
|
|
} else {
|
|
processing_type = webrtc::kRecordingPerChannel;
|
|
if (SignalTxMediaFrame.is_empty()) {
|
|
register_with_webrtc = true;
|
|
processor_ssrc = &tx_processor_ssrc_;
|
|
}
|
|
SignalTxMediaFrame.connect(voice_processor,
|
|
&VoiceProcessor::OnFrame);
|
|
}
|
|
}
|
|
if (register_with_webrtc) {
|
|
// TODO(janahan): when registering consider instantiating a
|
|
// a VoeMediaProcess object and not make the engine extend the interface.
|
|
if (voe()->media() && voe()->media()->
|
|
RegisterExternalMediaProcessing(channel_id,
|
|
processing_type,
|
|
*this) != -1) {
|
|
LOG(LS_INFO) << "Media Processing Registration Succeeded. channel:"
|
|
<< channel_id;
|
|
*processor_ssrc = ssrc;
|
|
success = true;
|
|
} else {
|
|
LOG_RTCERR2(RegisterExternalMediaProcessing,
|
|
channel_id,
|
|
processing_type);
|
|
success = false;
|
|
}
|
|
} else {
|
|
// If we don't have to register with the engine, we just needed to
|
|
// connect a new processor, set success to true;
|
|
success = true;
|
|
}
|
|
return success;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::UnregisterProcessorChannel(
|
|
MediaProcessorDirection channel_direction,
|
|
uint32 ssrc,
|
|
VoiceProcessor* voice_processor,
|
|
MediaProcessorDirection processor_direction) {
|
|
bool success = true;
|
|
FrameSignal* signal;
|
|
webrtc::ProcessingTypes processing_type;
|
|
uint32* processor_ssrc = NULL;
|
|
if (channel_direction == MPD_RX) {
|
|
signal = &SignalRxMediaFrame;
|
|
processing_type = webrtc::kPlaybackAllChannelsMixed;
|
|
processor_ssrc = &rx_processor_ssrc_;
|
|
} else {
|
|
signal = &SignalTxMediaFrame;
|
|
processing_type = webrtc::kRecordingPerChannel;
|
|
processor_ssrc = &tx_processor_ssrc_;
|
|
}
|
|
|
|
int deregister_id = -1;
|
|
{
|
|
rtc::CritScope cs(&signal_media_critical_);
|
|
if ((processor_direction & channel_direction) != 0 && !signal->is_empty()) {
|
|
signal->disconnect(voice_processor);
|
|
int channel_id = -1;
|
|
bool found_channel = FindChannelNumFromSsrc(ssrc,
|
|
channel_direction,
|
|
&channel_id);
|
|
if (signal->is_empty() && found_channel) {
|
|
deregister_id = channel_id;
|
|
}
|
|
}
|
|
}
|
|
if (deregister_id != -1) {
|
|
if (voe()->media() &&
|
|
voe()->media()->DeRegisterExternalMediaProcessing(deregister_id,
|
|
processing_type) != -1) {
|
|
*processor_ssrc = 0;
|
|
LOG(LS_INFO) << "Media Processing DeRegistration Succeeded. channel:"
|
|
<< deregister_id;
|
|
} else {
|
|
LOG_RTCERR2(DeRegisterExternalMediaProcessing,
|
|
deregister_id,
|
|
processing_type);
|
|
success = false;
|
|
}
|
|
}
|
|
return success;
|
|
}
|
|
|
|
bool WebRtcVoiceEngine::UnregisterProcessor(
|
|
uint32 ssrc,
|
|
VoiceProcessor* voice_processor,
|
|
MediaProcessorDirection direction) {
|
|
bool success = true;
|
|
if (voice_processor == NULL) {
|
|
LOG(LS_WARNING) << "Media Processing Deregistration Failed. ssrc: "
|
|
<< ssrc;
|
|
return false;
|
|
}
|
|
if (!UnregisterProcessorChannel(MPD_RX, ssrc, voice_processor, direction)) {
|
|
success = false;
|
|
}
|
|
if (!UnregisterProcessorChannel(MPD_TX, ssrc, voice_processor, direction)) {
|
|
success = false;
|
|
}
|
|
return success;
|
|
}
|
|
|
|
// Implementing method from WebRtc VoEMediaProcess interface
|
|
// Do not lock mux_channel_cs_ in this callback.
|
|
void WebRtcVoiceEngine::Process(int channel,
|
|
webrtc::ProcessingTypes type,
|
|
int16_t audio10ms[],
|
|
size_t length,
|
|
int sampling_freq,
|
|
bool is_stereo) {
|
|
rtc::CritScope cs(&signal_media_critical_);
|
|
AudioFrame frame(audio10ms, length, sampling_freq, is_stereo);
|
|
if (type == webrtc::kPlaybackAllChannelsMixed) {
|
|
SignalRxMediaFrame(rx_processor_ssrc_, MPD_RX, &frame);
|
|
} else if (type == webrtc::kRecordingPerChannel) {
|
|
SignalTxMediaFrame(tx_processor_ssrc_, MPD_TX, &frame);
|
|
} else {
|
|
LOG(LS_WARNING) << "Media Processing invoked unexpectedly."
|
|
<< " channel: " << channel << " type: " << type
|
|
<< " tx_ssrc: " << tx_processor_ssrc_
|
|
<< " rx_ssrc: " << rx_processor_ssrc_;
|
|
}
|
|
}
|
|
|
|
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
|
|
if (!is_dumping_aec_) {
|
|
// Start dumping AEC when we are not dumping.
|
|
if (voe_wrapper_->processing()->StartDebugRecording(
|
|
filename.c_str()) != webrtc::AudioProcessing::kNoError) {
|
|
LOG_RTCERR1(StartDebugRecording, filename.c_str());
|
|
} else {
|
|
is_dumping_aec_ = true;
|
|
}
|
|
}
|
|
}
|
|
|
|
void WebRtcVoiceEngine::StopAecDump() {
|
|
if (is_dumping_aec_) {
|
|
// Stop dumping AEC when we are dumping.
|
|
if (voe_wrapper_->processing()->StopDebugRecording() !=
|
|
webrtc::AudioProcessing::kNoError) {
|
|
LOG_RTCERR0(StopDebugRecording);
|
|
}
|
|
is_dumping_aec_ = false;
|
|
}
|
|
}
|
|
|
|
int WebRtcVoiceEngine::CreateVoiceChannel(VoEWrapper* voice_engine_wrapper) {
|
|
return voice_engine_wrapper->base()->CreateChannel(voe_config_);
|
|
}
|
|
|
|
int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
|
|
return CreateVoiceChannel(voe_wrapper_.get());
|
|
}
|
|
|
|
class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
|
: public AudioRenderer::Sink {
|
|
public:
|
|
WebRtcVoiceChannelRenderer(int ch,
|
|
webrtc::AudioTransport* voe_audio_transport)
|
|
: channel_(ch),
|
|
voe_audio_transport_(voe_audio_transport),
|
|
renderer_(NULL) {}
|
|
~WebRtcVoiceChannelRenderer() override { Stop(); }
|
|
|
|
// Starts the rendering by setting a sink to the renderer to get data
|
|
// callback.
|
|
// This method is called on the libjingle worker thread.
|
|
// TODO(xians): Make sure Start() is called only once.
|
|
void Start(AudioRenderer* renderer) {
|
|
rtc::CritScope lock(&lock_);
|
|
DCHECK(renderer != NULL);
|
|
if (renderer_ != NULL) {
|
|
DCHECK(renderer_ == renderer);
|
|
return;
|
|
}
|
|
|
|
// TODO(xians): Remove AddChannel() call after Chrome turns on APM
|
|
// in getUserMedia by default.
|
|
renderer->AddChannel(channel_);
|
|
renderer->SetSink(this);
|
|
renderer_ = renderer;
|
|
}
|
|
|
|
// Stops rendering by setting the sink of the renderer to NULL. No data
|
|
// callback will be received after this method.
|
|
// This method is called on the libjingle worker thread.
|
|
void Stop() {
|
|
rtc::CritScope lock(&lock_);
|
|
if (renderer_ == NULL)
|
|
return;
|
|
|
|
renderer_->RemoveChannel(channel_);
|
|
renderer_->SetSink(NULL);
|
|
renderer_ = NULL;
|
|
}
|
|
|
|
// AudioRenderer::Sink implementation.
|
|
// This method is called on the audio thread.
|
|
void OnData(const void* audio_data,
|
|
int bits_per_sample,
|
|
int sample_rate,
|
|
int number_of_channels,
|
|
size_t number_of_frames) override {
|
|
voe_audio_transport_->OnData(channel_,
|
|
audio_data,
|
|
bits_per_sample,
|
|
sample_rate,
|
|
number_of_channels,
|
|
number_of_frames);
|
|
}
|
|
|
|
// Callback from the |renderer_| when it is going away. In case Start() has
|
|
// never been called, this callback won't be triggered.
|
|
void OnClose() override {
|
|
rtc::CritScope lock(&lock_);
|
|
// Set |renderer_| to NULL to make sure no more callback will get into
|
|
// the renderer.
|
|
renderer_ = NULL;
|
|
}
|
|
|
|
// Accessor to the VoE channel ID.
|
|
int channel() const { return channel_; }
|
|
|
|
private:
|
|
const int channel_;
|
|
webrtc::AudioTransport* const voe_audio_transport_;
|
|
|
|
// Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
|
|
// PeerConnection will make sure invalidating the pointer before the object
|
|
// goes away.
|
|
AudioRenderer* renderer_;
|
|
|
|
// Protects |renderer_| in Start(), Stop() and OnClose().
|
|
rtc::CriticalSection lock_;
|
|
};
|
|
|
|
// WebRtcVoiceMediaChannel
|
|
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
|
|
webrtc::Call* call)
|
|
: engine_(engine),
|
|
voe_channel_(engine->CreateMediaVoiceChannel()),
|
|
send_bitrate_setting_(false),
|
|
send_bitrate_bps_(0),
|
|
options_(),
|
|
dtmf_allowed_(false),
|
|
desired_playout_(false),
|
|
nack_enabled_(false),
|
|
playout_(false),
|
|
typing_noise_detected_(false),
|
|
desired_send_(SEND_NOTHING),
|
|
send_(SEND_NOTHING),
|
|
call_(call),
|
|
default_receive_ssrc_(0) {
|
|
engine->RegisterChannel(this);
|
|
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
|
|
<< voe_channel();
|
|
DCHECK(nullptr != call);
|
|
ConfigureSendChannel(voe_channel());
|
|
}
|
|
|
|
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
|
|
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
|
|
<< voe_channel();
|
|
|
|
// Remove any remaining send streams, the default channel will be deleted
|
|
// later.
|
|
while (!send_channels_.empty())
|
|
RemoveSendStream(send_channels_.begin()->first);
|
|
|
|
// Unregister ourselves from the engine.
|
|
engine()->UnregisterChannel(this);
|
|
// Remove any remaining streams.
|
|
while (!receive_channels_.empty()) {
|
|
RemoveRecvStream(receive_channels_.begin()->first);
|
|
}
|
|
DCHECK(receive_streams_.empty());
|
|
|
|
// Delete the default channel.
|
|
DeleteChannel(voe_channel());
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetSendParameters(
|
|
const AudioSendParameters& params) {
|
|
// TODO(pthatcher): Refactor this to be more clean now that we have
|
|
// all the information at once.
|
|
return (SetSendCodecs(params.codecs) &&
|
|
SetSendRtpHeaderExtensions(params.extensions) &&
|
|
SetMaxSendBandwidth(params.max_bandwidth_bps) &&
|
|
SetOptions(params.options));
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetRecvParameters(
|
|
const AudioRecvParameters& params) {
|
|
// TODO(pthatcher): Refactor this to be more clean now that we have
|
|
// all the information at once.
|
|
return (SetRecvCodecs(params.codecs) &&
|
|
SetRecvRtpHeaderExtensions(params.extensions));
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
|
|
LOG(LS_INFO) << "Setting voice channel options: "
|
|
<< options.ToString();
|
|
|
|
// Check if DSCP value is changed from previous.
|
|
bool dscp_option_changed = (options_.dscp != options.dscp);
|
|
|
|
// TODO(xians): Add support to set different options for different send
|
|
// streams after we support multiple APMs.
|
|
|
|
// We retain all of the existing options, and apply the given ones
|
|
// on top. This means there is no way to "clear" options such that
|
|
// they go back to the engine default.
|
|
options_.SetAll(options);
|
|
|
|
if (send_ != SEND_NOTHING) {
|
|
if (!engine()->SetOptionOverrides(options_)) {
|
|
LOG(LS_WARNING) <<
|
|
"Failed to engine SetOptionOverrides during channel SetOptions.";
|
|
return false;
|
|
}
|
|
} else {
|
|
// Will be interpreted when appropriate.
|
|
}
|
|
|
|
// Receiver-side auto gain control happens per channel, so set it here from
|
|
// options. Note that, like conference mode, setting it on the engine won't
|
|
// have the desired effect, since voice channels don't inherit options from
|
|
// the media engine when those options are applied per-channel.
|
|
bool rx_auto_gain_control;
|
|
if (options.rx_auto_gain_control.Get(&rx_auto_gain_control)) {
|
|
if (engine()->voe()->processing()->SetRxAgcStatus(
|
|
voe_channel(), rx_auto_gain_control,
|
|
webrtc::kAgcFixedDigital) == -1) {
|
|
LOG_RTCERR1(SetRxAgcStatus, rx_auto_gain_control);
|
|
return false;
|
|
} else {
|
|
LOG(LS_VERBOSE) << "Rx auto gain set to " << rx_auto_gain_control
|
|
<< " with mode " << webrtc::kAgcFixedDigital;
|
|
}
|
|
}
|
|
if (options.rx_agc_target_dbov.IsSet() ||
|
|
options.rx_agc_digital_compression_gain.IsSet() ||
|
|
options.rx_agc_limiter.IsSet()) {
|
|
webrtc::AgcConfig config;
|
|
// If only some of the options are being overridden, get the current
|
|
// settings for the channel and bail if they aren't available.
|
|
if (!options.rx_agc_target_dbov.IsSet() ||
|
|
!options.rx_agc_digital_compression_gain.IsSet() ||
|
|
!options.rx_agc_limiter.IsSet()) {
|
|
if (engine()->voe()->processing()->GetRxAgcConfig(
|
|
voe_channel(), config) != 0) {
|
|
LOG(LS_ERROR) << "Failed to get default rx agc configuration for "
|
|
<< "channel " << voe_channel() << ". Since not all rx "
|
|
<< "agc options are specified, unable to safely set rx "
|
|
<< "agc options.";
|
|
return false;
|
|
}
|
|
}
|
|
config.targetLeveldBOv =
|
|
options.rx_agc_target_dbov.GetWithDefaultIfUnset(
|
|
config.targetLeveldBOv);
|
|
config.digitalCompressionGaindB =
|
|
options.rx_agc_digital_compression_gain.GetWithDefaultIfUnset(
|
|
config.digitalCompressionGaindB);
|
|
config.limiterEnable = options.rx_agc_limiter.GetWithDefaultIfUnset(
|
|
config.limiterEnable);
|
|
if (engine()->voe()->processing()->SetRxAgcConfig(
|
|
voe_channel(), config) == -1) {
|
|
LOG_RTCERR4(SetRxAgcConfig, voe_channel(), config.targetLeveldBOv,
|
|
config.digitalCompressionGaindB, config.limiterEnable);
|
|
return false;
|
|
}
|
|
}
|
|
if (dscp_option_changed) {
|
|
rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
|
|
if (options_.dscp.GetWithDefaultIfUnset(false))
|
|
dscp = kAudioDscpValue;
|
|
if (MediaChannel::SetDscp(dscp) != 0) {
|
|
LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
|
|
}
|
|
}
|
|
|
|
RecreateAudioReceiveStreams();
|
|
|
|
LOG(LS_INFO) << "Set voice channel options. Current options: "
|
|
<< options_.ToString();
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
|
|
const std::vector<AudioCodec>& codecs) {
|
|
// Set the payload types to be used for incoming media.
|
|
LOG(LS_INFO) << "Setting receive voice codecs:";
|
|
|
|
std::vector<AudioCodec> new_codecs;
|
|
// Find all new codecs. We allow adding new codecs but don't allow changing
|
|
// the payload type of codecs that is already configured since we might
|
|
// already be receiving packets with that payload type.
|
|
for (const AudioCodec& codec : codecs) {
|
|
AudioCodec old_codec;
|
|
if (FindCodec(recv_codecs_, codec, &old_codec)) {
|
|
if (old_codec.id != codec.id) {
|
|
LOG(LS_ERROR) << codec.name << " payload type changed.";
|
|
return false;
|
|
}
|
|
} else {
|
|
new_codecs.push_back(codec);
|
|
}
|
|
}
|
|
if (new_codecs.empty()) {
|
|
// There are no new codecs to configure. Already configured codecs are
|
|
// never removed.
|
|
return true;
|
|
}
|
|
|
|
if (playout_) {
|
|
// Receive codecs can not be changed while playing. So we temporarily
|
|
// pause playout.
|
|
PausePlayout();
|
|
}
|
|
|
|
bool result = SetRecvCodecsInternal(new_codecs);
|
|
if (result) {
|
|
recv_codecs_ = codecs;
|
|
}
|
|
|
|
if (desired_playout_ && !playout_) {
|
|
ResumePlayout();
|
|
}
|
|
return result;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
|
int channel, const std::vector<AudioCodec>& codecs) {
|
|
// Disable VAD, FEC, and RED unless we know the other side wants them.
|
|
engine()->voe()->codec()->SetVADStatus(channel, false);
|
|
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
|
|
engine()->voe()->rtp()->SetREDStatus(channel, false);
|
|
engine()->voe()->codec()->SetFECStatus(channel, false);
|
|
|
|
// Scan through the list to figure out the codec to use for sending, along
|
|
// with the proper configuration for VAD and DTMF.
|
|
bool found_send_codec = false;
|
|
webrtc::CodecInst send_codec;
|
|
memset(&send_codec, 0, sizeof(send_codec));
|
|
|
|
bool nack_enabled = nack_enabled_;
|
|
bool enable_codec_fec = false;
|
|
bool enable_opus_dtx = false;
|
|
int opus_max_playback_rate = 0;
|
|
|
|
// Set send codec (the first non-telephone-event/CN codec)
|
|
for (const AudioCodec& codec : codecs) {
|
|
// Ignore codecs we don't know about. The negotiation step should prevent
|
|
// this, but double-check to be sure.
|
|
webrtc::CodecInst voe_codec;
|
|
if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
|
|
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
|
|
continue;
|
|
}
|
|
|
|
if (IsCodec(codec, kDtmfCodecName) || IsCodec(codec, kCnCodecName)) {
|
|
// Skip telephone-event/CN codec, which will be handled later.
|
|
continue;
|
|
}
|
|
|
|
// We'll use the first codec in the list to actually send audio data.
|
|
// Be sure to use the payload type requested by the remote side.
|
|
// "red", for RED audio, is a special case where the actual codec to be
|
|
// used is specified in params.
|
|
if (IsCodec(codec, kRedCodecName)) {
|
|
// Parse out the RED parameters. If we fail, just ignore RED;
|
|
// we don't support all possible params/usage scenarios.
|
|
if (!GetRedSendCodec(codec, codecs, &send_codec)) {
|
|
continue;
|
|
}
|
|
|
|
// Enable redundant encoding of the specified codec. Treat any
|
|
// failure as a fatal internal error.
|
|
LOG(LS_INFO) << "Enabling RED on channel " << channel;
|
|
if (engine()->voe()->rtp()->SetREDStatus(channel, true, codec.id) == -1) {
|
|
LOG_RTCERR3(SetREDStatus, channel, true, codec.id);
|
|
return false;
|
|
}
|
|
} else {
|
|
send_codec = voe_codec;
|
|
nack_enabled = IsNackEnabled(codec);
|
|
// For Opus as the send codec, we are to determine inband FEC, maximum
|
|
// playback rate, and opus internal dtx.
|
|
if (IsCodec(codec, kOpusCodecName)) {
|
|
GetOpusConfig(codec, &send_codec, &enable_codec_fec,
|
|
&opus_max_playback_rate, &enable_opus_dtx);
|
|
}
|
|
|
|
// Set packet size if the AudioCodec param kCodecParamPTime is set.
|
|
int ptime_ms = 0;
|
|
if (codec.GetParam(kCodecParamPTime, &ptime_ms)) {
|
|
if (!SetPTimeAsPacketSize(&send_codec, ptime_ms)) {
|
|
LOG(LS_WARNING) << "Failed to set packet size for codec "
|
|
<< send_codec.plname;
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
found_send_codec = true;
|
|
break;
|
|
}
|
|
|
|
if (nack_enabled_ != nack_enabled) {
|
|
SetNack(channel, nack_enabled);
|
|
nack_enabled_ = nack_enabled;
|
|
}
|
|
|
|
if (!found_send_codec) {
|
|
LOG(LS_WARNING) << "Received empty list of codecs.";
|
|
return false;
|
|
}
|
|
|
|
// Set the codec immediately, since SetVADStatus() depends on whether
|
|
// the current codec is mono or stereo.
|
|
if (!SetSendCodec(channel, send_codec))
|
|
return false;
|
|
|
|
// FEC should be enabled after SetSendCodec.
|
|
if (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, 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 "
|
|
<< GetEnableString(enable_opus_dtx)
|
|
<< " Opus DTX on channel "
|
|
<< channel;
|
|
if (engine()->voe()->codec()->SetOpusDtx(channel, enable_opus_dtx)) {
|
|
LOG_RTCERR2(SetOpusDtx, channel, enable_opus_dtx);
|
|
return false;
|
|
}
|
|
|
|
// If opus_max_playback_rate <= 0, the default maximum playback rate
|
|
// (48 kHz) will be used.
|
|
if (opus_max_playback_rate > 0) {
|
|
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
|
|
<< opus_max_playback_rate
|
|
<< " Hz on channel "
|
|
<< channel;
|
|
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
|
|
channel, opus_max_playback_rate) == -1) {
|
|
LOG_RTCERR2(SetOpusMaxPlaybackRate, channel, opus_max_playback_rate);
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Always update the |send_codec_| to the currently set send codec.
|
|
send_codec_.reset(new webrtc::CodecInst(send_codec));
|
|
|
|
if (send_bitrate_setting_) {
|
|
SetSendBitrateInternal(send_bitrate_bps_);
|
|
}
|
|
|
|
// Loop through the codecs list again to config the telephone-event/CN codec.
|
|
for (const AudioCodec& codec : codecs) {
|
|
// Ignore codecs we don't know about. The negotiation step should prevent
|
|
// this, but double-check to be sure.
|
|
webrtc::CodecInst voe_codec;
|
|
if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
|
|
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
|
|
continue;
|
|
}
|
|
|
|
// Find the DTMF telephone event "codec" and tell VoiceEngine channels
|
|
// about it.
|
|
if (IsCodec(codec, kDtmfCodecName)) {
|
|
if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
|
|
channel, codec.id) == -1) {
|
|
LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, codec.id);
|
|
return false;
|
|
}
|
|
} else if (IsCodec(codec, kCnCodecName)) {
|
|
// Turn voice activity detection/comfort noise on if supported.
|
|
// Set the wideband CN payload type appropriately.
|
|
// (narrowband always uses the static payload type 13).
|
|
webrtc::PayloadFrequencies cn_freq;
|
|
switch (codec.clockrate) {
|
|
case 8000:
|
|
cn_freq = webrtc::kFreq8000Hz;
|
|
break;
|
|
case 16000:
|
|
cn_freq = webrtc::kFreq16000Hz;
|
|
break;
|
|
case 32000:
|
|
cn_freq = webrtc::kFreq32000Hz;
|
|
break;
|
|
default:
|
|
LOG(LS_WARNING) << "CN frequency " << codec.clockrate
|
|
<< " not supported.";
|
|
continue;
|
|
}
|
|
// Set the CN payloadtype and the VAD status.
|
|
// The CN payload type for 8000 Hz clockrate is fixed at 13.
|
|
if (cn_freq != webrtc::kFreq8000Hz) {
|
|
if (engine()->voe()->codec()->SetSendCNPayloadType(
|
|
channel, codec.id, cn_freq) == -1) {
|
|
LOG_RTCERR3(SetSendCNPayloadType, channel, codec.id, 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 (codec.clockrate == send_codec.plfreq && send_codec.channels != 2) {
|
|
// 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::SetSendCodecs(
|
|
const std::vector<AudioCodec>& codecs) {
|
|
dtmf_allowed_ = false;
|
|
for (const AudioCodec& codec : codecs) {
|
|
// Find the DTMF telephone event "codec".
|
|
if (IsCodec(codec, kDtmfCodecName)) {
|
|
dtmf_allowed_ = true;
|
|
}
|
|
}
|
|
|
|
// Cache the codecs in order to configure the channel created later.
|
|
send_codecs_ = codecs;
|
|
for (const auto& ch : send_channels_) {
|
|
if (!SetSendCodecs(ch.second->channel(), codecs)) {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
// Set nack status on receive channels and update |nack_enabled_|.
|
|
SetNack(receive_channels_, nack_enabled_);
|
|
return true;
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::SetNack(const ChannelMap& channels,
|
|
bool nack_enabled) {
|
|
for (const auto& ch : channels) {
|
|
SetNack(ch.second->channel(), nack_enabled);
|
|
}
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::SetNack(int channel, bool nack_enabled) {
|
|
if (nack_enabled) {
|
|
LOG(LS_INFO) << "Enabling NACK for channel " << channel;
|
|
engine()->voe()->rtp()->SetNACKStatus(channel, true, kNackMaxPackets);
|
|
} else {
|
|
LOG(LS_INFO) << "Disabling NACK for channel " << channel;
|
|
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
|
|
}
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetSendCodec(
|
|
const webrtc::CodecInst& send_codec) {
|
|
LOG(LS_INFO) << "Selected voice codec " << ToString(send_codec)
|
|
<< ", bitrate=" << send_codec.rate;
|
|
for (const auto& ch : send_channels_) {
|
|
if (!SetSendCodec(ch.second->channel(), send_codec))
|
|
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;
|
|
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;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
|
|
const std::vector<RtpHeaderExtension>& extensions) {
|
|
if (receive_extensions_ == extensions) {
|
|
return true;
|
|
}
|
|
|
|
// The default channel may or may not be in |receive_channels_|. Set the rtp
|
|
// header extensions for default channel regardless.
|
|
if (!SetChannelRecvRtpHeaderExtensions(voe_channel(), extensions)) {
|
|
return false;
|
|
}
|
|
|
|
// Loop through all receive channels and enable/disable the extensions.
|
|
for (const auto& ch : receive_channels_) {
|
|
if (!SetChannelRecvRtpHeaderExtensions(ch.second->channel(), extensions)) {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
receive_extensions_ = extensions;
|
|
|
|
// Recreate AudioReceiveStream:s.
|
|
{
|
|
std::vector<webrtc::RtpExtension> exts;
|
|
|
|
const RtpHeaderExtension* audio_level_extension =
|
|
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
|
|
if (audio_level_extension) {
|
|
exts.push_back({
|
|
kRtpAudioLevelHeaderExtension, audio_level_extension->id});
|
|
}
|
|
|
|
const RtpHeaderExtension* send_time_extension =
|
|
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
|
|
if (send_time_extension) {
|
|
exts.push_back({
|
|
kRtpAbsoluteSenderTimeHeaderExtension, send_time_extension->id});
|
|
}
|
|
|
|
recv_rtp_extensions_.swap(exts);
|
|
RecreateAudioReceiveStreams();
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
|
|
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
|
|
const RtpHeaderExtension* audio_level_extension =
|
|
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
|
|
if (!SetHeaderExtension(
|
|
&webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus, channel_id,
|
|
audio_level_extension)) {
|
|
return false;
|
|
}
|
|
|
|
const RtpHeaderExtension* send_time_extension =
|
|
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
|
|
if (!SetHeaderExtension(
|
|
&webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id,
|
|
send_time_extension)) {
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
|
|
const std::vector<RtpHeaderExtension>& extensions) {
|
|
if (send_extensions_ == extensions) {
|
|
return true;
|
|
}
|
|
|
|
// The default channel may or may not be in |send_channels_|. Set the rtp
|
|
// header extensions for default channel regardless.
|
|
|
|
if (!SetChannelSendRtpHeaderExtensions(voe_channel(), extensions)) {
|
|
return false;
|
|
}
|
|
|
|
// Loop through all send channels and enable/disable the extensions.
|
|
for (const auto& ch : send_channels_) {
|
|
if (!SetChannelSendRtpHeaderExtensions(ch.second->channel(), extensions)) {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
send_extensions_ = extensions;
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetChannelSendRtpHeaderExtensions(
|
|
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
|
|
const RtpHeaderExtension* audio_level_extension =
|
|
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
|
|
|
|
if (!SetHeaderExtension(
|
|
&webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id,
|
|
audio_level_extension)) {
|
|
return false;
|
|
}
|
|
|
|
const RtpHeaderExtension* send_time_extension =
|
|
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
|
|
if (!SetHeaderExtension(
|
|
&webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id,
|
|
send_time_extension)) {
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
|
|
desired_playout_ = playout;
|
|
return ChangePlayout(desired_playout_);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::PausePlayout() {
|
|
return ChangePlayout(false);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::ResumePlayout() {
|
|
return ChangePlayout(desired_playout_);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
|
|
if (playout_ == playout) {
|
|
return true;
|
|
}
|
|
|
|
// Change the playout of all channels to the new state.
|
|
bool result = true;
|
|
if (receive_channels_.empty()) {
|
|
// Only toggle the default channel if we don't have any other channels.
|
|
result = SetPlayout(voe_channel(), playout);
|
|
}
|
|
for (const auto& ch : receive_channels_) {
|
|
if (!SetPlayout(ch.second->channel(), playout)) {
|
|
LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
|
|
<< ch.second->channel() << " failed";
|
|
result = false;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (result) {
|
|
playout_ = playout;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
|
|
desired_send_ = send;
|
|
if (!send_channels_.empty())
|
|
return ChangeSend(desired_send_);
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::PauseSend() {
|
|
return ChangeSend(SEND_NOTHING);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::ResumeSend() {
|
|
return ChangeSend(desired_send_);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) {
|
|
if (send_ == send) {
|
|
return true;
|
|
}
|
|
|
|
// Change the settings on each send channel.
|
|
if (send == SEND_MICROPHONE)
|
|
engine()->SetOptionOverrides(options_);
|
|
|
|
// Change the settings on each send channel.
|
|
for (const auto& ch : send_channels_) {
|
|
if (!ChangeSend(ch.second->channel(), send))
|
|
return false;
|
|
}
|
|
|
|
// Clear up the options after stopping sending.
|
|
if (send == SEND_NOTHING)
|
|
engine()->ClearOptionOverrides();
|
|
|
|
send_ = send;
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
|
|
if (send == SEND_MICROPHONE) {
|
|
if (engine()->voe()->base()->StartSend(channel) == -1) {
|
|
LOG_RTCERR1(StartSend, channel);
|
|
return false;
|
|
}
|
|
if (engine()->voe()->file() &&
|
|
engine()->voe()->file()->StopPlayingFileAsMicrophone(channel) == -1) {
|
|
LOG_RTCERR1(StopPlayingFileAsMicrophone, channel);
|
|
return false;
|
|
}
|
|
} else { // SEND_NOTHING
|
|
DCHECK(send == SEND_NOTHING);
|
|
if (engine()->voe()->base()->StopSend(channel) == -1) {
|
|
LOG_RTCERR1(StopSend, channel);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetAudioSend(uint32 ssrc, bool mute,
|
|
const AudioOptions* options,
|
|
AudioRenderer* renderer) {
|
|
// TODO(solenberg): The state change should be fully rolled back if any one of
|
|
// these calls fail.
|
|
if (!SetLocalRenderer(ssrc, renderer)) {
|
|
return false;
|
|
}
|
|
if (!MuteStream(ssrc, mute)) {
|
|
return false;
|
|
}
|
|
if (!mute && options) {
|
|
return SetOptions(*options);
|
|
}
|
|
return true;
|
|
}
|
|
|
|
// TODO(ronghuawu): Change this method to return bool.
|
|
void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
|
|
if (engine()->voe()->network()->RegisterExternalTransport(
|
|
channel, *this) == -1) {
|
|
LOG_RTCERR2(RegisterExternalTransport, channel, this);
|
|
}
|
|
|
|
// Enable RTCP (for quality stats and feedback messages)
|
|
EnableRtcp(channel);
|
|
|
|
// Reset all recv codecs; they will be enabled via SetRecvCodecs.
|
|
ResetRecvCodecs(channel);
|
|
|
|
// Set RTP header extension for the new channel.
|
|
SetChannelSendRtpHeaderExtensions(channel, send_extensions_);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) {
|
|
if (engine()->voe()->network()->DeRegisterExternalTransport(channel) == -1) {
|
|
LOG_RTCERR1(DeRegisterExternalTransport, channel);
|
|
}
|
|
|
|
if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
|
|
LOG_RTCERR1(DeleteChannel, channel);
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
|
|
// If the default channel is already used for sending create a new channel
|
|
// otherwise use the default channel for sending.
|
|
int channel = GetSendChannelNum(sp.first_ssrc());
|
|
if (channel != -1) {
|
|
LOG(LS_ERROR) << "Stream already exists with ssrc " << sp.first_ssrc();
|
|
return false;
|
|
}
|
|
|
|
bool default_channel_is_available = true;
|
|
for (const auto& ch : send_channels_) {
|
|
if (IsDefaultChannel(ch.second->channel())) {
|
|
default_channel_is_available = false;
|
|
break;
|
|
}
|
|
}
|
|
if (default_channel_is_available) {
|
|
channel = voe_channel();
|
|
} else {
|
|
// Create a new channel for sending audio data.
|
|
channel = engine()->CreateMediaVoiceChannel();
|
|
if (channel == -1) {
|
|
LOG_RTCERR0(CreateChannel);
|
|
return false;
|
|
}
|
|
|
|
ConfigureSendChannel(channel);
|
|
}
|
|
|
|
// Save the channel to send_channels_, so that RemoveSendStream() can still
|
|
// delete the channel in case failure happens below.
|
|
webrtc::AudioTransport* audio_transport =
|
|
engine()->voe()->base()->audio_transport();
|
|
send_channels_.insert(
|
|
std::make_pair(sp.first_ssrc(),
|
|
new WebRtcVoiceChannelRenderer(channel, audio_transport)));
|
|
|
|
// Set the send (local) SSRC.
|
|
// If there are multiple send SSRCs, we can only set the first one here, and
|
|
// the rest of the SSRC(s) need to be set after SetSendCodec has been called
|
|
// (with a codec requires multiple SSRC(s)).
|
|
if (engine()->voe()->rtp()->SetLocalSSRC(channel, sp.first_ssrc()) == -1) {
|
|
LOG_RTCERR2(SetSendSSRC, channel, sp.first_ssrc());
|
|
return false;
|
|
}
|
|
|
|
// At this point the channel's local SSRC has been updated. If the channel is
|
|
// the default channel make sure that all the receive channels are updated as
|
|
// well. Receive channels have to have the same SSRC as the default channel in
|
|
// order to send receiver reports with this SSRC.
|
|
if (IsDefaultChannel(channel)) {
|
|
for (const auto& ch : receive_channels_) {
|
|
// Only update the SSRC for non-default channels.
|
|
if (!IsDefaultChannel(ch.second->channel())) {
|
|
if (engine()->voe()->rtp()->SetLocalSSRC(ch.second->channel(),
|
|
sp.first_ssrc()) != 0) {
|
|
LOG_RTCERR2(SetLocalSSRC, ch.second->channel(), sp.first_ssrc());
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
|
|
LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
|
|
return false;
|
|
}
|
|
|
|
// Set the current codecs to be used for the new channel.
|
|
if (!send_codecs_.empty() && !SetSendCodecs(channel, send_codecs_))
|
|
return false;
|
|
|
|
return ChangeSend(channel, desired_send_);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) {
|
|
ChannelMap::iterator it = send_channels_.find(ssrc);
|
|
if (it == send_channels_.end()) {
|
|
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
|
|
<< " which doesn't exist.";
|
|
return false;
|
|
}
|
|
|
|
int channel = it->second->channel();
|
|
ChangeSend(channel, SEND_NOTHING);
|
|
|
|
// Delete the WebRtcVoiceChannelRenderer object connected to the channel,
|
|
// this will disconnect the audio renderer with the send channel.
|
|
delete it->second;
|
|
send_channels_.erase(it);
|
|
|
|
if (IsDefaultChannel(channel)) {
|
|
// Do not delete the default channel since the receive channels depend on
|
|
// the default channel, recycle it instead.
|
|
ChangeSend(channel, SEND_NOTHING);
|
|
} else {
|
|
// Clean up and delete the send channel.
|
|
LOG(LS_INFO) << "Removing audio send stream " << ssrc
|
|
<< " with VoiceEngine channel #" << channel << ".";
|
|
if (!DeleteChannel(channel))
|
|
return false;
|
|
}
|
|
|
|
if (send_channels_.empty())
|
|
ChangeSend(SEND_NOTHING);
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
|
|
DCHECK(thread_checker_.CalledOnValidThread());
|
|
rtc::CritScope lock(&receive_channels_cs_);
|
|
|
|
if (!VERIFY(sp.ssrcs.size() == 1))
|
|
return false;
|
|
uint32 ssrc = sp.first_ssrc();
|
|
|
|
if (ssrc == 0) {
|
|
LOG(LS_WARNING) << "AddRecvStream with 0 ssrc is not supported.";
|
|
return false;
|
|
}
|
|
|
|
if (receive_channels_.find(ssrc) != receive_channels_.end()) {
|
|
LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
|
|
return false;
|
|
}
|
|
|
|
DCHECK(receive_stream_params_.find(ssrc) == receive_stream_params_.end());
|
|
|
|
// Reuse default channel for recv stream in non-conference mode call
|
|
// when the default channel is not being used.
|
|
webrtc::AudioTransport* audio_transport =
|
|
engine()->voe()->base()->audio_transport();
|
|
if (!InConferenceMode() && default_receive_ssrc_ == 0) {
|
|
LOG(LS_INFO) << "Recv stream " << ssrc << " reuse default channel";
|
|
default_receive_ssrc_ = ssrc;
|
|
WebRtcVoiceChannelRenderer* channel_renderer =
|
|
new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport);
|
|
receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
|
|
receive_stream_params_[ssrc] = sp;
|
|
AddAudioReceiveStream(ssrc);
|
|
return SetPlayout(voe_channel(), playout_);
|
|
}
|
|
|
|
// Create a new channel for receiving audio data.
|
|
int channel = engine()->CreateMediaVoiceChannel();
|
|
if (channel == -1) {
|
|
LOG_RTCERR0(CreateChannel);
|
|
return false;
|
|
}
|
|
|
|
if (!ConfigureRecvChannel(channel)) {
|
|
DeleteChannel(channel);
|
|
return false;
|
|
}
|
|
|
|
WebRtcVoiceChannelRenderer* channel_renderer =
|
|
new WebRtcVoiceChannelRenderer(channel, audio_transport);
|
|
receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
|
|
receive_stream_params_[ssrc] = sp;
|
|
AddAudioReceiveStream(ssrc);
|
|
|
|
LOG(LS_INFO) << "New audio stream " << ssrc
|
|
<< " registered to VoiceEngine channel #"
|
|
<< channel << ".";
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
|
|
// Configure to use external transport, like our default channel.
|
|
if (engine()->voe()->network()->RegisterExternalTransport(
|
|
channel, *this) == -1) {
|
|
LOG_RTCERR2(SetExternalTransport, channel, this);
|
|
return false;
|
|
}
|
|
|
|
// Use the same SSRC as our default channel (so the RTCP reports are correct).
|
|
unsigned int send_ssrc = 0;
|
|
webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp();
|
|
if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) {
|
|
LOG_RTCERR1(GetSendSSRC, channel);
|
|
return false;
|
|
}
|
|
if (rtp->SetLocalSSRC(channel, send_ssrc) == -1) {
|
|
LOG_RTCERR1(SetSendSSRC, channel);
|
|
return false;
|
|
}
|
|
|
|
// Associate receive channel to default channel (so the receive channel can
|
|
// obtain RTT from the send channel)
|
|
engine()->voe()->base()->AssociateSendChannel(channel, voe_channel());
|
|
LOG(LS_INFO) << "VoiceEngine channel #"
|
|
<< channel << " is associated with channel #"
|
|
<< voe_channel() << ".";
|
|
|
|
// Use the same recv payload types as our default channel.
|
|
ResetRecvCodecs(channel);
|
|
if (!recv_codecs_.empty()) {
|
|
for (const auto& codec : recv_codecs_) {
|
|
webrtc::CodecInst voe_codec;
|
|
if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
|
|
voe_codec.pltype = codec.id;
|
|
voe_codec.rate = 0; // Needed to make GetRecPayloadType work for ISAC
|
|
if (engine()->voe()->codec()->GetRecPayloadType(
|
|
voe_channel(), voe_codec) != -1) {
|
|
if (engine()->voe()->codec()->SetRecPayloadType(
|
|
channel, voe_codec) == -1) {
|
|
LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (InConferenceMode()) {
|
|
// To be in par with the video, voe_channel() is not used for receiving in
|
|
// a conference call.
|
|
if (receive_channels_.empty() && default_receive_ssrc_ == 0 && playout_) {
|
|
// This is the first stream in a multi user meeting. We can now
|
|
// disable playback of the default stream. This since the default
|
|
// stream will probably have received some initial packets before
|
|
// the new stream was added. This will mean that the CN state from
|
|
// the default channel will be mixed in with the other streams
|
|
// throughout the whole meeting, which might be disturbing.
|
|
LOG(LS_INFO) << "Disabling playback on the default voice channel";
|
|
SetPlayout(voe_channel(), false);
|
|
}
|
|
}
|
|
SetNack(channel, nack_enabled_);
|
|
|
|
// Set RTP header extension for the new channel.
|
|
if (!SetChannelRecvRtpHeaderExtensions(channel, receive_extensions_)) {
|
|
return false;
|
|
}
|
|
|
|
return SetPlayout(channel, playout_);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
|
|
DCHECK(thread_checker_.CalledOnValidThread());
|
|
rtc::CritScope lock(&receive_channels_cs_);
|
|
ChannelMap::iterator it = receive_channels_.find(ssrc);
|
|
if (it == receive_channels_.end()) {
|
|
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
|
|
<< " which doesn't exist.";
|
|
return false;
|
|
}
|
|
|
|
RemoveAudioReceiveStream(ssrc);
|
|
receive_stream_params_.erase(ssrc);
|
|
|
|
// Delete the WebRtcVoiceChannelRenderer object connected to the channel, this
|
|
// will disconnect the audio renderer with the receive channel.
|
|
// Cache the channel before the deletion.
|
|
const int channel = it->second->channel();
|
|
delete it->second;
|
|
receive_channels_.erase(it);
|
|
|
|
if (ssrc == default_receive_ssrc_) {
|
|
DCHECK(IsDefaultChannel(channel));
|
|
// Recycle the default channel is for recv stream.
|
|
if (playout_)
|
|
SetPlayout(voe_channel(), false);
|
|
|
|
default_receive_ssrc_ = 0;
|
|
return true;
|
|
}
|
|
|
|
LOG(LS_INFO) << "Removing audio stream " << ssrc
|
|
<< " with VoiceEngine channel #" << channel << ".";
|
|
if (!DeleteChannel(channel))
|
|
return false;
|
|
|
|
bool enable_default_channel_playout = false;
|
|
if (receive_channels_.empty()) {
|
|
// The last stream was removed. We can now enable the default
|
|
// channel for new channels to be played out immediately without
|
|
// waiting for AddStream messages.
|
|
// We do this for both conference mode and non-conference mode.
|
|
// TODO(oja): Does the default channel still have it's CN state?
|
|
enable_default_channel_playout = true;
|
|
}
|
|
if (!InConferenceMode() && receive_channels_.size() == 1 &&
|
|
default_receive_ssrc_ != 0) {
|
|
// Only the default channel is active, enable the playout on default
|
|
// channel.
|
|
enable_default_channel_playout = true;
|
|
}
|
|
if (enable_default_channel_playout && playout_) {
|
|
LOG(LS_INFO) << "Enabling playback on the default voice channel";
|
|
SetPlayout(voe_channel(), true);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc,
|
|
AudioRenderer* renderer) {
|
|
ChannelMap::iterator it = receive_channels_.find(ssrc);
|
|
if (it == receive_channels_.end()) {
|
|
if (renderer) {
|
|
// Return an error if trying to set a valid renderer with an invalid ssrc.
|
|
LOG(LS_ERROR) << "SetRemoteRenderer failed with ssrc "<< ssrc;
|
|
return false;
|
|
}
|
|
|
|
// The channel likely has gone away, do nothing.
|
|
return true;
|
|
}
|
|
|
|
if (renderer)
|
|
it->second->Start(renderer);
|
|
else
|
|
it->second->Stop();
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32 ssrc,
|
|
AudioRenderer* renderer) {
|
|
ChannelMap::iterator it = send_channels_.find(ssrc);
|
|
if (it == send_channels_.end()) {
|
|
if (renderer) {
|
|
// Return an error if trying to set a valid renderer with an invalid ssrc.
|
|
LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
|
|
return false;
|
|
}
|
|
|
|
// The channel likely has gone away, do nothing.
|
|
return true;
|
|
}
|
|
|
|
if (renderer)
|
|
it->second->Start(renderer);
|
|
else
|
|
it->second->Stop();
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::GetActiveStreams(
|
|
AudioInfo::StreamList* actives) {
|
|
// In conference mode, the default channel should not be in
|
|
// |receive_channels_|.
|
|
actives->clear();
|
|
for (const auto& ch : receive_channels_) {
|
|
int level = GetOutputLevel(ch.second->channel());
|
|
if (level > 0) {
|
|
actives->push_back(std::make_pair(ch.first, level));
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
int WebRtcVoiceMediaChannel::GetOutputLevel() {
|
|
// return the highest output level of all streams
|
|
int highest = GetOutputLevel(voe_channel());
|
|
for (const auto& ch : receive_channels_) {
|
|
int level = GetOutputLevel(ch.second->channel());
|
|
highest = std::max(level, highest);
|
|
}
|
|
return highest;
|
|
}
|
|
|
|
int WebRtcVoiceMediaChannel::GetTimeSinceLastTyping() {
|
|
int ret;
|
|
if (engine()->voe()->processing()->TimeSinceLastTyping(ret) == -1) {
|
|
// In case of error, log the info and continue
|
|
LOG_RTCERR0(TimeSinceLastTyping);
|
|
ret = -1;
|
|
} else {
|
|
ret *= 1000; // We return ms, webrtc returns seconds.
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window,
|
|
int cost_per_typing, int reporting_threshold, int penalty_decay,
|
|
int type_event_delay) {
|
|
if (engine()->voe()->processing()->SetTypingDetectionParameters(
|
|
time_window, cost_per_typing,
|
|
reporting_threshold, penalty_decay, type_event_delay) == -1) {
|
|
// In case of error, log the info and continue
|
|
LOG_RTCERR5(SetTypingDetectionParameters, time_window,
|
|
cost_per_typing, reporting_threshold, penalty_decay,
|
|
type_event_delay);
|
|
}
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetOutputScaling(
|
|
uint32 ssrc, double left, double right) {
|
|
rtc::CritScope lock(&receive_channels_cs_);
|
|
// Collect the channels to scale the output volume.
|
|
std::vector<int> channels;
|
|
if (0 == ssrc) { // Collect all channels, including the default one.
|
|
// Default channel is not in receive_channels_ if it is not being used for
|
|
// playout.
|
|
if (default_receive_ssrc_ == 0)
|
|
channels.push_back(voe_channel());
|
|
for (const auto& ch : receive_channels_) {
|
|
channels.push_back(ch.second->channel());
|
|
}
|
|
} else { // Collect only the channel of the specified ssrc.
|
|
int channel = GetReceiveChannelNum(ssrc);
|
|
if (-1 == channel) {
|
|
LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
|
|
return false;
|
|
}
|
|
channels.push_back(channel);
|
|
}
|
|
|
|
// Scale the output volume for the collected channels. We first normalize to
|
|
// scale the volume and then set the left and right pan.
|
|
float scale = static_cast<float>(std::max(left, right));
|
|
if (scale > 0.0001f) {
|
|
left /= scale;
|
|
right /= scale;
|
|
}
|
|
for (int ch_id : channels) {
|
|
if (-1 == engine()->voe()->volume()->SetChannelOutputVolumeScaling(
|
|
ch_id, scale)) {
|
|
LOG_RTCERR2(SetChannelOutputVolumeScaling, ch_id, scale);
|
|
return false;
|
|
}
|
|
if (-1 == engine()->voe()->volume()->SetOutputVolumePan(
|
|
ch_id, static_cast<float>(left), static_cast<float>(right))) {
|
|
LOG_RTCERR3(SetOutputVolumePan, ch_id, left, right);
|
|
// Do not return if fails. SetOutputVolumePan is not available for all
|
|
// pltforms.
|
|
}
|
|
LOG(LS_INFO) << "SetOutputScaling to left=" << left * scale
|
|
<< " right=" << right * scale
|
|
<< " for channel " << ch_id << " and ssrc " << ssrc;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetRingbackTone(const char *buf, int len) {
|
|
ringback_tone_.reset(new WebRtcSoundclipStream(buf, len));
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::PlayRingbackTone(uint32 ssrc,
|
|
bool play, bool loop) {
|
|
if (!ringback_tone_) {
|
|
return false;
|
|
}
|
|
|
|
// The voe file api is not available in chrome.
|
|
if (!engine()->voe()->file()) {
|
|
return false;
|
|
}
|
|
|
|
// Determine which VoiceEngine channel to play on.
|
|
int channel = (ssrc == 0) ? voe_channel() : GetReceiveChannelNum(ssrc);
|
|
if (channel == -1) {
|
|
return false;
|
|
}
|
|
|
|
// Make sure the ringtone is cued properly, and play it out.
|
|
if (play) {
|
|
ringback_tone_->set_loop(loop);
|
|
ringback_tone_->Rewind();
|
|
if (engine()->voe()->file()->StartPlayingFileLocally(channel,
|
|
ringback_tone_.get()) == -1) {
|
|
LOG_RTCERR2(StartPlayingFileLocally, channel, ringback_tone_.get());
|
|
LOG(LS_ERROR) << "Unable to start ringback tone";
|
|
return false;
|
|
}
|
|
ringback_channels_.insert(channel);
|
|
LOG(LS_INFO) << "Started ringback on channel " << channel;
|
|
} else {
|
|
if (engine()->voe()->file()->IsPlayingFileLocally(channel) == 1 &&
|
|
engine()->voe()->file()->StopPlayingFileLocally(channel) == -1) {
|
|
LOG_RTCERR1(StopPlayingFileLocally, channel);
|
|
return false;
|
|
}
|
|
LOG(LS_INFO) << "Stopped ringback on channel " << channel;
|
|
ringback_channels_.erase(channel);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
|
|
return dtmf_allowed_;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event,
|
|
int duration, int flags) {
|
|
if (!dtmf_allowed_) {
|
|
return false;
|
|
}
|
|
|
|
// Send the event.
|
|
if (flags & cricket::DF_SEND) {
|
|
int channel = -1;
|
|
if (ssrc == 0) {
|
|
bool default_channel_is_inuse = false;
|
|
for (const auto& ch : send_channels_) {
|
|
if (IsDefaultChannel(ch.second->channel())) {
|
|
default_channel_is_inuse = true;
|
|
break;
|
|
}
|
|
}
|
|
if (default_channel_is_inuse) {
|
|
channel = voe_channel();
|
|
} else if (!send_channels_.empty()) {
|
|
channel = send_channels_.begin()->second->channel();
|
|
}
|
|
} else {
|
|
channel = GetSendChannelNum(ssrc);
|
|
}
|
|
if (channel == -1) {
|
|
LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
|
|
<< ssrc << " is not in use.";
|
|
return false;
|
|
}
|
|
// Send DTMF using out-of-band DTMF. ("true", as 3rd arg)
|
|
if (engine()->voe()->dtmf()->SendTelephoneEvent(
|
|
channel, event, true, duration) == -1) {
|
|
LOG_RTCERR4(SendTelephoneEvent, channel, event, true, duration);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
// Play the event.
|
|
if (flags & cricket::DF_PLAY) {
|
|
// Play DTMF tone locally.
|
|
if (engine()->voe()->dtmf()->PlayDtmfTone(event, duration) == -1) {
|
|
LOG_RTCERR2(PlayDtmfTone, event, duration);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::OnPacketReceived(
|
|
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
|
|
DCHECK(thread_checker_.CalledOnValidThread());
|
|
|
|
// Forward packet to Call as well.
|
|
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
|
|
packet_time.not_before);
|
|
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
|
|
reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
|
|
webrtc_packet_time);
|
|
|
|
// Pick which channel to send this packet to. If this packet doesn't match
|
|
// any multiplexed streams, just send it to the default channel. Otherwise,
|
|
// send it to the specific decoder instance for that stream.
|
|
int which_channel =
|
|
GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), false));
|
|
if (which_channel == -1) {
|
|
which_channel = voe_channel();
|
|
}
|
|
|
|
// Stop any ringback that might be playing on the channel.
|
|
// It's possible the ringback has already stopped, ih which case we'll just
|
|
// use the opportunity to remove the channel from ringback_channels_.
|
|
if (engine()->voe()->file()) {
|
|
const std::set<int>::iterator it = ringback_channels_.find(which_channel);
|
|
if (it != ringback_channels_.end()) {
|
|
if (engine()->voe()->file()->IsPlayingFileLocally(
|
|
which_channel) == 1) {
|
|
engine()->voe()->file()->StopPlayingFileLocally(which_channel);
|
|
LOG(LS_INFO) << "Stopped ringback on channel " << which_channel
|
|
<< " due to incoming media";
|
|
}
|
|
ringback_channels_.erase(which_channel);
|
|
}
|
|
}
|
|
|
|
// Pass it off to the decoder.
|
|
engine()->voe()->network()->ReceivedRTPPacket(
|
|
which_channel, packet->data(), packet->size(),
|
|
webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::OnRtcpReceived(
|
|
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
|
|
DCHECK(thread_checker_.CalledOnValidThread());
|
|
|
|
// Forward packet to Call as well.
|
|
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
|
|
packet_time.not_before);
|
|
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
|
|
reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
|
|
webrtc_packet_time);
|
|
|
|
// Sending channels need all RTCP packets with feedback information.
|
|
// Even sender reports can contain attached report blocks.
|
|
// Receiving channels need sender reports in order to create
|
|
// correct receiver reports.
|
|
int type = 0;
|
|
if (!GetRtcpType(packet->data(), packet->size(), &type)) {
|
|
LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
|
|
return;
|
|
}
|
|
|
|
// If it is a sender report, find the channel that is listening.
|
|
bool has_sent_to_default_channel = false;
|
|
if (type == kRtcpTypeSR) {
|
|
int which_channel =
|
|
GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), true));
|
|
if (which_channel != -1) {
|
|
engine()->voe()->network()->ReceivedRTCPPacket(
|
|
which_channel, packet->data(), packet->size());
|
|
|
|
if (IsDefaultChannel(which_channel))
|
|
has_sent_to_default_channel = true;
|
|
}
|
|
}
|
|
|
|
// SR may continue RR and any RR entry may correspond to any one of the send
|
|
// channels. So all RTCP packets must be forwarded all send channels. VoE
|
|
// will filter out RR internally.
|
|
for (const auto& ch : send_channels_) {
|
|
// Make sure not sending the same packet to default channel more than once.
|
|
if (IsDefaultChannel(ch.second->channel()) &&
|
|
has_sent_to_default_channel)
|
|
continue;
|
|
|
|
engine()->voe()->network()->ReceivedRTCPPacket(
|
|
ch.second->channel(), packet->data(), packet->size());
|
|
}
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
|
|
int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
|
|
if (channel == -1) {
|
|
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
|
|
return false;
|
|
}
|
|
if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) {
|
|
LOG_RTCERR2(SetInputMute, channel, muted);
|
|
return false;
|
|
}
|
|
// We set the AGC to mute state only when all the channels are muted.
|
|
// This implementation is not ideal, instead we should signal the AGC when
|
|
// the mic channel is muted/unmuted. We can't do it today because there
|
|
// is no good way to know which stream is mapping to the mic channel.
|
|
bool all_muted = muted;
|
|
for (const auto& ch : send_channels_) {
|
|
if (!all_muted) {
|
|
break;
|
|
}
|
|
if (engine()->voe()->volume()->GetInputMute(ch.second->channel(),
|
|
all_muted)) {
|
|
LOG_RTCERR1(GetInputMute, ch.second->channel());
|
|
return false;
|
|
}
|
|
}
|
|
|
|
webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
|
|
if (ap)
|
|
ap->set_output_will_be_muted(all_muted);
|
|
return true;
|
|
}
|
|
|
|
// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
|
|
// SetMaxSendBitrate() in future.
|
|
bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
|
|
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
|
|
|
|
return SetSendBitrateInternal(bps);
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
|
|
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
|
|
|
|
send_bitrate_setting_ = true;
|
|
send_bitrate_bps_ = bps;
|
|
|
|
if (!send_codec_) {
|
|
LOG(LS_INFO) << "The send codec has not been set up yet. "
|
|
<< "The send bitrate setting will be applied later.";
|
|
return true;
|
|
}
|
|
|
|
// 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;
|
|
|
|
webrtc::CodecInst codec = *send_codec_;
|
|
bool is_multi_rate = IsCodecMultiRate(codec);
|
|
|
|
if (is_multi_rate) {
|
|
// If codec is multi-rate then just set the bitrate.
|
|
codec.rate = bps;
|
|
if (!SetSendCodec(codec)) {
|
|
LOG(LS_INFO) << "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_INFO) << "Failed to set codec " << codec.plname
|
|
<< " to bitrate " << bps << " bps"
|
|
<< ", requires at least " << codec.rate << " bps.";
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
|
|
bool echo_metrics_on = false;
|
|
// These can take on valid negative values, so use the lowest possible level
|
|
// as default rather than -1.
|
|
int echo_return_loss = -100;
|
|
int echo_return_loss_enhancement = -100;
|
|
// These can also be negative, but in practice -1 is only used to signal
|
|
// insufficient data, since the resolution is limited to multiples of 4 ms.
|
|
int echo_delay_median_ms = -1;
|
|
int echo_delay_std_ms = -1;
|
|
if (engine()->voe()->processing()->GetEcMetricsStatus(
|
|
echo_metrics_on) != -1 && echo_metrics_on) {
|
|
// TODO(ajm): we may want to use VoECallReport::GetEchoMetricsSummary
|
|
// here, but it appears to be unsuitable currently. Revisit after this is
|
|
// investigated: http://b/issue?id=5666755
|
|
int erl, erle, rerl, anlp;
|
|
if (engine()->voe()->processing()->GetEchoMetrics(
|
|
erl, erle, rerl, anlp) != -1) {
|
|
echo_return_loss = erl;
|
|
echo_return_loss_enhancement = erle;
|
|
}
|
|
|
|
int median, std;
|
|
float dummy;
|
|
if (engine()->voe()->processing()->GetEcDelayMetrics(
|
|
median, std, dummy) != -1) {
|
|
echo_delay_median_ms = median;
|
|
echo_delay_std_ms = std;
|
|
}
|
|
}
|
|
|
|
webrtc::CallStatistics cs;
|
|
unsigned int ssrc;
|
|
webrtc::CodecInst codec;
|
|
unsigned int level;
|
|
|
|
for (const auto& ch : send_channels_) {
|
|
const int channel = ch.second->channel();
|
|
|
|
// Fill in the sender info, based on what we know, and what the
|
|
// remote side told us it got from its RTCP report.
|
|
VoiceSenderInfo sinfo;
|
|
|
|
if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 ||
|
|
engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) {
|
|
continue;
|
|
}
|
|
|
|
sinfo.add_ssrc(ssrc);
|
|
sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
|
|
sinfo.bytes_sent = cs.bytesSent;
|
|
sinfo.packets_sent = cs.packetsSent;
|
|
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
|
|
// returns 0 to indicate an error value.
|
|
sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1;
|
|
|
|
// Get data from the last remote RTCP report. Use default values if no data
|
|
// available.
|
|
sinfo.fraction_lost = -1.0;
|
|
sinfo.jitter_ms = -1;
|
|
sinfo.packets_lost = -1;
|
|
sinfo.ext_seqnum = -1;
|
|
std::vector<webrtc::ReportBlock> receive_blocks;
|
|
if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
|
|
channel, &receive_blocks) != -1 &&
|
|
engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) {
|
|
for (const webrtc::ReportBlock& block : receive_blocks) {
|
|
// Lookup report for send ssrc only.
|
|
if (block.source_SSRC == sinfo.ssrc()) {
|
|
// Convert Q8 to floating point.
|
|
sinfo.fraction_lost = static_cast<float>(block.fraction_lost) / 256;
|
|
// Convert samples to milliseconds.
|
|
if (codec.plfreq / 1000 > 0) {
|
|
sinfo.jitter_ms = block.interarrival_jitter / (codec.plfreq / 1000);
|
|
}
|
|
sinfo.packets_lost = block.cumulative_num_packets_lost;
|
|
sinfo.ext_seqnum = block.extended_highest_sequence_number;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Local speech level.
|
|
sinfo.audio_level = (engine()->voe()->volume()->
|
|
GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
|
|
|
|
// TODO(xians): We are injecting the same APM logging to all the send
|
|
// channels here because there is no good way to know which send channel
|
|
// is using the APM. The correct fix is to allow the send channels to have
|
|
// their own APM so that we can feed the correct APM logging to different
|
|
// send channels. See issue crbug/264611 .
|
|
sinfo.echo_return_loss = echo_return_loss;
|
|
sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement;
|
|
sinfo.echo_delay_median_ms = echo_delay_median_ms;
|
|
sinfo.echo_delay_std_ms = echo_delay_std_ms;
|
|
// TODO(ajm): Re-enable this metric once we have a reliable implementation.
|
|
sinfo.aec_quality_min = -1;
|
|
sinfo.typing_noise_detected = typing_noise_detected_;
|
|
|
|
info->senders.push_back(sinfo);
|
|
}
|
|
|
|
// Build the list of receivers, one for each receiving channel, or 1 in
|
|
// a 1:1 call.
|
|
std::vector<int> channels;
|
|
for (const auto& ch : receive_channels_) {
|
|
channels.push_back(ch.second->channel());
|
|
}
|
|
if (channels.empty()) {
|
|
channels.push_back(voe_channel());
|
|
}
|
|
|
|
// Get the SSRC and stats for each receiver, based on our own calculations.
|
|
for (int ch_id : channels) {
|
|
memset(&cs, 0, sizeof(cs));
|
|
if (engine()->voe()->rtp()->GetRemoteSSRC(ch_id, ssrc) != -1 &&
|
|
engine()->voe()->rtp()->GetRTCPStatistics(ch_id, cs) != -1 &&
|
|
engine()->voe()->codec()->GetRecCodec(ch_id, codec) != -1) {
|
|
VoiceReceiverInfo rinfo;
|
|
rinfo.add_ssrc(ssrc);
|
|
rinfo.bytes_rcvd = cs.bytesReceived;
|
|
rinfo.packets_rcvd = cs.packetsReceived;
|
|
// The next four fields are from the most recently sent RTCP report.
|
|
// Convert Q8 to floating point.
|
|
rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
|
|
rinfo.packets_lost = cs.cumulativeLost;
|
|
rinfo.ext_seqnum = cs.extendedMax;
|
|
rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
|
|
if (codec.pltype != -1) {
|
|
rinfo.codec_name = codec.plname;
|
|
}
|
|
// Convert samples to milliseconds.
|
|
if (codec.plfreq / 1000 > 0) {
|
|
rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000);
|
|
}
|
|
|
|
// Get jitter buffer and total delay (alg + jitter + playout) stats.
|
|
webrtc::NetworkStatistics ns;
|
|
if (engine()->voe()->neteq() &&
|
|
engine()->voe()->neteq()->GetNetworkStatistics(
|
|
ch_id, ns) != -1) {
|
|
rinfo.jitter_buffer_ms = ns.currentBufferSize;
|
|
rinfo.jitter_buffer_preferred_ms = ns.preferredBufferSize;
|
|
rinfo.expand_rate =
|
|
static_cast<float>(ns.currentExpandRate) / (1 << 14);
|
|
rinfo.speech_expand_rate =
|
|
static_cast<float>(ns.currentSpeechExpandRate) / (1 << 14);
|
|
rinfo.secondary_decoded_rate =
|
|
static_cast<float>(ns.currentSecondaryDecodedRate) / (1 << 14);
|
|
rinfo.accelerate_rate =
|
|
static_cast<float>(ns.currentAccelerateRate) / (1 << 14);
|
|
rinfo.preemptive_expand_rate =
|
|
static_cast<float>(ns.currentPreemptiveRate) / (1 << 14);
|
|
}
|
|
|
|
webrtc::AudioDecodingCallStats ds;
|
|
if (engine()->voe()->neteq() &&
|
|
engine()->voe()->neteq()->GetDecodingCallStatistics(
|
|
ch_id, &ds) != -1) {
|
|
rinfo.decoding_calls_to_silence_generator =
|
|
ds.calls_to_silence_generator;
|
|
rinfo.decoding_calls_to_neteq = ds.calls_to_neteq;
|
|
rinfo.decoding_normal = ds.decoded_normal;
|
|
rinfo.decoding_plc = ds.decoded_plc;
|
|
rinfo.decoding_cng = ds.decoded_cng;
|
|
rinfo.decoding_plc_cng = ds.decoded_plc_cng;
|
|
}
|
|
|
|
if (engine()->voe()->sync()) {
|
|
int jitter_buffer_delay_ms = 0;
|
|
int playout_buffer_delay_ms = 0;
|
|
engine()->voe()->sync()->GetDelayEstimate(
|
|
ch_id, &jitter_buffer_delay_ms, &playout_buffer_delay_ms);
|
|
rinfo.delay_estimate_ms = jitter_buffer_delay_ms +
|
|
playout_buffer_delay_ms;
|
|
}
|
|
|
|
// Get speech level.
|
|
rinfo.audio_level = (engine()->voe()->volume()->
|
|
GetSpeechOutputLevelFullRange(ch_id, level) != -1) ? level : -1;
|
|
info->receivers.push_back(rinfo);
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::GetLastMediaError(
|
|
uint32* ssrc, VoiceMediaChannel::Error* error) {
|
|
DCHECK(ssrc != NULL);
|
|
DCHECK(error != NULL);
|
|
FindSsrc(voe_channel(), ssrc);
|
|
*error = WebRtcErrorToChannelError(GetLastEngineError());
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
|
|
rtc::CritScope lock(&receive_channels_cs_);
|
|
DCHECK(ssrc != NULL);
|
|
if (channel_num == -1 && send_ != SEND_NOTHING) {
|
|
// Sometimes the VoiceEngine core will throw error with channel_num = -1.
|
|
// This means the error is not limited to a specific channel. Signal the
|
|
// message using ssrc=0. If the current channel is sending, use this
|
|
// channel for sending the message.
|
|
*ssrc = 0;
|
|
return true;
|
|
} else {
|
|
// Check whether this is a sending channel.
|
|
for (const auto& ch : send_channels_) {
|
|
if (ch.second->channel() == channel_num) {
|
|
// This is a sending channel.
|
|
uint32 local_ssrc = 0;
|
|
if (engine()->voe()->rtp()->GetLocalSSRC(
|
|
channel_num, local_ssrc) != -1) {
|
|
*ssrc = local_ssrc;
|
|
}
|
|
return true;
|
|
}
|
|
}
|
|
|
|
// Check whether this is a receiving channel.
|
|
for (const auto& ch : receive_channels_) {
|
|
if (ch.second->channel() == channel_num) {
|
|
*ssrc = ch.first;
|
|
return true;
|
|
}
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
|
|
if (error == VE_TYPING_NOISE_WARNING) {
|
|
typing_noise_detected_ = true;
|
|
} else if (error == VE_TYPING_NOISE_OFF_WARNING) {
|
|
typing_noise_detected_ = false;
|
|
}
|
|
SignalMediaError(ssrc, WebRtcErrorToChannelError(error));
|
|
}
|
|
|
|
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
|
|
unsigned int ulevel;
|
|
int ret =
|
|
engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
|
|
return (ret == 0) ? static_cast<int>(ulevel) : -1;
|
|
}
|
|
|
|
int WebRtcVoiceMediaChannel::GetReceiveChannelNum(uint32 ssrc) const {
|
|
ChannelMap::const_iterator it = receive_channels_.find(ssrc);
|
|
if (it != receive_channels_.end())
|
|
return it->second->channel();
|
|
return (ssrc == default_receive_ssrc_) ? voe_channel() : -1;
|
|
}
|
|
|
|
int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) const {
|
|
ChannelMap::const_iterator it = send_channels_.find(ssrc);
|
|
if (it != send_channels_.end())
|
|
return it->second->channel();
|
|
|
|
return -1;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec,
|
|
const std::vector<AudioCodec>& all_codecs, webrtc::CodecInst* send_codec) {
|
|
// Get the RED encodings from the parameter with no name. This may
|
|
// change based on what is discussed on the Jingle list.
|
|
// The encoding parameter is of the form "a/b"; we only support where
|
|
// a == b. Verify this and parse out the value into red_pt.
|
|
// If the parameter value is absent (as it will be until we wire up the
|
|
// signaling of this message), use the second codec specified (i.e. the
|
|
// one after "red") as the encoding parameter.
|
|
int red_pt = -1;
|
|
std::string red_params;
|
|
CodecParameterMap::const_iterator it = red_codec.params.find("");
|
|
if (it != red_codec.params.end()) {
|
|
red_params = it->second;
|
|
std::vector<std::string> red_pts;
|
|
if (rtc::split(red_params, '/', &red_pts) != 2 ||
|
|
red_pts[0] != red_pts[1] ||
|
|
!rtc::FromString(red_pts[0], &red_pt)) {
|
|
LOG(LS_WARNING) << "RED params " << red_params << " not supported.";
|
|
return false;
|
|
}
|
|
} else if (red_codec.params.empty()) {
|
|
LOG(LS_WARNING) << "RED params not present, using defaults";
|
|
if (all_codecs.size() > 1) {
|
|
red_pt = all_codecs[1].id;
|
|
}
|
|
}
|
|
|
|
// Try to find red_pt in |codecs|.
|
|
for (const AudioCodec& codec : all_codecs) {
|
|
if (codec.id == red_pt) {
|
|
// If we find the right codec, that will be the codec we pass to
|
|
// SetSendCodec, with the desired payload type.
|
|
if (engine()->FindWebRtcCodec(codec, send_codec)) {
|
|
return true;
|
|
} else {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
LOG(LS_WARNING) << "RED params " << red_params << " are invalid.";
|
|
return false;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::EnableRtcp(int channel) {
|
|
if (engine()->voe()->rtp()->SetRTCPStatus(channel, true) == -1) {
|
|
LOG_RTCERR2(SetRTCPStatus, channel, 1);
|
|
return false;
|
|
}
|
|
// TODO(juberti): Enable VQMon and RTCP XR reports, once we know what
|
|
// what we want to do with them.
|
|
// engine()->voe().EnableVQMon(voe_channel(), true);
|
|
// engine()->voe().EnableRTCP_XR(voe_channel(), true);
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::ResetRecvCodecs(int channel) {
|
|
int ncodecs = engine()->voe()->codec()->NumOfCodecs();
|
|
for (int i = 0; i < ncodecs; ++i) {
|
|
webrtc::CodecInst voe_codec;
|
|
if (engine()->voe()->codec()->GetCodec(i, voe_codec) != -1) {
|
|
voe_codec.pltype = -1;
|
|
if (engine()->voe()->codec()->SetRecPayloadType(
|
|
channel, voe_codec) == -1) {
|
|
LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) {
|
|
if (playout) {
|
|
LOG(LS_INFO) << "Starting playout for channel #" << channel;
|
|
if (engine()->voe()->base()->StartPlayout(channel) == -1) {
|
|
LOG_RTCERR1(StartPlayout, channel);
|
|
return false;
|
|
}
|
|
} else {
|
|
LOG(LS_INFO) << "Stopping playout for channel #" << channel;
|
|
engine()->voe()->base()->StopPlayout(channel);
|
|
}
|
|
return true;
|
|
}
|
|
|
|
uint32 WebRtcVoiceMediaChannel::ParseSsrc(const void* data, size_t len,
|
|
bool rtcp) {
|
|
size_t ssrc_pos = (!rtcp) ? 8 : 4;
|
|
uint32 ssrc = 0;
|
|
if (len >= (ssrc_pos + sizeof(ssrc))) {
|
|
ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
|
|
}
|
|
return ssrc;
|
|
}
|
|
|
|
// Convert VoiceEngine error code into VoiceMediaChannel::Error enum.
|
|
VoiceMediaChannel::Error
|
|
WebRtcVoiceMediaChannel::WebRtcErrorToChannelError(int err_code) {
|
|
switch (err_code) {
|
|
case 0:
|
|
return ERROR_NONE;
|
|
case VE_CANNOT_START_RECORDING:
|
|
case VE_MIC_VOL_ERROR:
|
|
case VE_GET_MIC_VOL_ERROR:
|
|
case VE_CANNOT_ACCESS_MIC_VOL:
|
|
return ERROR_REC_DEVICE_OPEN_FAILED;
|
|
case VE_SATURATION_WARNING:
|
|
return ERROR_REC_DEVICE_SATURATION;
|
|
case VE_REC_DEVICE_REMOVED:
|
|
return ERROR_REC_DEVICE_REMOVED;
|
|
case VE_RUNTIME_REC_WARNING:
|
|
case VE_RUNTIME_REC_ERROR:
|
|
return ERROR_REC_RUNTIME_ERROR;
|
|
case VE_CANNOT_START_PLAYOUT:
|
|
case VE_SPEAKER_VOL_ERROR:
|
|
case VE_GET_SPEAKER_VOL_ERROR:
|
|
case VE_CANNOT_ACCESS_SPEAKER_VOL:
|
|
return ERROR_PLAY_DEVICE_OPEN_FAILED;
|
|
case VE_RUNTIME_PLAY_WARNING:
|
|
case VE_RUNTIME_PLAY_ERROR:
|
|
return ERROR_PLAY_RUNTIME_ERROR;
|
|
case VE_TYPING_NOISE_WARNING:
|
|
return ERROR_REC_TYPING_NOISE_DETECTED;
|
|
default:
|
|
return VoiceMediaChannel::ERROR_OTHER;
|
|
}
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
|
|
int channel_id, const RtpHeaderExtension* extension) {
|
|
bool enable = false;
|
|
int id = 0;
|
|
std::string uri;
|
|
if (extension) {
|
|
enable = true;
|
|
id = extension->id;
|
|
uri = extension->uri;
|
|
}
|
|
if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) {
|
|
LOG_RTCERR4(*setter, uri, channel_id, enable, id);
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::RecreateAudioReceiveStreams() {
|
|
DCHECK(thread_checker_.CalledOnValidThread());
|
|
for (const auto& it : receive_channels_) {
|
|
RemoveAudioReceiveStream(it.first);
|
|
}
|
|
for (const auto& it : receive_channels_) {
|
|
AddAudioReceiveStream(it.first);
|
|
}
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32 ssrc) {
|
|
DCHECK(thread_checker_.CalledOnValidThread());
|
|
WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
|
|
DCHECK(channel != nullptr);
|
|
DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
|
|
webrtc::AudioReceiveStream::Config config;
|
|
config.rtp.remote_ssrc = ssrc;
|
|
// Only add RTP extensions if we support combined A/V BWE.
|
|
config.rtp.extensions = recv_rtp_extensions_;
|
|
config.combined_audio_video_bwe =
|
|
options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false);
|
|
config.voe_channel_id = channel->channel();
|
|
config.sync_group = receive_stream_params_[ssrc].sync_label;
|
|
webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
|
|
receive_streams_.insert(std::make_pair(ssrc, s));
|
|
}
|
|
|
|
void WebRtcVoiceMediaChannel::RemoveAudioReceiveStream(uint32 ssrc) {
|
|
DCHECK(thread_checker_.CalledOnValidThread());
|
|
auto stream_it = receive_streams_.find(ssrc);
|
|
if (stream_it != receive_streams_.end()) {
|
|
call_->DestroyAudioReceiveStream(stream_it->second);
|
|
receive_streams_.erase(stream_it);
|
|
}
|
|
}
|
|
|
|
bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal(
|
|
const std::vector<AudioCodec>& new_codecs) {
|
|
for (const AudioCodec& codec : new_codecs) {
|
|
webrtc::CodecInst voe_codec;
|
|
if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
|
|
LOG(LS_INFO) << ToString(codec);
|
|
voe_codec.pltype = codec.id;
|
|
if (default_receive_ssrc_ == 0) {
|
|
// Set the receive codecs on the default channel explicitly if the
|
|
// default channel is not used by |receive_channels_|, this happens in
|
|
// conference mode or in non-conference mode when there is no playout
|
|
// channel.
|
|
// TODO(xians): Figure out how we use the default channel in conference
|
|
// mode.
|
|
if (engine()->voe()->codec()->SetRecPayloadType(
|
|
voe_channel(), voe_codec) == -1) {
|
|
LOG_RTCERR2(SetRecPayloadType, voe_channel(), ToString(voe_codec));
|
|
return false;
|
|
}
|
|
}
|
|
|
|
// Set the receive codecs on all receiving channels.
|
|
for (const auto& ch : receive_channels_) {
|
|
if (engine()->voe()->codec()->SetRecPayloadType(
|
|
ch.second->channel(), voe_codec) == -1) {
|
|
LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
|
|
ToString(voe_codec));
|
|
return false;
|
|
}
|
|
}
|
|
} else {
|
|
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
int WebRtcSoundclipStream::Read(void *buf, size_t len) {
|
|
size_t res = 0;
|
|
mem_.Read(buf, len, &res, NULL);
|
|
return static_cast<int>(res);
|
|
}
|
|
|
|
int WebRtcSoundclipStream::Rewind() {
|
|
mem_.Rewind();
|
|
// Return -1 to keep VoiceEngine from looping.
|
|
return (loop_) ? 0 : -1;
|
|
}
|
|
|
|
} // namespace cricket
|
|
|
|
#endif // HAVE_WEBRTC_VOICE
|