| Index: talk/media/webrtc/webrtcvoiceengine.cc
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
|
| deleted file mode 100644
|
| index 150042300de48cd572d46854784ba21550602a31..0000000000000000000000000000000000000000
|
| --- a/talk/media/webrtc/webrtcvoiceengine.cc
|
| +++ /dev/null
|
| @@ -1,2579 +0,0 @@
|
| -/*
|
| - * libjingle
|
| - * Copyright 2004 Google Inc.
|
| - *
|
| - * Redistribution and use in source and binary forms, with or without
|
| - * modification, are permitted provided that the following conditions are met:
|
| - *
|
| - * 1. Redistributions of source code must retain the above copyright notice,
|
| - * this list of conditions and the following disclaimer.
|
| - * 2. Redistributions in binary form must reproduce the above copyright notice,
|
| - * this list of conditions and the following disclaimer in the documentation
|
| - * and/or other materials provided with the distribution.
|
| - * 3. The name of the author may not be used to endorse or promote products
|
| - * derived from this software without specific prior written permission.
|
| - *
|
| - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
| - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
| - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
| - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
| - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
| - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
| - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
| - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
| - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
| - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
| - */
|
| -
|
| -#ifdef HAVE_CONFIG_H
|
| -#include <config.h>
|
| -#endif
|
| -
|
| -#ifdef HAVE_WEBRTC_VOICE
|
| -
|
| -#include "talk/media/webrtc/webrtcvoiceengine.h"
|
| -
|
| -#include <algorithm>
|
| -#include <cstdio>
|
| -#include <string>
|
| -#include <vector>
|
| -
|
| -#include "talk/media/base/audioframe.h"
|
| -#include "talk/media/base/audiorenderer.h"
|
| -#include "talk/media/base/constants.h"
|
| -#include "talk/media/base/streamparams.h"
|
| -#include "talk/media/webrtc/webrtcmediaengine.h"
|
| -#include "talk/media/webrtc/webrtcvoe.h"
|
| -#include "webrtc/audio/audio_sink.h"
|
| -#include "webrtc/base/arraysize.h"
|
| -#include "webrtc/base/base64.h"
|
| -#include "webrtc/base/byteorder.h"
|
| -#include "webrtc/base/common.h"
|
| -#include "webrtc/base/helpers.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/base/stringencode.h"
|
| -#include "webrtc/base/stringutils.h"
|
| -#include "webrtc/call/rtc_event_log.h"
|
| -#include "webrtc/common.h"
|
| -#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
|
| -#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
| -#include "webrtc/system_wrappers/include/field_trial.h"
|
| -#include "webrtc/system_wrappers/include/trace.h"
|
| -
|
| -namespace cricket {
|
| -namespace {
|
| -
|
| -const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
|
| - webrtc::kTraceWarning | webrtc::kTraceError |
|
| - webrtc::kTraceCritical;
|
| -const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo |
|
| - webrtc::kTraceInfo;
|
| -
|
| -// On Windows Vista and newer, Microsoft introduced the concept of "Default
|
| -// Communications Device". This means that there are two types of default
|
| -// devices (old Wave Audio style default and Default Communications Device).
|
| -//
|
| -// On Windows systems which only support Wave Audio style default, uses either
|
| -// -1 or 0 to select the default device.
|
| -#ifdef WIN32
|
| -const int kDefaultAudioDeviceId = -1;
|
| -#else
|
| -const int kDefaultAudioDeviceId = 0;
|
| -#endif
|
| -
|
| -// Parameter used for NACK.
|
| -// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
|
| -const int kNackMaxPackets = 250;
|
| -
|
| -// Codec parameters for Opus.
|
| -// draft-spittka-payload-rtp-opus-03
|
| -
|
| -// Recommended bitrates:
|
| -// 8-12 kb/s for NB speech,
|
| -// 16-20 kb/s for WB speech,
|
| -// 28-40 kb/s for FB speech,
|
| -// 48-64 kb/s for FB mono music, and
|
| -// 64-128 kb/s for FB stereo music.
|
| -// The current implementation applies the following values to mono signals,
|
| -// and multiplies them by 2 for stereo.
|
| -const int kOpusBitrateNb = 12000;
|
| -const int kOpusBitrateWb = 20000;
|
| -const int kOpusBitrateFb = 32000;
|
| -
|
| -// Opus bitrate should be in the range between 6000 and 510000.
|
| -const int kOpusMinBitrate = 6000;
|
| -const int kOpusMaxBitrate = 510000;
|
| -
|
| -// Default audio dscp value.
|
| -// See http://tools.ietf.org/html/rfc2474 for details.
|
| -// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
|
| -const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
|
| -
|
| -// Ensure we open the file in a writeable path on ChromeOS and Android. This
|
| -// workaround can be removed when it's possible to specify a filename for audio
|
| -// option based AEC dumps.
|
| -//
|
| -// TODO(grunell): Use a string in the options instead of hardcoding it here
|
| -// and let the embedder choose the filename (crbug.com/264223).
|
| -//
|
| -// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
|
| -// below.
|
| -#if defined(CHROMEOS)
|
| -const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
|
| -#elif defined(ANDROID)
|
| -const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
|
| -#else
|
| -const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
|
| -#endif
|
| -
|
| -// Constants from voice_engine_defines.h.
|
| -const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1)
|
| -const int kMaxTelephoneEventCode = 255;
|
| -const int kMinTelephoneEventDuration = 100;
|
| -const int kMaxTelephoneEventDuration = 60000; // Actual limit is 2^16
|
| -
|
| -class ProxySink : public webrtc::AudioSinkInterface {
|
| - public:
|
| - ProxySink(AudioSinkInterface* sink) : sink_(sink) { RTC_DCHECK(sink); }
|
| -
|
| - void OnData(const Data& audio) override { sink_->OnData(audio); }
|
| -
|
| - private:
|
| - webrtc::AudioSinkInterface* sink_;
|
| -};
|
| -
|
| -bool ValidateStreamParams(const StreamParams& sp) {
|
| - if (sp.ssrcs.empty()) {
|
| - LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
|
| - return false;
|
| - }
|
| - if (sp.ssrcs.size() > 1) {
|
| - LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " << sp.ToString();
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -// Dumps an AudioCodec in RFC 2327-ish format.
|
| -std::string ToString(const AudioCodec& codec) {
|
| - std::stringstream ss;
|
| - ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
|
| - << " (" << codec.id << ")";
|
| - return ss.str();
|
| -}
|
| -
|
| -std::string ToString(const webrtc::CodecInst& codec) {
|
| - std::stringstream ss;
|
| - ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
|
| - << " (" << codec.pltype << ")";
|
| - return ss.str();
|
| -}
|
| -
|
| -bool IsCodec(const AudioCodec& codec, const char* ref_name) {
|
| - return (_stricmp(codec.name.c_str(), ref_name) == 0);
|
| -}
|
| -
|
| -bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
|
| - return (_stricmp(codec.plname, ref_name) == 0);
|
| -}
|
| -
|
| -bool FindCodec(const std::vector<AudioCodec>& codecs,
|
| - const AudioCodec& codec,
|
| - AudioCodec* found_codec) {
|
| - for (const AudioCodec& c : codecs) {
|
| - if (c.Matches(codec)) {
|
| - if (found_codec != NULL) {
|
| - *found_codec = c;
|
| - }
|
| - return true;
|
| - }
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) {
|
| - if (codecs.empty()) {
|
| - return true;
|
| - }
|
| - std::vector<int> payload_types;
|
| - for (const AudioCodec& codec : codecs) {
|
| - payload_types.push_back(codec.id);
|
| - }
|
| - std::sort(payload_types.begin(), payload_types.end());
|
| - auto it = std::unique(payload_types.begin(), payload_types.end());
|
| - return it == payload_types.end();
|
| -}
|
| -
|
| -// Return true if codec.params[feature] == "1", false otherwise.
|
| -bool IsCodecFeatureEnabled(const AudioCodec& codec, const char* feature) {
|
| - int value;
|
| - return codec.GetParam(feature, &value) && value == 1;
|
| -}
|
| -
|
| -// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
|
| -// otherwise. If the value (either from params or codec.bitrate) <=0, use the
|
| -// default configuration. If the value is beyond feasible bit rate of Opus,
|
| -// clamp it. Returns the Opus bit rate for operation.
|
| -int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
|
| - int bitrate = 0;
|
| - bool use_param = true;
|
| - if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
|
| - bitrate = codec.bitrate;
|
| - use_param = false;
|
| - }
|
| - if (bitrate <= 0) {
|
| - if (max_playback_rate <= 8000) {
|
| - bitrate = kOpusBitrateNb;
|
| - } else if (max_playback_rate <= 16000) {
|
| - bitrate = kOpusBitrateWb;
|
| - } else {
|
| - bitrate = kOpusBitrateFb;
|
| - }
|
| -
|
| - if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
|
| - bitrate *= 2;
|
| - }
|
| - } else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
|
| - bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
|
| - std::string rate_source =
|
| - use_param ? "Codec parameter \"maxaveragebitrate\"" :
|
| - "Supplied Opus bitrate";
|
| - LOG(LS_WARNING) << rate_source
|
| - << " is invalid and is replaced by: "
|
| - << bitrate;
|
| - }
|
| - return bitrate;
|
| -}
|
| -
|
| -// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
|
| -// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
|
| -int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
|
| - int value;
|
| - if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
|
| - return value;
|
| - }
|
| - return kOpusDefaultMaxPlaybackRate;
|
| -}
|
| -
|
| -void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
|
| - bool* enable_codec_fec, int* max_playback_rate,
|
| - bool* enable_codec_dtx) {
|
| - *enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
|
| - *enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
|
| - *max_playback_rate = GetOpusMaxPlaybackRate(codec);
|
| -
|
| - // If OPUS, change what we send according to the "stereo" codec
|
| - // parameter, and not the "channels" parameter. We set
|
| - // voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
|
| - // the bitrate is not specified, i.e. is <= zero, we set it to the
|
| - // appropriate default value for mono or stereo Opus.
|
| -
|
| - voe_codec->channels = IsCodecFeatureEnabled(codec, kCodecParamStereo) ? 2 : 1;
|
| - voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
|
| -}
|
| -
|
| -webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
|
| - webrtc::AudioState::Config config;
|
| - config.voice_engine = voe_wrapper->engine();
|
| - return config;
|
| -}
|
| -
|
| -class WebRtcVoiceCodecs final {
|
| - public:
|
| - // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
|
| - // list and add a test which verifies VoE supports the listed codecs.
|
| - static std::vector<AudioCodec> SupportedCodecs() {
|
| - LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
|
| - std::vector<AudioCodec> result;
|
| - for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
|
| - // Change the sample rate of G722 to 8000 to match SDP.
|
| - MaybeFixupG722(&voe_codec, 8000);
|
| - // Skip uncompressed formats.
|
| - if (IsCodec(voe_codec, kL16CodecName)) {
|
| - continue;
|
| - }
|
| -
|
| - const CodecPref* pref = NULL;
|
| - for (size_t j = 0; j < arraysize(kCodecPrefs); ++j) {
|
| - if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
|
| - kCodecPrefs[j].clockrate == voe_codec.plfreq &&
|
| - kCodecPrefs[j].channels == voe_codec.channels) {
|
| - pref = &kCodecPrefs[j];
|
| - break;
|
| - }
|
| - }
|
| -
|
| - 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,
|
| - static_cast<int>(arraysize(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);
|
| - codec.AddFeedbackParam(
|
| - FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
|
| -
|
| - // TODO(hellner): Add ptime, sprop-stereo, and stereo
|
| - // when they can be set to values other than the default.
|
| - }
|
| - result.push_back(codec);
|
| - } else {
|
| - LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec);
|
| - }
|
| - }
|
| - // Make sure they are in local preference order.
|
| - std::sort(result.begin(), result.end(), &AudioCodec::Preferable);
|
| - return result;
|
| - }
|
| -
|
| - static bool ToCodecInst(const AudioCodec& in,
|
| - webrtc::CodecInst* out) {
|
| - for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
|
| - // Change the sample rate of G722 to 8000 to match SDP.
|
| - MaybeFixupG722(&voe_codec, 8000);
|
| - 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;
|
| - }
|
| -
|
| - static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
|
| - for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
|
| - if (IsCodec(codec, kCodecPrefs[i].name) &&
|
| - kCodecPrefs[i].clockrate == codec.plfreq) {
|
| - return kCodecPrefs[i].is_multi_rate;
|
| - }
|
| - }
|
| - return false;
|
| - }
|
| -
|
| - // If the AudioCodec param kCodecParamPTime is set, then we will set it to
|
| - // codec pacsize if it's valid, or we will pick the next smallest value we
|
| - // support.
|
| - // TODO(Brave): Query supported packet sizes from ACM when the API is ready.
|
| - static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
|
| - for (const CodecPref& codec_pref : kCodecPrefs) {
|
| - if ((IsCodec(*codec, codec_pref.name) &&
|
| - codec_pref.clockrate == codec->plfreq) ||
|
| - IsCodec(*codec, kG722CodecName)) {
|
| - int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
|
| - if (packet_size_ms) {
|
| - // Convert unit from milli-seconds to samples.
|
| - codec->pacsize = (codec->plfreq / 1000) * packet_size_ms;
|
| - return true;
|
| - }
|
| - }
|
| - }
|
| - return false;
|
| - }
|
| -
|
| - static const AudioCodec* GetPreferredCodec(
|
| - const std::vector<AudioCodec>& codecs,
|
| - webrtc::CodecInst* voe_codec,
|
| - int* red_payload_type) {
|
| - RTC_DCHECK(voe_codec);
|
| - RTC_DCHECK(red_payload_type);
|
| - // Select the preferred send codec (the first non-telephone-event/CN codec).
|
| - for (const AudioCodec& codec : codecs) {
|
| - *red_payload_type = -1;
|
| - 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.
|
| - const AudioCodec* found_codec = &codec;
|
| - if (IsCodec(*found_codec, kRedCodecName)) {
|
| - // Parse out the RED parameters. If we fail, just ignore RED;
|
| - // we don't support all possible params/usage scenarios.
|
| - *red_payload_type = codec.id;
|
| - found_codec = GetRedSendCodec(*found_codec, codecs);
|
| - if (!found_codec) {
|
| - continue;
|
| - }
|
| - }
|
| - // Ignore codecs we don't know about. The negotiation step should prevent
|
| - // this, but double-check to be sure.
|
| - if (!ToCodecInst(*found_codec, voe_codec)) {
|
| - LOG(LS_WARNING) << "Unknown codec " << ToString(*found_codec);
|
| - continue;
|
| - }
|
| - return found_codec;
|
| - }
|
| - return nullptr;
|
| - }
|
| -
|
| - private:
|
| - static const int kMaxNumPacketSize = 6;
|
| - struct CodecPref {
|
| - const char* name;
|
| - int clockrate;
|
| - size_t channels;
|
| - int payload_type;
|
| - bool is_multi_rate;
|
| - int packet_sizes_ms[kMaxNumPacketSize];
|
| - };
|
| - // Note: keep the supported packet sizes in ascending order.
|
| - static const CodecPref kCodecPrefs[12];
|
| -
|
| - static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
|
| - int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
|
| - for (int packet_size_ms : codec_pref.packet_sizes_ms) {
|
| - if (packet_size_ms && packet_size_ms <= ptime_ms) {
|
| - selected_packet_size_ms = packet_size_ms;
|
| - }
|
| - }
|
| - return selected_packet_size_ms;
|
| - }
|
| -
|
| - // Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
|
| - // which says that G722 should be advertised as 8 kHz although it is a 16 kHz
|
| - // codec.
|
| - static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
|
| - if (IsCodec(*voe_codec, kG722CodecName)) {
|
| - // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
|
| - // has changed, and this special case is no longer needed.
|
| - RTC_DCHECK(voe_codec->plfreq != new_plfreq);
|
| - voe_codec->plfreq = new_plfreq;
|
| - }
|
| - }
|
| -
|
| - static const AudioCodec* GetRedSendCodec(
|
| - const AudioCodec& red_codec,
|
| - const std::vector<AudioCodec>& all_codecs) {
|
| - // 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 nullptr;
|
| - }
|
| - } 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) {
|
| - return &codec;
|
| - }
|
| - }
|
| - LOG(LS_WARNING) << "RED params " << red_params << " are invalid.";
|
| - return nullptr;
|
| - }
|
| -};
|
| -
|
| -const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[12] = {
|
| - { kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
|
| - { kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
|
| - { kIsacCodecName, 32000, 1, 104, true, { 30 } },
|
| - // G722 should be advertised as 8000 Hz because of the RFC "bug".
|
| - { kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } },
|
| - { kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } },
|
| - { kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } },
|
| - { kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } },
|
| - { kCnCodecName, 32000, 1, 106, false, { } },
|
| - { kCnCodecName, 16000, 1, 105, false, { } },
|
| - { kCnCodecName, 8000, 1, 13, false, { } },
|
| - { kRedCodecName, 8000, 1, 127, false, { } },
|
| - { kDtmfCodecName, 8000, 1, 126, false, { } },
|
| -};
|
| -} // namespace {
|
| -
|
| -bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
|
| - webrtc::CodecInst* out) {
|
| - return WebRtcVoiceCodecs::ToCodecInst(in, out);
|
| -}
|
| -
|
| -WebRtcVoiceEngine::WebRtcVoiceEngine()
|
| - : voe_wrapper_(new VoEWrapper()),
|
| - audio_state_(webrtc::AudioState::Create(MakeAudioStateConfig(voe()))) {
|
| - Construct();
|
| -}
|
| -
|
| -WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper)
|
| - : voe_wrapper_(voe_wrapper) {
|
| - Construct();
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::Construct() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
|
| -
|
| - signal_thread_checker_.DetachFromThread();
|
| - std::memset(&default_agc_config_, 0, sizeof(default_agc_config_));
|
| - voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
|
| -
|
| - webrtc::Trace::set_level_filter(kDefaultTraceFilter);
|
| - webrtc::Trace::SetTraceCallback(this);
|
| -
|
| - // Load our audio codec list.
|
| - codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
|
| -}
|
| -
|
| -WebRtcVoiceEngine::~WebRtcVoiceEngine() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
|
| - if (adm_) {
|
| - voe_wrapper_.reset();
|
| - adm_->Release();
|
| - adm_ = NULL;
|
| - }
|
| - webrtc::Trace::SetTraceCallback(nullptr);
|
| -}
|
| -
|
| -bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_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() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - // Temporarily turn logging level up for the Init call
|
| - webrtc::Trace::set_level_filter(kElevatedTraceFilter);
|
| - LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
|
| - if (voe_wrapper_->base()->Init(adm_) == -1) {
|
| - LOG_RTCERR0_EX(Init, voe_wrapper_->error());
|
| - return false;
|
| - }
|
| - webrtc::Trace::set_level_filter(kDefaultTraceFilter);
|
| -
|
| - // Save the default AGC configuration settings. This must happen before
|
| - // calling ApplyOptions or the default will be overwritten.
|
| - if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
|
| - LOG_RTCERR0(GetAgcConfig);
|
| - return false;
|
| - }
|
| -
|
| - // Set default engine options.
|
| - {
|
| - AudioOptions options;
|
| - options.echo_cancellation = rtc::Optional<bool>(true);
|
| - options.auto_gain_control = rtc::Optional<bool>(true);
|
| - options.noise_suppression = rtc::Optional<bool>(true);
|
| - options.highpass_filter = rtc::Optional<bool>(true);
|
| - options.stereo_swapping = rtc::Optional<bool>(false);
|
| - options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
|
| - options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
|
| - options.typing_detection = rtc::Optional<bool>(true);
|
| - options.adjust_agc_delta = rtc::Optional<int>(0);
|
| - options.experimental_agc = rtc::Optional<bool>(false);
|
| - options.extended_filter_aec = rtc::Optional<bool>(false);
|
| - options.delay_agnostic_aec = rtc::Optional<bool>(false);
|
| - options.experimental_ns = rtc::Optional<bool>(false);
|
| - options.aec_dump = rtc::Optional<bool>(false);
|
| - if (!ApplyOptions(options)) {
|
| - 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);
|
| - }
|
| -
|
| - SetDefaultDevices();
|
| -
|
| - initialized_ = true;
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::Terminate() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
|
| - initialized_ = false;
|
| -
|
| - StopAecDump();
|
| -
|
| - voe_wrapper_->base()->Terminate();
|
| -}
|
| -
|
| -rtc::scoped_refptr<webrtc::AudioState>
|
| - WebRtcVoiceEngine::GetAudioState() const {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - return audio_state_;
|
| -}
|
| -
|
| -VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
|
| - const AudioOptions& options) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - return new WebRtcVoiceMediaChannel(this, options, call);
|
| -}
|
| -
|
| -bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString();
|
| - 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;
|
| - if (options.aecm_generate_comfort_noise) {
|
| - LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
|
| - << *options.aecm_generate_comfort_noise
|
| - << " (default is false).";
|
| - }
|
| -
|
| -#if defined(WEBRTC_IOS)
|
| - // On iOS, VPIO provides built-in EC and AGC.
|
| - options.echo_cancellation = rtc::Optional<bool>(false);
|
| - options.auto_gain_control = rtc::Optional<bool>(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(WEBRTC_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 = rtc::Optional<bool>(false);
|
| - options.experimental_agc = rtc::Optional<bool>(false);
|
| - options.extended_filter_aec = rtc::Optional<bool>(false);
|
| - options.experimental_ns = rtc::Optional<bool>(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(WEBRTC_IOS)
|
| - if (options.delay_agnostic_aec) {
|
| - use_delay_agnostic_aec = *options.delay_agnostic_aec;
|
| - if (use_delay_agnostic_aec) {
|
| - options.echo_cancellation = rtc::Optional<bool>(true);
|
| - options.extended_filter_aec = rtc::Optional<bool>(true);
|
| - ec_mode = webrtc::kEcConference;
|
| - }
|
| - }
|
| -#endif
|
| -
|
| - webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
|
| -
|
| - if (options.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 =
|
| - *options.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 = rtc::Optional<bool>(false);
|
| - LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
|
| - }
|
| - }
|
| - if (voep->SetEcStatus(*options.echo_cancellation, ec_mode) == -1) {
|
| - LOG_RTCERR2(SetEcStatus, *options.echo_cancellation, ec_mode);
|
| - return false;
|
| - } else {
|
| - LOG(LS_INFO) << "Echo control set to " << *options.echo_cancellation
|
| - << " with mode " << ec_mode;
|
| - }
|
| -#if !defined(ANDROID)
|
| - // TODO(ajm): Remove the error return on Android from webrtc.
|
| - if (voep->SetEcMetricsStatus(*options.echo_cancellation) == -1) {
|
| - LOG_RTCERR1(SetEcMetricsStatus, *options.echo_cancellation);
|
| - return false;
|
| - }
|
| -#endif
|
| - if (ec_mode == webrtc::kEcAecm) {
|
| - bool cn = options.aecm_generate_comfort_noise.value_or(false);
|
| - if (voep->SetAecmMode(aecm_mode, cn) != 0) {
|
| - LOG_RTCERR2(SetAecmMode, aecm_mode, cn);
|
| - return false;
|
| - }
|
| - }
|
| - }
|
| -
|
| - if (options.auto_gain_control) {
|
| - const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable();
|
| - if (built_in_agc) {
|
| - if (voe_wrapper_->hw()->EnableBuiltInAGC(*options.auto_gain_control) ==
|
| - 0 &&
|
| - *options.auto_gain_control) {
|
| - // Disable internal software AGC if built-in AGC is enabled,
|
| - // i.e., replace the software AGC with the built-in AGC.
|
| - options.auto_gain_control = rtc::Optional<bool>(false);
|
| - LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
|
| - }
|
| - }
|
| - if (voep->SetAgcStatus(*options.auto_gain_control, agc_mode) == -1) {
|
| - LOG_RTCERR2(SetAgcStatus, *options.auto_gain_control, agc_mode);
|
| - return false;
|
| - } else {
|
| - LOG(LS_INFO) << "Auto gain set to " << *options.auto_gain_control
|
| - << " with mode " << agc_mode;
|
| - }
|
| - }
|
| -
|
| - if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
|
| - options.tx_agc_limiter) {
|
| - // 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.value_or(
|
| - default_agc_config_.targetLeveldBOv);
|
| - default_agc_config_.digitalCompressionGaindB =
|
| - options.tx_agc_digital_compression_gain.value_or(
|
| - default_agc_config_.digitalCompressionGaindB);
|
| - default_agc_config_.limiterEnable =
|
| - options.tx_agc_limiter.value_or(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;
|
| - }
|
| - }
|
| -
|
| - if (options.noise_suppression) {
|
| - const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable();
|
| - if (built_in_ns) {
|
| - if (voe_wrapper_->hw()->EnableBuiltInNS(*options.noise_suppression) ==
|
| - 0 &&
|
| - *options.noise_suppression) {
|
| - // Disable internal software NS if built-in NS is enabled,
|
| - // i.e., replace the software NS with the built-in NS.
|
| - options.noise_suppression = rtc::Optional<bool>(false);
|
| - LOG(LS_INFO) << "Disabling NS since built-in NS will be used instead";
|
| - }
|
| - }
|
| - if (voep->SetNsStatus(*options.noise_suppression, ns_mode) == -1) {
|
| - LOG_RTCERR2(SetNsStatus, *options.noise_suppression, ns_mode);
|
| - return false;
|
| - } else {
|
| - LOG(LS_INFO) << "Noise suppression set to " << *options.noise_suppression
|
| - << " with mode " << ns_mode;
|
| - }
|
| - }
|
| -
|
| - if (options.highpass_filter) {
|
| - LOG(LS_INFO) << "High pass filter enabled? " << *options.highpass_filter;
|
| - if (voep->EnableHighPassFilter(*options.highpass_filter) == -1) {
|
| - LOG_RTCERR1(SetHighpassFilterStatus, *options.highpass_filter);
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - if (options.stereo_swapping) {
|
| - LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;
|
| - voep->EnableStereoChannelSwapping(*options.stereo_swapping);
|
| - if (voep->IsStereoChannelSwappingEnabled() != *options.stereo_swapping) {
|
| - LOG_RTCERR1(EnableStereoChannelSwapping, *options.stereo_swapping);
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - if (options.audio_jitter_buffer_max_packets) {
|
| - LOG(LS_INFO) << "NetEq capacity is "
|
| - << *options.audio_jitter_buffer_max_packets;
|
| - voe_config_.Set<webrtc::NetEqCapacityConfig>(
|
| - new webrtc::NetEqCapacityConfig(
|
| - *options.audio_jitter_buffer_max_packets));
|
| - }
|
| -
|
| - if (options.audio_jitter_buffer_fast_accelerate) {
|
| - LOG(LS_INFO) << "NetEq fast mode? "
|
| - << *options.audio_jitter_buffer_fast_accelerate;
|
| - voe_config_.Set<webrtc::NetEqFastAccelerate>(
|
| - new webrtc::NetEqFastAccelerate(
|
| - *options.audio_jitter_buffer_fast_accelerate));
|
| - }
|
| -
|
| - if (options.typing_detection) {
|
| - LOG(LS_INFO) << "Typing detection is enabled? "
|
| - << *options.typing_detection;
|
| - if (voep->SetTypingDetectionStatus(*options.typing_detection) == -1) {
|
| - // In case of error, log the info and continue
|
| - LOG_RTCERR1(SetTypingDetectionStatus, *options.typing_detection);
|
| - }
|
| - }
|
| -
|
| - if (options.adjust_agc_delta) {
|
| - LOG(LS_INFO) << "Adjust agc delta is " << *options.adjust_agc_delta;
|
| - if (!AdjustAgcLevel(*options.adjust_agc_delta)) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - if (options.aec_dump) {
|
| - LOG(LS_INFO) << "Aec dump is enabled? " << *options.aec_dump;
|
| - if (*options.aec_dump)
|
| - StartAecDump(kAecDumpByAudioOptionFilename);
|
| - else
|
| - StopAecDump();
|
| - }
|
| -
|
| - webrtc::Config config;
|
| -
|
| - if (options.delay_agnostic_aec)
|
| - delay_agnostic_aec_ = options.delay_agnostic_aec;
|
| - if (delay_agnostic_aec_) {
|
| - LOG(LS_INFO) << "Delay agnostic aec is enabled? " << *delay_agnostic_aec_;
|
| - config.Set<webrtc::DelayAgnostic>(
|
| - new webrtc::DelayAgnostic(*delay_agnostic_aec_));
|
| - }
|
| -
|
| - if (options.extended_filter_aec) {
|
| - extended_filter_aec_ = options.extended_filter_aec;
|
| - }
|
| - if (extended_filter_aec_) {
|
| - LOG(LS_INFO) << "Extended filter aec is enabled? " << *extended_filter_aec_;
|
| - config.Set<webrtc::ExtendedFilter>(
|
| - new webrtc::ExtendedFilter(*extended_filter_aec_));
|
| - }
|
| -
|
| - if (options.experimental_ns) {
|
| - experimental_ns_ = options.experimental_ns;
|
| - }
|
| - if (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);
|
| - }
|
| -
|
| - if (options.recording_sample_rate) {
|
| - LOG(LS_INFO) << "Recording sample rate is "
|
| - << *options.recording_sample_rate;
|
| - if (voe_wrapper_->hw()->SetRecordingSampleRate(
|
| - *options.recording_sample_rate)) {
|
| - LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
|
| - }
|
| - }
|
| -
|
| - if (options.playout_sample_rate) {
|
| - LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
|
| - if (voe_wrapper_->hw()->SetPlayoutSampleRate(
|
| - *options.playout_sample_rate)) {
|
| - LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
|
| - }
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::SetDefaultDevices() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| -#if !defined(WEBRTC_IOS)
|
| - int in_id = kDefaultAudioDeviceId;
|
| - int out_id = kDefaultAudioDeviceId;
|
| - LOG(LS_INFO) << "Setting microphone to (id=" << in_id
|
| - << ") and speaker to (id=" << out_id << ")";
|
| -
|
| - bool ret = true;
|
| - if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
|
| - LOG_RTCERR1(SetRecordingDevice, in_id);
|
| - ret = false;
|
| - }
|
| - webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
|
| - if (ap) {
|
| - ap->Initialize();
|
| - }
|
| -
|
| - if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
|
| - LOG_RTCERR1(SetPlayoutDevice, out_id);
|
| - ret = false;
|
| - }
|
| -
|
| - if (ret) {
|
| - LOG(LS_INFO) << "Set microphone to (id=" << in_id
|
| - << ") and speaker to (id=" << out_id << ")";
|
| - }
|
| -#endif // !WEBRTC_IOS
|
| -}
|
| -
|
| -bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - 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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(level >= 0 && level <= 255);
|
| - if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
|
| - LOG_RTCERR1(SetSpeakerVolume, level);
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -int WebRtcVoiceEngine::GetInputLevel() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - unsigned int ulevel;
|
| - return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
|
| - static_cast<int>(ulevel) : -1;
|
| -}
|
| -
|
| -const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
|
| - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
|
| - return codecs_;
|
| -}
|
| -
|
| -RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
|
| - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
|
| - RtpCapabilities capabilities;
|
| - capabilities.header_extensions.push_back(RtpHeaderExtension(
|
| - kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId));
|
| - capabilities.header_extensions.push_back(
|
| - RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
|
| - kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
|
| - if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") ==
|
| - "Enabled") {
|
| - capabilities.header_extensions.push_back(RtpHeaderExtension(
|
| - kRtpTransportSequenceNumberHeaderExtension,
|
| - kRtpTransportSequenceNumberHeaderExtensionDefaultId));
|
| - }
|
| - return capabilities;
|
| -}
|
| -
|
| -int WebRtcVoiceEngine::GetLastEngineError() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - return voe_wrapper_->error();
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
|
| - int length) {
|
| - // Note: This callback can happen on any thread!
|
| - 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);
|
| - LOG_V(sev) << "webrtc: " << msg;
|
| - }
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(channel);
|
| - channels_.push_back(channel);
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - auto it = std::find(channels_.begin(), channels_.end(), channel);
|
| - RTC_DCHECK(it != channels_.end());
|
| - channels_.erase(it);
|
| -}
|
| -
|
| -// 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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - 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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - 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,
|
| - int64_t max_size_bytes) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - 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_->base()->audio_processing()->StartDebugRecording(
|
| - aec_dump_file_stream, max_size_bytes) !=
|
| - webrtc::AudioProcessing::kNoError) {
|
| - LOG_RTCERR0(StartDebugRecording);
|
| - fclose(aec_dump_file_stream);
|
| - return false;
|
| - }
|
| - is_dumping_aec_ = true;
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - if (!is_dumping_aec_) {
|
| - // Start dumping AEC when we are not dumping.
|
| - if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
|
| - filename.c_str(), -1) != webrtc::AudioProcessing::kNoError) {
|
| - LOG_RTCERR1(StartDebugRecording, filename.c_str());
|
| - } else {
|
| - is_dumping_aec_ = true;
|
| - }
|
| - }
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::StopAecDump() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - if (is_dumping_aec_) {
|
| - // Stop dumping AEC when we are dumping.
|
| - if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
|
| - webrtc::AudioProcessing::kNoError) {
|
| - LOG_RTCERR0(StopDebugRecording);
|
| - }
|
| - is_dumping_aec_ = false;
|
| - }
|
| -}
|
| -
|
| -bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
|
| - if (event_log) {
|
| - return event_log->StartLogging(file);
|
| - }
|
| - LOG_RTCERR0(StartRtcEventLog);
|
| - return false;
|
| -}
|
| -
|
| -void WebRtcVoiceEngine::StopRtcEventLog() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
|
| - if (event_log) {
|
| - event_log->StopLogging();
|
| - return;
|
| - }
|
| - LOG_RTCERR0(StopRtcEventLog);
|
| -}
|
| -
|
| -int WebRtcVoiceEngine::CreateVoEChannel() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - return voe_wrapper_->base()->CreateChannel(voe_config_);
|
| -}
|
| -
|
| -class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
|
| - : public AudioRenderer::Sink {
|
| - public:
|
| - WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
|
| - uint32_t ssrc, const std::string& c_name,
|
| - const std::vector<webrtc::RtpExtension>& extensions,
|
| - webrtc::Call* call)
|
| - : voe_audio_transport_(voe_audio_transport),
|
| - call_(call),
|
| - config_(nullptr) {
|
| - RTC_DCHECK_GE(ch, 0);
|
| - // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
|
| - // RTC_DCHECK(voe_audio_transport);
|
| - RTC_DCHECK(call);
|
| - audio_capture_thread_checker_.DetachFromThread();
|
| - config_.rtp.ssrc = ssrc;
|
| - config_.rtp.c_name = c_name;
|
| - config_.voe_channel_id = ch;
|
| - RecreateAudioSendStream(extensions);
|
| - }
|
| -
|
| - ~WebRtcAudioSendStream() override {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - Stop();
|
| - call_->DestroyAudioSendStream(stream_);
|
| - }
|
| -
|
| - void RecreateAudioSendStream(
|
| - const std::vector<webrtc::RtpExtension>& extensions) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - if (stream_) {
|
| - call_->DestroyAudioSendStream(stream_);
|
| - stream_ = nullptr;
|
| - }
|
| - config_.rtp.extensions = extensions;
|
| - RTC_DCHECK(!stream_);
|
| - stream_ = call_->CreateAudioSendStream(config_);
|
| - RTC_CHECK(stream_);
|
| - }
|
| -
|
| - bool SendTelephoneEvent(int payload_type, uint8_t event,
|
| - uint32_t duration_ms) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(stream_);
|
| - return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
|
| - }
|
| -
|
| - webrtc::AudioSendStream::Stats GetStats() const {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(stream_);
|
| - return stream_->GetStats();
|
| - }
|
| -
|
| - // 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_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(renderer);
|
| - if (renderer_) {
|
| - RTC_DCHECK(renderer_ == renderer);
|
| - return;
|
| - }
|
| - renderer->SetSink(this);
|
| - renderer_ = renderer;
|
| - }
|
| -
|
| - // Stops rendering by setting the sink of the renderer to nullptr. No data
|
| - // callback will be received after this method.
|
| - // This method is called on the libjingle worker thread.
|
| - void Stop() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - if (renderer_) {
|
| - renderer_->SetSink(nullptr);
|
| - renderer_ = nullptr;
|
| - }
|
| - }
|
| -
|
| - // AudioRenderer::Sink implementation.
|
| - // This method is called on the audio thread.
|
| - void OnData(const void* audio_data,
|
| - int bits_per_sample,
|
| - int sample_rate,
|
| - size_t number_of_channels,
|
| - size_t number_of_frames) override {
|
| - RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(voe_audio_transport_);
|
| - voe_audio_transport_->OnData(config_.voe_channel_id,
|
| - 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_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - // Set |renderer_| to nullptr to make sure no more callback will get into
|
| - // the renderer.
|
| - renderer_ = nullptr;
|
| - }
|
| -
|
| - // Accessor to the VoE channel ID.
|
| - int channel() const {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - return config_.voe_channel_id;
|
| - }
|
| -
|
| - private:
|
| - rtc::ThreadChecker worker_thread_checker_;
|
| - rtc::ThreadChecker audio_capture_thread_checker_;
|
| - webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
|
| - webrtc::Call* call_ = nullptr;
|
| - webrtc::AudioSendStream::Config config_;
|
| - // The stream is owned by WebRtcAudioSendStream and may be reallocated if
|
| - // configuration changes.
|
| - webrtc::AudioSendStream* stream_ = nullptr;
|
| -
|
| - // Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
|
| - // PeerConnection will make sure invalidating the pointer before the object
|
| - // goes away.
|
| - AudioRenderer* renderer_ = nullptr;
|
| -
|
| - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
|
| -};
|
| -
|
| -class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
|
| - public:
|
| - WebRtcAudioReceiveStream(int ch,
|
| - uint32_t remote_ssrc,
|
| - uint32_t local_ssrc,
|
| - bool use_transport_cc,
|
| - const std::string& sync_group,
|
| - const std::vector<webrtc::RtpExtension>& extensions,
|
| - webrtc::Call* call)
|
| - : call_(call), config_() {
|
| - RTC_DCHECK_GE(ch, 0);
|
| - RTC_DCHECK(call);
|
| - config_.rtp.remote_ssrc = remote_ssrc;
|
| - config_.rtp.local_ssrc = local_ssrc;
|
| - config_.voe_channel_id = ch;
|
| - config_.sync_group = sync_group;
|
| - RecreateAudioReceiveStream(use_transport_cc, extensions);
|
| - }
|
| -
|
| - ~WebRtcAudioReceiveStream() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - call_->DestroyAudioReceiveStream(stream_);
|
| - }
|
| -
|
| - void RecreateAudioReceiveStream(
|
| - const std::vector<webrtc::RtpExtension>& extensions) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RecreateAudioReceiveStream(config_.rtp.transport_cc, extensions);
|
| - }
|
| - void RecreateAudioReceiveStream(bool use_transport_cc) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RecreateAudioReceiveStream(use_transport_cc, config_.rtp.extensions);
|
| - }
|
| -
|
| - webrtc::AudioReceiveStream::Stats GetStats() const {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(stream_);
|
| - return stream_->GetStats();
|
| - }
|
| -
|
| - int channel() const {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - return config_.voe_channel_id;
|
| - }
|
| -
|
| - void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - stream_->SetSink(std::move(sink));
|
| - }
|
| -
|
| - private:
|
| - void RecreateAudioReceiveStream(
|
| - bool use_transport_cc,
|
| - const std::vector<webrtc::RtpExtension>& extensions) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - if (stream_) {
|
| - call_->DestroyAudioReceiveStream(stream_);
|
| - stream_ = nullptr;
|
| - }
|
| - config_.rtp.extensions = extensions;
|
| - config_.rtp.transport_cc = use_transport_cc;
|
| - RTC_DCHECK(!stream_);
|
| - stream_ = call_->CreateAudioReceiveStream(config_);
|
| - RTC_CHECK(stream_);
|
| - }
|
| -
|
| - rtc::ThreadChecker worker_thread_checker_;
|
| - webrtc::Call* call_ = nullptr;
|
| - webrtc::AudioReceiveStream::Config config_;
|
| - // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
|
| - // configuration changes.
|
| - webrtc::AudioReceiveStream* stream_ = nullptr;
|
| -
|
| - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
|
| -};
|
| -
|
| -WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
|
| - const AudioOptions& options,
|
| - webrtc::Call* call)
|
| - : engine_(engine), call_(call) {
|
| - LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
|
| - RTC_DCHECK(call);
|
| - engine->RegisterChannel(this);
|
| - SetOptions(options);
|
| -}
|
| -
|
| -WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
|
| - // TODO(solenberg): Should be able to delete the streams directly, without
|
| - // going through RemoveNnStream(), once stream objects handle
|
| - // all (de)configuration.
|
| - while (!send_streams_.empty()) {
|
| - RemoveSendStream(send_streams_.begin()->first);
|
| - }
|
| - while (!recv_streams_.empty()) {
|
| - RemoveRecvStream(recv_streams_.begin()->first);
|
| - }
|
| - engine()->UnregisterChannel(this);
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::SetSendParameters(
|
| - const AudioSendParameters& params) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
|
| - << params.ToString();
|
| - // TODO(pthatcher): Refactor this to be more clean now that we have
|
| - // all the information at once.
|
| -
|
| - if (!SetSendCodecs(params.codecs)) {
|
| - return false;
|
| - }
|
| -
|
| - if (!ValidateRtpExtensions(params.extensions)) {
|
| - return false;
|
| - }
|
| - std::vector<webrtc::RtpExtension> filtered_extensions =
|
| - FilterRtpExtensions(params.extensions,
|
| - webrtc::RtpExtension::IsSupportedForAudio, true);
|
| - if (send_rtp_extensions_ != filtered_extensions) {
|
| - send_rtp_extensions_.swap(filtered_extensions);
|
| - for (auto& it : send_streams_) {
|
| - it.second->RecreateAudioSendStream(send_rtp_extensions_);
|
| - }
|
| - }
|
| -
|
| - if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
|
| - return false;
|
| - }
|
| - return SetOptions(params.options);
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::SetRecvParameters(
|
| - const AudioRecvParameters& params) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "
|
| - << params.ToString();
|
| - // TODO(pthatcher): Refactor this to be more clean now that we have
|
| - // all the information at once.
|
| -
|
| - if (!SetRecvCodecs(params.codecs)) {
|
| - return false;
|
| - }
|
| -
|
| - if (!ValidateRtpExtensions(params.extensions)) {
|
| - return false;
|
| - }
|
| - std::vector<webrtc::RtpExtension> filtered_extensions =
|
| - FilterRtpExtensions(params.extensions,
|
| - webrtc::RtpExtension::IsSupportedForAudio, false);
|
| - if (recv_rtp_extensions_ != filtered_extensions) {
|
| - recv_rtp_extensions_.swap(filtered_extensions);
|
| - for (auto& it : recv_streams_) {
|
| - it.second->RecreateAudioReceiveStream(recv_rtp_extensions_);
|
| - }
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - 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);
|
| -
|
| - // 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 (!engine()->ApplyOptions(options_)) {
|
| - LOG(LS_WARNING) <<
|
| - "Failed to apply engine options during channel SetOptions.";
|
| - return false;
|
| - }
|
| -
|
| - if (dscp_option_changed) {
|
| - rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
|
| - if (options_.dscp.value_or(false)) {
|
| - dscp = kAudioDscpValue;
|
| - }
|
| - if (MediaChannel::SetDscp(dscp) != 0) {
|
| - LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
|
| - }
|
| - }
|
| -
|
| - LOG(LS_INFO) << "Set voice channel options. Current options: "
|
| - << options_.ToString();
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::SetRecvCodecs(
|
| - const std::vector<AudioCodec>& codecs) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| -
|
| - // Set the payload types to be used for incoming media.
|
| - LOG(LS_INFO) << "Setting receive voice codecs.";
|
| -
|
| - if (!VerifyUniquePayloadTypes(codecs)) {
|
| - LOG(LS_ERROR) << "Codec payload types overlap.";
|
| - return false;
|
| - }
|
| -
|
| - 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 = true;
|
| - for (const AudioCodec& codec : new_codecs) {
|
| - webrtc::CodecInst voe_codec;
|
| - if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
|
| - LOG(LS_INFO) << ToString(codec);
|
| - voe_codec.pltype = codec.id;
|
| - for (const auto& ch : recv_streams_) {
|
| - if (engine()->voe()->codec()->SetRecPayloadType(
|
| - ch.second->channel(), voe_codec) == -1) {
|
| - LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
|
| - ToString(voe_codec));
|
| - result = false;
|
| - }
|
| - }
|
| - } else {
|
| - LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
|
| - result = false;
|
| - break;
|
| - }
|
| - }
|
| - 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.
|
| - 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;
|
| - int red_payload_type = -1;
|
| -
|
| - // Set send codec (the first non-telephone-event/CN codec)
|
| - const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
|
| - codecs, &send_codec, &red_payload_type);
|
| - if (codec) {
|
| - if (red_payload_type != -1) {
|
| - // 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,
|
| - red_payload_type) == -1) {
|
| - LOG_RTCERR3(SetREDStatus, channel, true, red_payload_type);
|
| - return false;
|
| - }
|
| - } else {
|
| - nack_enabled = HasNack(*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 (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(&send_codec, ptime_ms)) {
|
| - LOG(LS_WARNING) << "Failed to set packet size for codec "
|
| - << send_codec.plname;
|
| - return false;
|
| - }
|
| - }
|
| - }
|
| - }
|
| -
|
| - if (nack_enabled_ != nack_enabled) {
|
| - SetNack(channel, nack_enabled);
|
| - nack_enabled_ = nack_enabled;
|
| - }
|
| - if (!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 "
|
| - << (enable_opus_dtx ? "enable" : "disable")
|
| - << " 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 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 (!WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
|
| - LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
|
| - continue;
|
| - }
|
| -
|
| - 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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - // TODO(solenberg): Validate input - that payload types don't overlap, are
|
| - // within range, filter out codecs we don't support,
|
| - // redundant codecs etc.
|
| -
|
| - // Find the DTMF telephone event "codec" payload type.
|
| - dtmf_payload_type_ = rtc::Optional<int>();
|
| - for (const AudioCodec& codec : codecs) {
|
| - if (IsCodec(codec, kDtmfCodecName)) {
|
| - dtmf_payload_type_ = rtc::Optional<int>(codec.id);
|
| - break;
|
| - }
|
| - }
|
| -
|
| - // Cache the codecs in order to configure the channel created later.
|
| - send_codecs_ = codecs;
|
| - for (const auto& ch : send_streams_) {
|
| - if (!SetSendCodecs(ch.second->channel(), codecs)) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - // Set nack status on receive channels and update |nack_enabled_|.
|
| - for (const auto& ch : recv_streams_) {
|
| - SetNack(ch.second->channel(), nack_enabled_);
|
| - }
|
| -
|
| - // Check if the transport cc feedback has changed on the preferred send codec,
|
| - // and in that case reconfigure all receive streams.
|
| - webrtc::CodecInst voe_codec;
|
| - int red_payload_type;
|
| - const AudioCodec* send_codec = WebRtcVoiceCodecs::GetPreferredCodec(
|
| - send_codecs_, &voe_codec, &red_payload_type);
|
| - if (send_codec) {
|
| - bool transport_cc = HasTransportCc(*send_codec);
|
| - if (transport_cc_enabled_ != transport_cc) {
|
| - LOG(LS_INFO) << "Recreate all the receive streams because the send "
|
| - "codec has changed.";
|
| - transport_cc_enabled_ = transport_cc;
|
| - for (auto& kv : recv_streams_) {
|
| - RTC_DCHECK(kv.second != nullptr);
|
| - kv.second->RecreateAudioReceiveStream(transport_cc_enabled_);
|
| - }
|
| - }
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -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(
|
| - 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::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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - if (playout_ == playout) {
|
| - return true;
|
| - }
|
| -
|
| - for (const auto& ch : recv_streams_) {
|
| - if (!SetPlayout(ch.second->channel(), playout)) {
|
| - LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
|
| - << ch.second->channel() << " failed";
|
| - return false;
|
| - }
|
| - }
|
| - playout_ = playout;
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
|
| - desired_send_ = send;
|
| - if (!send_streams_.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;
|
| - }
|
| -
|
| - // Apply channel specific options when channel is enabled for sending.
|
| - if (send == SEND_MICROPHONE) {
|
| - engine()->ApplyOptions(options_);
|
| - }
|
| -
|
| - // Change the settings on each send channel.
|
| - for (const auto& ch : send_streams_) {
|
| - if (!ChangeSend(ch.second->channel(), send)) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - 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;
|
| - }
|
| - } else { // SEND_NOTHING
|
| - RTC_DCHECK(send == SEND_NOTHING);
|
| - if (engine()->voe()->base()->StopSend(channel) == -1) {
|
| - LOG_RTCERR1(StopSend, channel);
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
|
| - bool enable,
|
| - const AudioOptions* options,
|
| - AudioRenderer* renderer) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - // 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, !enable)) {
|
| - return false;
|
| - }
|
| - if (enable && options) {
|
| - return SetOptions(*options);
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -int WebRtcVoiceMediaChannel::CreateVoEChannel() {
|
| - int id = engine()->CreateVoEChannel();
|
| - if (id == -1) {
|
| - LOG_RTCERR0(CreateVoEChannel);
|
| - return -1;
|
| - }
|
| - if (engine()->voe()->network()->RegisterExternalTransport(id, *this) == -1) {
|
| - LOG_RTCERR2(RegisterExternalTransport, id, this);
|
| - engine()->voe()->base()->DeleteChannel(id);
|
| - return -1;
|
| - }
|
| - return id;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::DeleteVoEChannel(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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
|
| -
|
| - uint32_t ssrc = sp.first_ssrc();
|
| - RTC_DCHECK(0 != ssrc);
|
| -
|
| - if (GetSendChannelId(ssrc) != -1) {
|
| - LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
|
| - return false;
|
| - }
|
| -
|
| - // Create a new channel for sending audio data.
|
| - int channel = CreateVoEChannel();
|
| - if (channel == -1) {
|
| - return false;
|
| - }
|
| -
|
| - // Save the channel to send_streams_, so that RemoveSendStream() can still
|
| - // delete the channel in case failure happens below.
|
| - webrtc::AudioTransport* audio_transport =
|
| - engine()->voe()->base()->audio_transport();
|
| - send_streams_.insert(std::make_pair(ssrc, new WebRtcAudioSendStream(
|
| - channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_)));
|
| -
|
| - // Set the current codecs to be used for the new channel. We need to do this
|
| - // after adding the channel to send_channels_, because of how max bitrate is
|
| - // currently being configured by SetSendCodec().
|
| - if (!send_codecs_.empty() && !SetSendCodecs(channel, send_codecs_)) {
|
| - RemoveSendStream(ssrc);
|
| - return false;
|
| - }
|
| -
|
| - // At this point the channel's local SSRC has been updated. If the channel is
|
| - // the first send channel make sure that all the receive channels are updated
|
| - // with the same SSRC in order to send receiver reports.
|
| - if (send_streams_.size() == 1) {
|
| - receiver_reports_ssrc_ = ssrc;
|
| - for (const auto& stream : recv_streams_) {
|
| - int recv_channel = stream.second->channel();
|
| - if (engine()->voe()->rtp()->SetLocalSSRC(recv_channel, ssrc) != 0) {
|
| - LOG_RTCERR2(SetLocalSSRC, recv_channel, ssrc);
|
| - return false;
|
| - }
|
| - engine()->voe()->base()->AssociateSendChannel(recv_channel, channel);
|
| - LOG(LS_INFO) << "VoiceEngine channel #" << recv_channel
|
| - << " is associated with channel #" << channel << ".";
|
| - }
|
| - }
|
| -
|
| - return ChangeSend(channel, desired_send_);
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
|
| -
|
| - auto it = send_streams_.find(ssrc);
|
| - if (it == send_streams_.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);
|
| -
|
| - // Clean up and delete the send stream+channel.
|
| - LOG(LS_INFO) << "Removing audio send stream " << ssrc
|
| - << " with VoiceEngine channel #" << channel << ".";
|
| - delete it->second;
|
| - send_streams_.erase(it);
|
| - if (!DeleteVoEChannel(channel)) {
|
| - return false;
|
| - }
|
| - if (send_streams_.empty()) {
|
| - ChangeSend(SEND_NOTHING);
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
|
| -
|
| - if (!ValidateStreamParams(sp)) {
|
| - return false;
|
| - }
|
| -
|
| - const uint32_t ssrc = sp.first_ssrc();
|
| - if (ssrc == 0) {
|
| - LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
|
| - return false;
|
| - }
|
| -
|
| - // Remove the default receive stream if one had been created with this ssrc;
|
| - // we'll recreate it then.
|
| - if (IsDefaultRecvStream(ssrc)) {
|
| - RemoveRecvStream(ssrc);
|
| - }
|
| -
|
| - if (GetReceiveChannelId(ssrc) != -1) {
|
| - LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
|
| - return false;
|
| - }
|
| -
|
| - // Create a new channel for receiving audio data.
|
| - const int channel = CreateVoEChannel();
|
| - if (channel == -1) {
|
| - return false;
|
| - }
|
| -
|
| - // Turn off all supported codecs.
|
| - // TODO(solenberg): Remove once "no codecs" is the default state of a stream.
|
| - for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
|
| - voe_codec.pltype = -1;
|
| - if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) {
|
| - LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
|
| - DeleteVoEChannel(channel);
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - // Only enable those configured for this channel.
|
| - for (const auto& codec : recv_codecs_) {
|
| - webrtc::CodecInst voe_codec;
|
| - if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
|
| - voe_codec.pltype = codec.id;
|
| - if (engine()->voe()->codec()->SetRecPayloadType(
|
| - channel, voe_codec) == -1) {
|
| - LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
|
| - DeleteVoEChannel(channel);
|
| - return false;
|
| - }
|
| - }
|
| - }
|
| -
|
| - const int send_channel = GetSendChannelId(receiver_reports_ssrc_);
|
| - if (send_channel != -1) {
|
| - // Associate receive channel with first send channel (so the receive channel
|
| - // can obtain RTT from the send channel)
|
| - engine()->voe()->base()->AssociateSendChannel(channel, send_channel);
|
| - LOG(LS_INFO) << "VoiceEngine channel #" << channel
|
| - << " is associated with channel #" << send_channel << ".";
|
| - }
|
| -
|
| - transport_cc_enabled_ =
|
| - !send_codecs_.empty() ? HasTransportCc(send_codecs_[0]) : false;
|
| -
|
| - recv_streams_.insert(std::make_pair(
|
| - ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
|
| - transport_cc_enabled_, sp.sync_label,
|
| - recv_rtp_extensions_, call_)));
|
| -
|
| - SetNack(channel, nack_enabled_);
|
| - SetPlayout(channel, playout_);
|
| -
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
|
| -
|
| - const auto it = recv_streams_.find(ssrc);
|
| - if (it == recv_streams_.end()) {
|
| - LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
|
| - << " which doesn't exist.";
|
| - return false;
|
| - }
|
| -
|
| - // Deregister default channel, if that's the one being destroyed.
|
| - if (IsDefaultRecvStream(ssrc)) {
|
| - default_recv_ssrc_ = -1;
|
| - }
|
| -
|
| - const int channel = it->second->channel();
|
| -
|
| - // Clean up and delete the receive stream+channel.
|
| - LOG(LS_INFO) << "Removing audio receive stream " << ssrc
|
| - << " with VoiceEngine channel #" << channel << ".";
|
| - it->second->SetRawAudioSink(nullptr);
|
| - delete it->second;
|
| - recv_streams_.erase(it);
|
| - return DeleteVoEChannel(channel);
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc,
|
| - AudioRenderer* renderer) {
|
| - auto it = send_streams_.find(ssrc);
|
| - if (it == send_streams_.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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - actives->clear();
|
| - for (const auto& ch : recv_streams_) {
|
| - int level = GetOutputLevel(ch.second->channel());
|
| - if (level > 0) {
|
| - actives->push_back(std::make_pair(ch.first, level));
|
| - }
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -int WebRtcVoiceMediaChannel::GetOutputLevel() {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - int highest = 0;
|
| - for (const auto& ch : recv_streams_) {
|
| - highest = std::max(GetOutputLevel(ch.second->channel()), 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::SetOutputVolume(uint32_t ssrc, double volume) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - if (ssrc == 0) {
|
| - default_recv_volume_ = volume;
|
| - if (default_recv_ssrc_ == -1) {
|
| - return true;
|
| - }
|
| - ssrc = static_cast<uint32_t>(default_recv_ssrc_);
|
| - }
|
| - int ch_id = GetReceiveChannelId(ssrc);
|
| - if (ch_id < 0) {
|
| - LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
|
| - return false;
|
| - }
|
| -
|
| - if (-1 == engine()->voe()->volume()->SetChannelOutputVolumeScaling(ch_id,
|
| - volume)) {
|
| - LOG_RTCERR2(SetChannelOutputVolumeScaling, ch_id, volume);
|
| - return false;
|
| - }
|
| - LOG(LS_INFO) << "SetOutputVolume to " << volume
|
| - << " for channel " << ch_id << " and ssrc " << ssrc;
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
|
| - return dtmf_payload_type_ ? true : false;
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
|
| - int duration) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf";
|
| - if (!dtmf_payload_type_) {
|
| - return false;
|
| - }
|
| -
|
| - // Figure out which WebRtcAudioSendStream to send the event on.
|
| - auto it = ssrc != 0 ? send_streams_.find(ssrc) : send_streams_.begin();
|
| - if (it == send_streams_.end()) {
|
| - LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
|
| - return false;
|
| - }
|
| - if (event < kMinTelephoneEventCode ||
|
| - event > kMaxTelephoneEventCode) {
|
| - LOG(LS_WARNING) << "DTMF event code " << event << " out of range.";
|
| - return false;
|
| - }
|
| - if (duration < kMinTelephoneEventDuration ||
|
| - duration > kMaxTelephoneEventDuration) {
|
| - LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range.";
|
| - return false;
|
| - }
|
| - return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration);
|
| -}
|
| -
|
| -void WebRtcVoiceMediaChannel::OnPacketReceived(
|
| - rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| -
|
| - uint32_t ssrc = 0;
|
| - if (!GetRtpSsrc(packet->data(), packet->size(), &ssrc)) {
|
| - return;
|
| - }
|
| -
|
| - // If we don't have a default channel, and the SSRC is unknown, create a
|
| - // default channel.
|
| - if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) {
|
| - StreamParams sp;
|
| - sp.ssrcs.push_back(ssrc);
|
| - LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
|
| - if (!AddRecvStream(sp)) {
|
| - LOG(LS_WARNING) << "Could not create default receive stream.";
|
| - return;
|
| - }
|
| - default_recv_ssrc_ = ssrc;
|
| - SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
|
| - if (default_sink_) {
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
|
| - new ProxySink(default_sink_.get()));
|
| - SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
|
| - }
|
| - }
|
| -
|
| - // Forward packet to Call. If the SSRC is unknown we'll return after this.
|
| - const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
|
| - packet_time.not_before);
|
| - webrtc::PacketReceiver::DeliveryStatus delivery_result =
|
| - call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
|
| - reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
|
| - webrtc_packet_time);
|
| - if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) {
|
| - // If the SSRC is unknown here, route it to the default channel, if we have
|
| - // one. See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
|
| - if (default_recv_ssrc_ == -1) {
|
| - return;
|
| - } else {
|
| - ssrc = default_recv_ssrc_;
|
| - }
|
| - }
|
| -
|
| - // Find the channel to send this packet to. It must exist since webrtc::Call
|
| - // was able to demux the packet.
|
| - int channel = GetReceiveChannelId(ssrc);
|
| - RTC_DCHECK(channel != -1);
|
| -
|
| - // Pass it off to the decoder.
|
| - engine()->voe()->network()->ReceivedRTPPacket(
|
| - channel, packet->data(), packet->size(), webrtc_packet_time);
|
| -}
|
| -
|
| -void WebRtcVoiceMediaChannel::OnRtcpReceived(
|
| - rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
|
| - RTC_DCHECK(worker_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 receive channel that is listening.
|
| - if (type == kRtcpTypeSR) {
|
| - uint32_t ssrc = 0;
|
| - if (!GetRtcpSsrc(packet->data(), packet->size(), &ssrc)) {
|
| - return;
|
| - }
|
| - int recv_channel_id = GetReceiveChannelId(ssrc);
|
| - if (recv_channel_id != -1) {
|
| - engine()->voe()->network()->ReceivedRTCPPacket(
|
| - recv_channel_id, packet->data(), packet->size());
|
| - }
|
| - }
|
| -
|
| - // 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_streams_) {
|
| - engine()->voe()->network()->ReceivedRTCPPacket(
|
| - ch.second->channel(), packet->data(), packet->size());
|
| - }
|
| -}
|
| -
|
| -bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - int channel = GetSendChannelId(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_streams_) {
|
| - 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 = WebRtcVoiceCodecs::IsCodecMultiRate(codec);
|
| -
|
| - if (is_multi_rate) {
|
| - // If codec is multi-rate then just set the bitrate.
|
| - codec.rate = bps;
|
| - for (const auto& ch : send_streams_) {
|
| - if (!SetSendCodec(ch.second->channel(), 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) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - RTC_DCHECK(info);
|
| -
|
| - // Get SSRC and stats for each sender.
|
| - RTC_DCHECK(info->senders.size() == 0);
|
| - for (const auto& stream : send_streams_) {
|
| - webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
|
| - VoiceSenderInfo sinfo;
|
| - sinfo.add_ssrc(stats.local_ssrc);
|
| - sinfo.bytes_sent = stats.bytes_sent;
|
| - sinfo.packets_sent = stats.packets_sent;
|
| - sinfo.packets_lost = stats.packets_lost;
|
| - sinfo.fraction_lost = stats.fraction_lost;
|
| - sinfo.codec_name = stats.codec_name;
|
| - sinfo.ext_seqnum = stats.ext_seqnum;
|
| - sinfo.jitter_ms = stats.jitter_ms;
|
| - sinfo.rtt_ms = stats.rtt_ms;
|
| - sinfo.audio_level = stats.audio_level;
|
| - sinfo.aec_quality_min = stats.aec_quality_min;
|
| - sinfo.echo_delay_median_ms = stats.echo_delay_median_ms;
|
| - sinfo.echo_delay_std_ms = stats.echo_delay_std_ms;
|
| - sinfo.echo_return_loss = stats.echo_return_loss;
|
| - sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement;
|
| - sinfo.typing_noise_detected =
|
| - (send_ == SEND_NOTHING ? false : stats.typing_noise_detected);
|
| - info->senders.push_back(sinfo);
|
| - }
|
| -
|
| - // Get SSRC and stats for each receiver.
|
| - RTC_DCHECK(info->receivers.size() == 0);
|
| - for (const auto& stream : recv_streams_) {
|
| - webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
|
| - VoiceReceiverInfo rinfo;
|
| - rinfo.add_ssrc(stats.remote_ssrc);
|
| - rinfo.bytes_rcvd = stats.bytes_rcvd;
|
| - rinfo.packets_rcvd = stats.packets_rcvd;
|
| - rinfo.packets_lost = stats.packets_lost;
|
| - rinfo.fraction_lost = stats.fraction_lost;
|
| - rinfo.codec_name = stats.codec_name;
|
| - rinfo.ext_seqnum = stats.ext_seqnum;
|
| - rinfo.jitter_ms = stats.jitter_ms;
|
| - rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
|
| - rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
|
| - rinfo.delay_estimate_ms = stats.delay_estimate_ms;
|
| - rinfo.audio_level = stats.audio_level;
|
| - rinfo.expand_rate = stats.expand_rate;
|
| - rinfo.speech_expand_rate = stats.speech_expand_rate;
|
| - rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
|
| - rinfo.accelerate_rate = stats.accelerate_rate;
|
| - rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
|
| - rinfo.decoding_calls_to_silence_generator =
|
| - stats.decoding_calls_to_silence_generator;
|
| - rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
|
| - rinfo.decoding_normal = stats.decoding_normal;
|
| - rinfo.decoding_plc = stats.decoding_plc;
|
| - rinfo.decoding_cng = stats.decoding_cng;
|
| - rinfo.decoding_plc_cng = stats.decoding_plc_cng;
|
| - rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
|
| - info->receivers.push_back(rinfo);
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVoiceMediaChannel::SetRawAudioSink(
|
| - uint32_t ssrc,
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
|
| - << " " << (sink ? "(ptr)" : "NULL");
|
| - if (ssrc == 0) {
|
| - if (default_recv_ssrc_ != -1) {
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
|
| - sink ? new ProxySink(sink.get()) : nullptr);
|
| - SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
|
| - }
|
| - default_sink_ = std::move(sink);
|
| - return;
|
| - }
|
| - const auto it = recv_streams_.find(ssrc);
|
| - if (it == recv_streams_.end()) {
|
| - LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
|
| - return;
|
| - }
|
| - it->second->SetRawAudioSink(std::move(sink));
|
| -}
|
| -
|
| -int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
|
| - unsigned int ulevel = 0;
|
| - int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
|
| - return (ret == 0) ? static_cast<int>(ulevel) : -1;
|
| -}
|
| -
|
| -int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - const auto it = recv_streams_.find(ssrc);
|
| - if (it != recv_streams_.end()) {
|
| - return it->second->channel();
|
| - }
|
| - return -1;
|
| -}
|
| -
|
| -int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
|
| - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - const auto it = send_streams_.find(ssrc);
|
| - if (it != send_streams_.end()) {
|
| - return it->second->channel();
|
| - }
|
| - return -1;
|
| -}
|
| -
|
| -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;
|
| -}
|
| -} // namespace cricket
|
| -
|
| -#endif // HAVE_WEBRTC_VOICE
|
|
|
Chromium Code Reviews
Issue 1587193006:
Move talk/media to webrtc/media (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master