diff options
Diffstat (limited to 'pica-client/audiovideocallcontroller.cpp')
| -rw-r--r-- | pica-client/audiovideocallcontroller.cpp | 57 |
1 files changed, 30 insertions, 27 deletions
diff --git a/pica-client/audiovideocallcontroller.cpp b/pica-client/audiovideocallcontroller.cpp index 72b6fc2..9b4ae4d 100644 --- a/pica-client/audiovideocallcontroller.cpp +++ b/pica-client/audiovideocallcontroller.cpp @@ -18,32 +18,11 @@ #include "globals.h" #include "skynet.h" #include "settings.h" +#include "callsettings.h" #include <QByteArray> #include <QDebug> #include <QMetaObject> -// Capture/encode resolution announced to the peer in the 0x75 message. Fixed -// for now - a resolution setting is a planned follow-up, see VideoDevice for -// the rest of the encoding parameters. -static const quint16 kVideoWidth = 640; -static const quint16 kVideoHeight = 480; - -// Rate we capture and encode at, announced to the peer in the 0x74 message. -// -// 16kHz rather than 48kHz because it is what the echo canceller wants. AEC3 -// does its filtering in the 0-8kHz band whatever it is handed; at 48kHz it -// splits the signal into four bands, cancels in the lowest and gates the rest, -// which is more work for a result that is no better on speech. -// -// Costs nothing worth having for speech: Opus codes this as wideband, an 8kHz -// audio bandwidth, which is well past telephone quality and past where voice -// intelligibility stops improving. -// -// The playback direction is NOT this constant - it follows whatever the peer -// announces (see incoming_audio_params()), so a call with an older build that -// still sends 48kHz keeps working. -static const int kCallSampleRate = 16000; - AudioVideoCallController::AudioVideoCallController(QObject *parent) : QObject(parent), callwindow(0), ringdevice(0), is_active(false), @@ -174,6 +153,21 @@ void AudioVideoCallController::startAudioPipeline() { Settings st(config_dbname); QString capDev = st.loadValue("audio.capture_device", "default").toString(); + const CallSettings cs = CallSettings::load(); + + // What we capture and encode at, announced to the peer in the 0x74 + // message. It follows from the codec - see callAudioSampleRate(). + // + // The playback direction is NOT this: it follows whatever the peer + // announces in its own 0x74 (see incoming_audio_params()), so a call with + // a peer that picked a different codec or rate keeps working. + const int sampleRate = callAudioSampleRate(cs.audioCodec); + + // G.722 and A-law spend a fixed number of bits per sample and have no rate + // to ask for; the setting only means anything to Opus. + const int bitrate = callAudioBitrateAdjustable(cs.audioCodec) + ? cs.audioBitrateKbps * 1000 + : 0; m_audioSeq = 0; m_callClock.start(); @@ -188,8 +182,12 @@ void AudioVideoCallController::startAudioPipeline() // below us is doing the job or nobody is, and either way running this as // well would only take a second bite out of the speech - the devices are // opened accordingly, see AudioDevice::captureNative(). + // + // Built at the rate the codec runs at rather than at a fixed one, so that + // choosing a codec does not quietly switch cancellation off: the capture + // side drops a canceller whose rate does not match its own. if (AudioDevice::EchoCancellation() == EchoCancellationOwn) - m_echoCanceller = EchoCancellerPtr(new EchoCanceller(kCallSampleRate)); + m_echoCanceller = EchoCancellerPtr(new EchoCanceller(sampleRate)); else m_echoCanceller.clear(); @@ -205,10 +203,11 @@ void AudioVideoCallController::startAudioPipeline() // encoding right away - the peer's decoder is only ready once it has // processed this 0x74 message, but since the call is carried over TCP, // packets sent immediately after are guaranteed to arrive after it. - skynet->SendAudioParams(m_peer_id, QStringLiteral("opus"), kCallSampleRate); + skynet->SendAudioParams(m_peer_id, cs.audioCodec, (quint16)sampleRate); QMetaObject::invokeMethod(microphone, "configureCapture", Qt::QueuedConnection, - Q_ARG(QString, capDev), Q_ARG(QString, QStringLiteral("opus")), Q_ARG(int, kCallSampleRate)); + Q_ARG(QString, capDev), Q_ARG(QString, cs.audioCodec), + Q_ARG(int, sampleRate), Q_ARG(int, bitrate)); QMetaObject::invokeMethod(microphone, "Capture", Qt::QueuedConnection); // output is configured lazily, once the peer's own 0x74 (see @@ -239,6 +238,7 @@ void AudioVideoCallController::startVideoPipeline() { Settings st(config_dbname); QString camDev = st.loadValue("video.capture_device", QString()).toString(); + const CallSettings cs = CallSettings::load(); // Unlike the audio devices, which fall back to the platform's "default" // device, there is no such name for cameras - so with nothing configured @@ -278,8 +278,11 @@ void AudioVideoCallController::startVideoPipeline() // video_capture_started()). QMetaObject::invokeMethod(cam, "configureCapture", Qt::QueuedConnection, Q_ARG(QString, camDev), - Q_ARG(int, kVideoWidth), Q_ARG(int, kVideoHeight), - Q_ARG(bool, preferCompressed), Q_ARG(bool, vaapiEncoding)); + Q_ARG(int, cs.captureWidth), Q_ARG(int, cs.captureHeight), + Q_ARG(int, cs.captureFrameRate), + Q_ARG(bool, preferCompressed), + Q_ARG(QString, cs.videoCodec), Q_ARG(int, cs.videoBitrateKbps * 1000), + Q_ARG(bool, vaapiEncoding)); QMetaObject::invokeMethod(cam, "Capture", Qt::QueuedConnection); // remotevideo is configured lazily, once the peer's own 0x75 (see |