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New upstream version 21.0.2+dfsg1

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This commit is contained in:
Sebastian Ramacher 2018-02-19 20:54:37 +01:00
parent 1f1bbb3518
commit baafb6325b
706 changed files with 49633 additions and 5044 deletions
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49
plugins/coreaudio-encoder/encoder.cpp
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@ -546,7 +546,7 @@ static void *aac_create(obs_data_t *settings, obs_encoder_t *encoder)
UInt32 rate_control = kAudioCodecBitRateControlMode_Constant;
if (obs_data_get_bool(settings, "allow he-aac")) {
if (obs_data_get_bool(settings, "allow he-aac") && ca->channels != 3) {
ca->allowed_formats = &aac_formats;
} else {
ca->allowed_formats = &aac_lc_formats;
@ -601,6 +601,35 @@ static void *aac_create(obs_data_t *settings, obs_encoder_t *encoder)
kAudioConverterCurrentOutputStreamDescription,
&size, &out));
/*
* Fix channel map differences between CoreAudio AAC, FFmpeg, Wav
* New channel mappings below assume 2.1, 4.1, 5.1, 7.1 resp.
*/
if (ca->channels == 3) {
SInt32 channelMap3[3] = {2, 0, 1};
AudioConverterSetProperty(ca->converter,
kAudioConverterChannelMap,
sizeof(channelMap3), channelMap3);
} else if (ca->channels == 5) {
SInt32 channelMap5[5] = {2, 0, 1, 3, 4};
AudioConverterSetProperty(ca->converter,
kAudioConverterChannelMap,
sizeof(channelMap5), channelMap5);
} else if (ca->channels == 6) {
SInt32 channelMap6[6] = {2, 0, 1, 4, 5, 3};
AudioConverterSetProperty(ca->converter,
kAudioConverterChannelMap,
sizeof(channelMap6), channelMap6);
} else if (ca->channels == 8) {
SInt32 channelMap8[8] = {2, 0, 1, 6, 7, 4, 5, 3};
AudioConverterSetProperty(ca->converter,
kAudioConverterChannelMap,
sizeof(channelMap8), channelMap8);
}
ca->in_frame_size = in.mBytesPerFrame;
ca->in_packets = out.mFramesPerPacket / in.mFramesPerPacket;
ca->in_bytes_required = ca->in_packets * ca->in_frame_size;
@ -877,10 +906,9 @@ static bool aac_extra_data(void *data, uint8_t **extra_data, size_t *size)
}
static asbd_builder fill_common_asbd_fields(asbd_builder builder,
bool in=false)
bool in=false, UInt32 channels=2)
{
UInt32 bytes_per_frame = 8;
UInt32 channels = 2;
UInt32 bytes_per_frame = sizeof(float) * channels;
UInt32 bits_per_channel = bytes_per_frame / channels * 8;
builder.channels_per_frame(channels);
@ -908,9 +936,9 @@ static AudioStreamBasicDescription get_default_in_asbd()
.asbd;
}
static asbd_builder get_default_out_asbd_builder()
static asbd_builder get_default_out_asbd_builder(UInt32 channels)
{
return fill_common_asbd_fields(asbd_builder())
return fill_common_asbd_fields(asbd_builder(), false, channels)
.sample_rate(44100);
}
@ -975,7 +1003,7 @@ static bool find_best_match(DStr &log, ca_encoder *ca, UInt32 bitrate,
log_to_dstr(log, ca, "Trying %s (0x%x)\n",
format_id_to_str(format_id), format_id);
auto out = get_default_out_asbd_builder()
auto out = get_default_out_asbd_builder(2)
.format_id(format_id)
.asbd;
@ -1206,6 +1234,11 @@ static vector<UInt32> get_bitrates(DStr &log, ca_encoder *ca,
Float64 samplerate)
{
vector<UInt32> bitrates;
struct obs_audio_info aoi;
int channels;
obs_get_audio_info(&aoi);
channels = get_audio_channels(aoi.speakers);
auto handle_bitrate = [&](UInt32 bitrate)
{
@ -1240,7 +1273,7 @@ static vector<UInt32> get_bitrates(DStr &log, ca_encoder *ca,
static_cast<uint32_t>(format_id),
samplerate);
auto out = get_default_out_asbd_builder()
auto out = get_default_out_asbd_builder(channels)
.format_id(format_id)
.sample_rate(samplerate)
.asbd;