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xbmc: add PR4475

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Signed-off-by: Stephan Raue <stephan@openelec.tv>
This commit is contained in:
Stephan Raue 2014-03-29 02:25:47 +01:00
parent 909839d12e
commit 0fc44fc909
1 changed files with 443 additions and 0 deletions
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packages/mediacenter/xbmc/patches/xbmc-999.80.013-PR4475.patch Normal file
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@ -0,0 +1,443 @@
From fce3a979731f23732acd047fa7a58e9352a49030 Mon Sep 17 00:00:00 2001
From: popcornmix <popcornmix@gmail.com>
Date: Wed, 26 Mar 2014 00:36:04 +0000
Subject: [PATCH 1/5] [PiSink] Ensure audio buffers use reasonable sized chunks
Previously the sink tried to consume exactly the number of samples required to maintain the desired audio latency.
That has a problem that we reapeatedly get called, and just consume a few samples each call.
As the cost of sending the data to the GPU is quite high, this results in a lot of CPU being used.
This changes the buffering to sleep for a quarter of buffer size, which ensures the samples submitted are always at lease a quarter of the total
Also, now the latency is never more than AUDIO_PLAYBUFFER, so report that directly in GetCacheTotal.
---
xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp | 39 +++++++++++++++++++------------
1 file changed, 24 insertions(+), 15 deletions(-)
diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
index c77fe47..bd6635d 100644
--- a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
+++ b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
@@ -209,8 +209,7 @@ double CAESinkPi::GetCacheTime()
double CAESinkPi::GetCacheTotal()
{
- double audioplus_buffer = AUDIO_PLAYBUFFER;
- return m_sinkbuffer_sec_per_byte * (double)m_sinkbuffer_size + audioplus_buffer;
+ return AUDIO_PLAYBUFFER;
}
unsigned int CAESinkPi::AddPackets(uint8_t *data, unsigned int frames, bool hasAudio, bool blocking)
@@ -224,41 +223,51 @@ unsigned int CAESinkPi::AddPackets(uint8_t *data, unsigned int frames, bool hasA
OMX_BUFFERHEADERTYPE *omx_buffer = NULL;
while (sent < frames)
{
- int timeout = blocking ? 1000 : 0;
-
- // delay compared to maximum we'd like (to keep lag low)
double delay = GetDelay();
- bool too_laggy = delay - AUDIO_PLAYBUFFER > 0.0;
- omx_buffer = too_laggy ? NULL : m_omx_render.GetInputBuffer(timeout);
-
- if (omx_buffer == NULL)
+ double ideal_submission_time = AUDIO_PLAYBUFFER - delay;
+ // ideal amount of audio we'd like submit (to make delay match AUDIO_PLAYBUFFER)
+ int timeout = blocking ? 1000 : 0;
+ int ideal_submission_samples = ideal_submission_time / (m_sinkbuffer_sec_per_byte * m_format.m_frameSize);
+ // if we are almost full then sleep (to avoid repeatedly sending a few samples)
+ bool too_laggy = ideal_submission_time < 0.25 * AUDIO_PLAYBUFFER;
+ int sleeptime = (int)(AUDIO_PLAYBUFFER * 0.25 * 1000.0);
+ if (too_laggy)
{
- if (too_laggy)
+ if (blocking)
{
- Sleep((int)((delay - AUDIO_PLAYBUFFER) * 1000.0));
+ Sleep(sleeptime);
continue;
}
+ break;
+ }
+ omx_buffer = m_omx_render.GetInputBuffer(timeout);
+ if (omx_buffer == NULL)
+ {
if (blocking)
CLog::Log(LOGERROR, "COMXAudio::Decode timeout");
break;
}
- omx_buffer->nFilledLen = std::min(omx_buffer->nAllocLen, (frames - sent) * m_format.m_frameSize);
+ unsigned int space = omx_buffer->nAllocLen / m_format.m_frameSize;
+ unsigned int samples = std::min(std::min(space, (unsigned int)ideal_submission_samples), frames - sent);
+
+ omx_buffer->nFilledLen = samples * m_format.m_frameSize;
omx_buffer->nTimeStamp = ToOMXTime(0);
omx_buffer->nFlags = 0;
memcpy(omx_buffer->pBuffer, (uint8_t *)data + sent * m_format.m_frameSize, omx_buffer->nFilledLen);
- sent += omx_buffer->nFilledLen / m_format.m_frameSize;
+
+ sent += samples;
if (sent == frames)
omx_buffer->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME;
if (delay <= 0.0 && m_submitted)
- CLog::Log(LOGERROR, "%s:%s Underrun (delay:%.2f frames:%d)", CLASSNAME, __func__, delay, frames);
+ CLog::Log(LOGNOTICE, "%s:%s Underrun (delay:%.2f frames:%d)", CLASSNAME, __func__, delay, frames);
omx_err = m_omx_render.EmptyThisBuffer(omx_buffer);
if (omx_err != OMX_ErrorNone)
CLog::Log(LOGERROR, "%s:%s frames=%d err=%x", CLASSNAME, __func__, frames, omx_err);
- m_submitted += omx_buffer->nFilledLen;
+ m_submitted++;
}
return sent;
--
1.8.5.5
From 700384b4b7119751b5a383faad9e1bfb3464a921 Mon Sep 17 00:00:00 2001
From: popcornmix <popcornmix@gmail.com>
Date: Wed, 26 Mar 2014 00:43:59 +0000
Subject: [PATCH 2/5] [PiSink] Add support for float and 32-bit formats
Significant CPU is consumed in converting audio to the sink's format, so add support for common formats to the sink.
Requires update firmware to handle this
---
xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp | 35 +++++++++++++++++--------------
xbmc/cores/AudioEngine/Sinks/AESinkPi.h | 1 -
2 files changed, 19 insertions(+), 17 deletions(-)
diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
index bd6635d..3242b4b 100644
--- a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
+++ b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
@@ -41,7 +41,6 @@
CAEDeviceInfo CAESinkPi::m_info;
CAESinkPi::CAESinkPi() :
- m_sinkbuffer_size(0),
m_sinkbuffer_sec_per_byte(0),
m_Initialized(false),
m_submitted(0)
@@ -84,19 +83,22 @@ bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
m_initDevice = device;
m_initFormat = format;
// setup for a 50ms sink feed from SoftAE
- format.m_dataFormat = AE_FMT_S16NE;
+ if (format.m_dataFormat != AE_FMT_FLOAT && format.m_dataFormat != AE_FMT_S32LE)
+ format.m_dataFormat = AE_FMT_S16LE;
+ unsigned int channels = format.m_channelLayout.Count();
+ unsigned int sample_size = CAEUtil::DataFormatToBits(format.m_dataFormat) >> 3;
+ format.m_frameSize = sample_size * channels;
+ format.m_sampleRate = std::max(8000U, std::min(192000U, format.m_sampleRate));
format.m_frames = format.m_sampleRate * AUDIO_PLAYBUFFER;
- format.m_frameSamples = format.m_channelLayout.Count();
- format.m_frameSize = format.m_frameSamples * (CAEUtil::DataFormatToBits(format.m_dataFormat) >> 3);
- format.m_sampleRate = std::max(8000U, std::min(96000U, format.m_sampleRate));
+ format.m_frameSamples = format.m_frames * channels;
m_format = format;
- m_sinkbuffer_size = format.m_frameSize * format.m_frames * NUM_OMX_BUFFERS;
- m_sinkbuffer_sec_per_byte = 1.0 / (double)(format.m_frameSize * format.m_sampleRate);
+ m_format = format;
+ m_sinkbuffer_sec_per_byte = 1.0 / (double)(m_format.m_frameSize * m_format.m_sampleRate);
CLog::Log(LOGDEBUG, "%s:%s Format:%d Channels:%d Samplerate:%d framesize:%d bufsize:%d bytes/s=%.2f", CLASSNAME, __func__,
- format.m_dataFormat, format.m_channelLayout.Count(), format.m_sampleRate, format.m_frameSize, m_sinkbuffer_size, 1.0/m_sinkbuffer_sec_per_byte);
+ m_format.m_dataFormat, channels, m_format.m_sampleRate, m_format.m_frameSize, m_format.m_frameSize * m_format.m_frames, 1.0/m_sinkbuffer_sec_per_byte);
// This may be called before Application calls g_RBP.Initialise, so call it here too
g_RBP.Initialize();
@@ -113,17 +115,14 @@ bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
m_pcm_input.eNumData = OMX_NumericalDataSigned;
m_pcm_input.eEndian = OMX_EndianLittle;
m_pcm_input.bInterleaved = OMX_TRUE;
- m_pcm_input.nBitPerSample = 16;
- m_pcm_input.ePCMMode = OMX_AUDIO_PCMModeLinear;
- m_pcm_input.nChannels = m_format.m_frameSamples;
+ m_pcm_input.nBitPerSample = sample_size * 8;
+ m_pcm_input.ePCMMode = m_format.m_dataFormat == AE_FMT_FLOAT ? (OMX_AUDIO_PCMMODETYPE)0x8000 : OMX_AUDIO_PCMModeLinear;
+ m_pcm_input.nChannels = channels;
m_pcm_input.nSamplingRate = m_format.m_sampleRate;
- m_pcm_input.eChannelMapping[0] = OMX_AUDIO_ChannelLF;
- m_pcm_input.eChannelMapping[1] = OMX_AUDIO_ChannelRF;
- m_pcm_input.eChannelMapping[2] = OMX_AUDIO_ChannelMax;
omx_err = m_omx_render.SetParameter(OMX_IndexParamAudioPcm, &m_pcm_input);
if (omx_err != OMX_ErrorNone)
- CLog::Log(LOGERROR, "%s::%s - error m_omx_render SetParameter omx_err(0x%08x)", CLASSNAME, __func__, omx_err);
+ CLog::Log(LOGERROR, "%s::%s - error m_omx_render SetParameter in omx_err(0x%08x)", CLASSNAME, __func__, omx_err);
m_omx_render.ResetEos();
@@ -139,7 +138,7 @@ bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
CLog::Log(LOGERROR, "%s:%s - error get OMX_IndexParamPortDefinition (input) omx_err(0x%08x)", CLASSNAME, __func__, omx_err);
port_param.nBufferCountActual = std::max((unsigned int)port_param.nBufferCountMin, (unsigned int)NUM_OMX_BUFFERS);
- port_param.nBufferSize = m_sinkbuffer_size / port_param.nBufferCountActual;
+ port_param.nBufferSize = m_format.m_frameSize * m_format.m_frames / port_param.nBufferCountActual;
omx_err = m_omx_render.SetParameter(OMX_IndexParamPortDefinition, &port_param);
if (omx_err != OMX_ErrorNone)
@@ -295,6 +294,8 @@ void CAESinkPi::EnumerateDevicesEx(AEDeviceInfoList &list, bool force)
m_info.m_channels += AE_CH_FR;
for (unsigned int i=0; i<sizeof PassthroughSampleRates/sizeof *PassthroughSampleRates; i++)
m_info.m_sampleRates.push_back(PassthroughSampleRates[i]);
+ m_info.m_dataFormats.push_back(AE_FMT_FLOAT);
+ m_info.m_dataFormats.push_back(AE_FMT_S32LE);
m_info.m_dataFormats.push_back(AE_FMT_S16LE);
m_info.m_dataFormats.push_back(AE_FMT_AC3);
m_info.m_dataFormats.push_back(AE_FMT_DTS);
@@ -313,6 +314,8 @@ void CAESinkPi::EnumerateDevicesEx(AEDeviceInfoList &list, bool force)
m_info.m_channels += AE_CH_FL;
m_info.m_channels += AE_CH_FR;
m_info.m_sampleRates.push_back(48000);
+ m_info.m_dataFormats.push_back(AE_FMT_FLOAT);
+ m_info.m_dataFormats.push_back(AE_FMT_S32LE);
m_info.m_dataFormats.push_back(AE_FMT_S16LE);
list.push_back(m_info);
diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkPi.h b/xbmc/cores/AudioEngine/Sinks/AESinkPi.h
index 4cca143..6296ecb 100644
--- a/xbmc/cores/AudioEngine/Sinks/AESinkPi.h
+++ b/xbmc/cores/AudioEngine/Sinks/AESinkPi.h
@@ -53,7 +53,6 @@ class CAESinkPi : public IAESink
std::string m_initDevice;
AEAudioFormat m_initFormat;
AEAudioFormat m_format;
- unsigned int m_sinkbuffer_size; ///< total size of the buffer
double m_sinkbuffer_sec_per_byte;
static CAEDeviceInfo m_info;
bool m_Initialized;
--
1.8.5.5
From 189111a39ca475b763fb607349973c6154a318f3 Mon Sep 17 00:00:00 2001
From: popcornmix <popcornmix@gmail.com>
Date: Wed, 26 Mar 2014 01:06:24 +0000
Subject: [PATCH 3/5] [PiSink] Fix multichannel and passthough
The current Pi sink has random speaker allocation and plays noise when passthrough is enabled.
This adds messages to gpu to describe speaker layout for multichannel, and signal when passthrough is used
fixing both these issues
---
xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp | 118 ++++++++++++++++++++++++++----
1 file changed, 104 insertions(+), 14 deletions(-)
diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
index 3242b4b..08633cf 100644
--- a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
+++ b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
@@ -65,20 +65,112 @@ void CAESinkPi::SetAudioDest()
CLog::Log(LOGERROR, "%s::%s - m_omx_render.SetConfig omx_err(0x%08x)", CLASSNAME, __func__, omx_err);
}
-bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
+static void SetAudioProps(bool stream_channels, uint32_t channel_map)
{
- char response[80];
- /* if we are raw need to let gpu know */
- if (AE_IS_RAW(format.m_dataFormat))
+ char command[80], response[80];
+
+ sprintf(command, "hdmi_stream_channels %d", stream_channels ? 1 : 0);
+ vc_gencmd(response, sizeof response, command);
+
+ sprintf(command, "hdmi_channel_map 0x%08x", channel_map);
+ vc_gencmd(response, sizeof response, command);
+
+ CLog::Log(LOGDEBUG, "%s:%s hdmi_stream_channels %d hdmi_channel_map %08x", CLASSNAME, __func__, stream_channels, channel_map);
+}
+
+static uint32_t GetChannelMap(AEAudioFormat &format, bool passthrough)
+{
+ unsigned int channels = format.m_channelLayout.Count();
+ uint32_t channel_map = 0;
+ if (passthrough)
+ return 0;
+
+ static const unsigned char map_normal[] =
{
- vc_gencmd(response, sizeof response, "hdmi_stream_channels 1");
- m_passthrough = true;
- }
- else
+ 0, //AE_CH_RAW ,
+ 1, //AE_CH_FL
+ 2, //AE_CH_FR
+ 4, //AE_CH_FC
+ 3, //AE_CH_LFE
+ 7, //AE_CH_BL
+ 8, //AE_CH_BR
+ 1, //AE_CH_FLOC,
+ 2, //AE_CH_FROC,
+ 4, //AE_CH_BC,
+ 5, //AE_CH_SL
+ 6, //AE_CH_SR
+ };
+ static const unsigned char map_back[] =
+ {
+ 0, //AE_CH_RAW ,
+ 1, //AE_CH_FL
+ 2, //AE_CH_FR
+ 4, //AE_CH_FC
+ 3, //AE_CH_LFE
+ 5, //AE_CH_BL
+ 6, //AE_CH_BR
+ 1, //AE_CH_FLOC,
+ 2, //AE_CH_FROC,
+ 4, //AE_CH_BC,
+ 5, //AE_CH_SL
+ 6, //AE_CH_SR
+ };
+ const unsigned char *map = map_normal;
+ // According to CEA-861-D only RL and RR are known. In case of a format having SL and SR channels
+ // but no BR BL channels, we use the wide map in order to open only the num of channels really
+ // needed.
+ if (format.m_channelLayout.HasChannel(AE_CH_BL) && !format.m_channelLayout.HasChannel(AE_CH_SL))
+ map = map_back;
+
+ for (unsigned int i = 0; i < channels; ++i)
{
- vc_gencmd(response, sizeof response, "hdmi_stream_channels 0");
- m_passthrough = false;
+ AEChannel c = format.m_channelLayout[i];
+ unsigned int chan = 0;
+ if ((unsigned int)c < sizeof map_normal / sizeof *map_normal)
+ chan = map[(unsigned int)c];
+ if (chan > 0)
+ channel_map |= (chan-1) << (3*i);
}
+ // These numbers are from Table 28 Audio InfoFrame Data byte 4 of CEA 861
+ // and describe the speaker layout
+ static const uint8_t cea_map[] = {
+ 0xff, // 0
+ 0xff, // 1
+ 0x00, // 2.0
+ 0x02, // 3.0
+ 0x08, // 4.0
+ 0x0a, // 5.0
+ 0xff, // 6
+ 0x12, // 7.0
+ 0xff, // 8
+ };
+ static const uint8_t cea_map_lfe[] = {
+ 0xff, // 0
+ 0xff, // 1
+ 0xff, // 2
+ 0x01, // 2.1
+ 0x03, // 3.1
+ 0x09, // 4.1
+ 0x0b, // 5.1
+ 0xff, // 7
+ 0x13, // 7.1
+ };
+ uint8_t cea = format.m_channelLayout.HasChannel(AE_CH_LFE) ? cea_map_lfe[channels] : cea_map[channels];
+ if (cea == 0xff)
+ CLog::Log(LOGERROR, "%s::%s - Unexpected CEA mapping %d,%d", CLASSNAME, __func__, format.m_channelLayout.HasChannel(AE_CH_LFE), channels);
+
+ channel_map |= cea << 24;
+
+ return channel_map;
+}
+
+bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
+{
+ // This may be called before Application calls g_RBP.Initialise, so call it here too
+ g_RBP.Initialize();
+
+ /* if we are raw need to let gpu know */
+ m_passthrough = AE_IS_RAW(format.m_dataFormat);
m_initDevice = device;
m_initFormat = format;
@@ -92,7 +184,7 @@ bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
format.m_frames = format.m_sampleRate * AUDIO_PLAYBUFFER;
format.m_frameSamples = format.m_frames * channels;
- m_format = format;
+ SetAudioProps(m_passthrough, GetChannelMap(format, m_passthrough));
m_format = format;
m_sinkbuffer_sec_per_byte = 1.0 / (double)(m_format.m_frameSize * m_format.m_sampleRate);
@@ -100,9 +192,6 @@ bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
CLog::Log(LOGDEBUG, "%s:%s Format:%d Channels:%d Samplerate:%d framesize:%d bufsize:%d bytes/s=%.2f", CLASSNAME, __func__,
m_format.m_dataFormat, channels, m_format.m_sampleRate, m_format.m_frameSize, m_format.m_frameSize * m_format.m_frames, 1.0/m_sinkbuffer_sec_per_byte);
- // This may be called before Application calls g_RBP.Initialise, so call it here too
- g_RBP.Initialize();
-
CLog::Log(LOGDEBUG, "%s:%s", CLASSNAME, __func__);
OMX_ERRORTYPE omx_err = OMX_ErrorNone;
@@ -160,6 +249,7 @@ bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
void CAESinkPi::Deinitialize()
{
CLog::Log(LOGDEBUG, "%s:%s", CLASSNAME, __func__);
+ SetAudioProps(false, 0);
if (m_Initialized)
{
m_omx_render.FlushAll();
--
1.8.5.5
From d875fb7bd03e26f2d4780daf1d0aa809a641ed27 Mon Sep 17 00:00:00 2001
From: popcornmix <popcornmix@gmail.com>
Date: Wed, 26 Mar 2014 00:44:53 +0000
Subject: [PATCH 4/5] [PiSink] Increase Pi Sink's buffering
We do get underruns and breakups at 50ms latency. Increase to 100ms.
---
xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
index 08633cf..5c9bb8d 100644
--- a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
+++ b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
@@ -34,7 +34,7 @@
#define CLASSNAME "CAESinkPi"
#define NUM_OMX_BUFFERS 2
-#define AUDIO_PLAYBUFFER (1.0/20.0)
+#define AUDIO_PLAYBUFFER (0.1) // 100ms
static const unsigned int PassthroughSampleRates[] = { 8000, 11025, 16000, 22050, 24000, 32000, 41400, 48000, 88200, 96000, 176400, 192000 };
--
1.8.5.5
From ba0313c1a37d7d002f145f80b1526c1cd4fc7152 Mon Sep 17 00:00:00 2001
From: popcornmix <popcornmix@gmail.com>
Date: Thu, 27 Mar 2014 00:22:54 +0000
Subject: [PATCH 5/5] [PiSink] Analogue only supports stereo
---
xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
index 5c9bb8d..9ce00e3 100644
--- a/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
+++ b/xbmc/cores/AudioEngine/Sinks/AESinkPi.cpp
@@ -174,6 +174,11 @@ bool CAESinkPi::Initialize(AEAudioFormat &format, std::string &device)
m_initDevice = device;
m_initFormat = format;
+
+ // analogue only supports stereo
+ if (CSettings::Get().GetString("audiooutput.audiodevice") == "PI:Analogue")
+ format.m_channelLayout = AE_CH_LAYOUT_2_0;
+
// setup for a 50ms sink feed from SoftAE
if (format.m_dataFormat != AE_FMT_FLOAT && format.m_dataFormat != AE_FMT_S32LE)
format.m_dataFormat = AE_FMT_S16LE;
--
1.8.5.5