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https://github.com/LibreELEC/LibreELEC.tv.git
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Revert "xbmc: add upstream patch"
This reverts commit 8b0cd8daf38588935765726413b6c126fe1bf84a. not mainline. ref #2994
This commit is contained in:
Stefan Saraev
parent
c75b77c934
commit
d29f822d46
@ -1,402 +0,0 @@
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From 74b06eccc06a0b4bd00114b20c929dfa1b73c783 Mon Sep 17 00:00:00 2001
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From: popcornmix <popcornmix@gmail.com>
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Date: Sat, 1 Mar 2014 14:24:08 +0000
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Subject: [PATCH] [omxplayer] Allow small audio packets to be concatenated to
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make better use of audio fifo
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Some audio codecs produce small packets which causes a high overhead when submitting to GPU, and doesn't make full use of GPU side buffering.
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TrueHD in particular can produce packets with 40 samples (so 1200 packets per second) which causes very high overhead.
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What this aims to do is to concatenate audio packets until they approach the ideal audio packet size,
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and then deal with the awkardness of concatenated planar formats.
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---
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xbmc/cores/omxplayer/OMXAudio.cpp | 67 ++++++++++++++++++++++---------
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xbmc/cores/omxplayer/OMXAudio.h | 3 +-
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xbmc/cores/omxplayer/OMXAudioCodecOMX.cpp | 62 ++++++++++++++++++----------
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xbmc/cores/omxplayer/OMXAudioCodecOMX.h | 8 +++-
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xbmc/cores/omxplayer/OMXPlayerAudio.cpp | 9 +++--
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5 files changed, 100 insertions(+), 49 deletions(-)
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diff --git a/xbmc/cores/omxplayer/OMXAudio.cpp b/xbmc/cores/omxplayer/OMXAudio.cpp
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index 3f91e89..70177db 100644
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--- a/xbmc/cores/omxplayer/OMXAudio.cpp
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+++ b/xbmc/cores/omxplayer/OMXAudio.cpp
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@@ -43,6 +43,10 @@
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using namespace std;
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+// the size of the audio_render output port buffers
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+#define AUDIO_DECODE_OUTPUT_BUFFER (32*1024)
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+static const char rounded_up_channels_shift[] = {0,0,1,2,2,3,3,3,3};
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+
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static const uint16_t AC3Bitrates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640};
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static const uint16_t AC3FSCod [] = {48000, 44100, 32000, 0};
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@@ -61,6 +65,7 @@
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m_Passthrough (false ),
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m_HWDecode (false ),
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m_BytesPerSec (0 ),
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+ m_InputBytesPerSec(0 ),
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m_BufferLen (0 ),
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m_ChunkLen (0 ),
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m_InputChannels (0 ),
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@@ -491,11 +496,15 @@ bool COMXAudio::Initialize(AEAudioFormat format, OMXClock *clock, CDVDStreamInfo
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m_SampleRate = m_format.m_sampleRate;
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m_BitsPerSample = CAEUtil::DataFormatToBits(m_format.m_dataFormat);
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- m_BufferLen = m_BytesPerSec = m_format.m_sampleRate * (16 >> 3) * m_InputChannels;
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- m_BufferLen *= AUDIO_BUFFER_SECONDS;
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+ m_BytesPerSec = m_SampleRate * 2 << rounded_up_channels_shift[m_InputChannels];
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+ m_BufferLen = m_BytesPerSec * AUDIO_BUFFER_SECONDS;
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+ m_InputBytesPerSec = m_SampleRate * m_BitsPerSample * m_InputChannels >> 3;
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+
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+ // should be big enough that common formats (e.g. 6 channel DTS) fit in a single packet.
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+ // we don't mind less common formats being split (e.g. ape/wma output large frames)
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// the audio_decode output buffer size is 32K, and typically we convert from
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- // 6 channel 32bpp float to 8 channel 16bpp in, so a full 48K input buffer will fit the outbut buffer
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- m_ChunkLen = 48*1024;
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+ // 6 channel 32bpp float to 8 channel 16bpp in, so a full 48K input buffer will fit the output buffer
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+ m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (m_InputChannels * m_BitsPerSample) >> (rounded_up_channels_shift[m_InputChannels] + 4);
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m_wave_header.Samples.wSamplesPerBlock = 0;
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m_wave_header.Format.nChannels = m_InputChannels;
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@@ -683,7 +692,7 @@ bool COMXAudio::Initialize(AEAudioFormat format, OMXClock *clock, CDVDStreamInfo
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m_maxLevel = 0.0f;
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CLog::Log(LOGDEBUG, "COMXAudio::Initialize Input bps %d samplerate %d channels %d buffer size %d bytes per second %d",
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- (int)m_pcm_input.nBitPerSample, (int)m_pcm_input.nSamplingRate, (int)m_pcm_input.nChannels, m_BufferLen, m_BytesPerSec);
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+ (int)m_pcm_input.nBitPerSample, (int)m_pcm_input.nSamplingRate, (int)m_pcm_input.nChannels, m_BufferLen, m_InputBytesPerSec);
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PrintPCM(&m_pcm_input, std::string("input"));
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CLog::Log(LOGDEBUG, "COMXAudio::Initialize device passthrough %d hwdecode %d",
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m_Passthrough, m_HWDecode);
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@@ -866,11 +875,11 @@ bool COMXAudio::ApplyVolume(void)
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//***********************************************************************************************
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unsigned int COMXAudio::AddPackets(const void* data, unsigned int len)
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{
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- return AddPackets(data, len, 0, 0);
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+ return AddPackets(data, len, 0, 0, 0);
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}
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//***********************************************************************************************
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-unsigned int COMXAudio::AddPackets(const void* data, unsigned int len, double dts, double pts)
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+unsigned int COMXAudio::AddPackets(const void* data, unsigned int len, double dts, double pts, unsigned int frame_size)
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{
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CSingleLock lock (m_critSection);
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@@ -917,24 +926,40 @@ unsigned int COMXAudio::AddPackets(const void* data, unsigned int len, double dt
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omx_buffer->nOffset = 0;
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omx_buffer->nFlags = 0;
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+ // we want audio_decode output buffer size to be no more than AUDIO_DECODE_OUTPUT_BUFFER.
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+ // it will be 16-bit and rounded up to next power of 2 in channels
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+ unsigned int max_buffer = AUDIO_DECODE_OUTPUT_BUFFER * (m_InputChannels * m_BitsPerSample) >> (rounded_up_channels_shift[m_InputChannels] + 4);
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+
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unsigned int remaining = demuxer_samples-demuxer_samples_sent;
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- unsigned int samples_space = omx_buffer->nAllocLen/pitch;
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+ unsigned int samples_space = std::min(max_buffer, omx_buffer->nAllocLen)/pitch;
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unsigned int samples = std::min(remaining, samples_space);
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omx_buffer->nFilledLen = samples * pitch;
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- if (samples < demuxer_samples && m_BitsPerSample==32 && !(m_Passthrough || m_HWDecode))
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+ unsigned int frames = frame_size ? len/frame_size:0;
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+ if ((samples < demuxer_samples || frames > 1) && m_BitsPerSample==32 && !(m_Passthrough || m_HWDecode))
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{
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- uint8_t *dst = omx_buffer->pBuffer;
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- uint8_t *src = demuxer_content + demuxer_samples_sent * (m_BitsPerSample >> 3);
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- // we need to extract samples from planar audio, so the copying needs to be done per plane
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- for (int i=0; i<(int)m_InputChannels; i++)
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- {
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- memcpy(dst, src, omx_buffer->nFilledLen / m_InputChannels);
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- dst += omx_buffer->nFilledLen / m_InputChannels;
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- src += demuxer_samples * (m_BitsPerSample >> 3);
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- }
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- assert(dst <= omx_buffer->pBuffer + m_ChunkLen);
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+ const unsigned int sample_pitch = m_BitsPerSample >> 3;
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+ const unsigned int frame_samples = frame_size / pitch;
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+ const unsigned int plane_size = frame_samples * sample_pitch;
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+ const unsigned int out_plane_size = samples * sample_pitch;
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+ //CLog::Log(LOGDEBUG, "%s::%s samples:%d/%d ps:%d ops:%d fs:%d pitch:%d filled:%d frames=%d", CLASSNAME, __func__, samples, demuxer_samples, plane_size, out_plane_size, frame_size, pitch, omx_buffer->nFilledLen, frames);
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+ for (unsigned int sample = 0; sample < samples; )
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+ {
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+ unsigned int frame = (demuxer_samples_sent + sample) / frame_samples;
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+ unsigned int sample_in_frame = (demuxer_samples_sent + sample) - frame * frame_samples;
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+ int out_remaining = std::min(std::min(frame_samples - sample_in_frame, samples), samples-sample);
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+ uint8_t *src = demuxer_content + frame*frame_size + sample_in_frame * sample_pitch;
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+ uint8_t *dst = (uint8_t *)omx_buffer->pBuffer + sample * sample_pitch;
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+ for (unsigned int channel = 0; channel < m_InputChannels; channel++)
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+ {
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+ //CLog::Log(LOGDEBUG, "%s::%s copy(%d,%d,%d) (s:%d f:%d sin:%d c:%d)", CLASSNAME, __func__, dst-(uint8_t *)omx_buffer->pBuffer, src-demuxer_content, out_remaining, sample, frame, sample_in_frame, channel);
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+ memcpy(dst, src, out_remaining * sample_pitch);
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+ src += plane_size;
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+ dst += out_plane_size;
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+ }
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+ sample += out_remaining;
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+ }
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}
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else
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{
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@@ -1103,7 +1128,9 @@ float COMXAudio::GetCacheTime()
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float COMXAudio::GetCacheTotal()
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{
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- return m_BytesPerSec ? (float)m_BufferLen / (float)m_BytesPerSec : 0.0f;
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+ float audioplus_buffer = m_SampleRate ? 0.0f : 32.0f * 512.0f / m_SampleRate;
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+ float input_buffer = (float)m_omx_decoder.GetInputBufferSize() / (float)m_InputBytesPerSec;
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+ return AUDIO_BUFFER_SECONDS + input_buffer + audioplus_buffer;
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}
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//***********************************************************************************************
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diff --git a/xbmc/cores/omxplayer/OMXAudio.h b/xbmc/cores/omxplayer/OMXAudio.h
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index f6bc023..7eba151 100644
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--- a/xbmc/cores/omxplayer/OMXAudio.h
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+++ b/xbmc/cores/omxplayer/OMXAudio.h
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@@ -66,7 +66,7 @@ class COMXAudio
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~COMXAudio();
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unsigned int AddPackets(const void* data, unsigned int len);
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- unsigned int AddPackets(const void* data, unsigned int len, double dts, double pts);
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+ unsigned int AddPackets(const void* data, unsigned int len, double dts, double pts, unsigned int frame_size);
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unsigned int GetSpace();
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bool Deinitialize();
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@@ -114,6 +114,7 @@ class COMXAudio
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bool m_Passthrough;
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bool m_HWDecode;
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unsigned int m_BytesPerSec;
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+ unsigned int m_InputBytesPerSec;
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unsigned int m_BufferLen;
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unsigned int m_ChunkLen;
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unsigned int m_InputChannels;
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diff --git a/xbmc/cores/omxplayer/OMXAudioCodecOMX.cpp b/xbmc/cores/omxplayer/OMXAudioCodecOMX.cpp
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index 5503a0e..cc6558a 100644
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--- a/xbmc/cores/omxplayer/OMXAudioCodecOMX.cpp
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+++ b/xbmc/cores/omxplayer/OMXAudioCodecOMX.cpp
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@@ -26,10 +26,15 @@
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#include "cores/AudioEngine/Utils/AEUtil.h"
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+// the size of the audio_render output port buffers
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+#define AUDIO_DECODE_OUTPUT_BUFFER (32*1024)
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+static const char rounded_up_channels_shift[] = {0,0,1,2,2,3,3,3,3};
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+
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COMXAudioCodecOMX::COMXAudioCodecOMX()
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{
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m_pBufferOutput = NULL;
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m_iBufferOutputAlloced = 0;
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+ m_iBufferOutputUsed = 0;
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m_pCodecContext = NULL;
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m_pConvert = NULL;
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@@ -37,6 +42,7 @@
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m_channels = 0;
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m_pFrame1 = NULL;
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+ m_frameSize = 0;
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m_bGotFrame = false;
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m_iSampleFormat = AV_SAMPLE_FMT_NONE;
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m_desiredSampleFormat = AV_SAMPLE_FMT_NONE;
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@@ -47,6 +53,7 @@
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m_dllAvUtil.av_free(m_pBufferOutput);
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m_pBufferOutput = NULL;
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m_iBufferOutputAlloced = 0;
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+ m_iBufferOutputUsed = 0;
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Dispose();
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}
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@@ -132,7 +139,7 @@ void COMXAudioCodecOMX::Dispose()
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m_bGotFrame = false;
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}
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-int COMXAudioCodecOMX::Decode(BYTE* pData, int iSize)
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+int COMXAudioCodecOMX::Decode(BYTE* pData, int iSize, double dts, double pts)
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{
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int iBytesUsed, got_frame;
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if (!m_pCodecContext) return -1;
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@@ -167,10 +174,15 @@ int COMXAudioCodecOMX::Decode(BYTE* pData, int iSize)
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}
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m_bGotFrame = true;
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+ if (!m_iBufferOutputUsed)
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+ {
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+ m_dts = dts;
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+ m_pts = pts;
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+ }
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return iBytesUsed;
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}
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-int COMXAudioCodecOMX::GetData(BYTE** dst)
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+int COMXAudioCodecOMX::GetData(BYTE** dst, double &dts, double &pts)
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{
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if (!m_bGotFrame)
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return 0;
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@@ -179,13 +191,12 @@ int COMXAudioCodecOMX::GetData(BYTE** dst)
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int inputSize = m_dllAvUtil.av_samples_get_buffer_size(&inLineSize, m_pCodecContext->channels, m_pFrame1->nb_samples, m_pCodecContext->sample_fmt, 0);
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/* output audio will be packed */
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int outputSize = m_dllAvUtil.av_samples_get_buffer_size(&outLineSize, m_pCodecContext->channels, m_pFrame1->nb_samples, m_desiredSampleFormat, 1);
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- bool cont = !m_pFrame1->data[1] || (m_pFrame1->data[1] == m_pFrame1->data[0] + inLineSize && inLineSize == outLineSize && inLineSize * m_pCodecContext->channels == inputSize);
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- if (m_iBufferOutputAlloced < outputSize)
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+ if (m_iBufferOutputAlloced < m_iBufferOutputUsed + outputSize)
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{
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m_dllAvUtil.av_free(m_pBufferOutput);
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- m_pBufferOutput = (BYTE*)m_dllAvUtil.av_malloc(outputSize + FF_INPUT_BUFFER_PADDING_SIZE);
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- m_iBufferOutputAlloced = outputSize;
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+ m_pBufferOutput = (BYTE*)m_dllAvUtil.av_malloc(m_iBufferOutputUsed + outputSize + FF_INPUT_BUFFER_PADDING_SIZE);
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+ m_iBufferOutputAlloced = m_iBufferOutputUsed + outputSize;
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}
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*dst = m_pBufferOutput;
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@@ -217,7 +228,7 @@ int COMXAudioCodecOMX::GetData(BYTE** dst)
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/* use unaligned flag to keep output packed */
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uint8_t *out_planes[m_pCodecContext->channels];
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- if(m_dllAvUtil.av_samples_fill_arrays(out_planes, NULL, m_pBufferOutput, m_pCodecContext->channels, m_pFrame1->nb_samples, m_desiredSampleFormat, 1) < 0 ||
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+ if(m_dllAvUtil.av_samples_fill_arrays(out_planes, NULL, m_pBufferOutput + m_iBufferOutputUsed, m_pCodecContext->channels, m_pFrame1->nb_samples, m_desiredSampleFormat, 1) < 0 ||
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m_dllSwResample.swr_convert(m_pConvert, out_planes, m_pFrame1->nb_samples, (const uint8_t **)m_pFrame1->data, m_pFrame1->nb_samples) < 0)
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{
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CLog::Log(LOGERROR, "COMXAudioCodecOMX::Decode - Unable to convert format %d to %d", (int)m_pCodecContext->sample_fmt, m_desiredSampleFormat);
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@@ -226,35 +237,42 @@ int COMXAudioCodecOMX::GetData(BYTE** dst)
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}
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else
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{
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- /* if it is already contiguous, just return decoded frame */
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- if (cont)
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- {
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- *dst = m_pFrame1->data[0];
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- }
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- else
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+ /* copy to a contiguous buffer */
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+ uint8_t *out_planes[m_pCodecContext->channels];
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+ if (m_dllAvUtil.av_samples_fill_arrays(out_planes, NULL, m_pBufferOutput + m_iBufferOutputUsed, m_pCodecContext->channels, m_pFrame1->nb_samples, m_desiredSampleFormat, 1) < 0 ||
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+ m_dllAvUtil.av_samples_copy(out_planes, m_pFrame1->data, 0, 0, m_pFrame1->nb_samples, m_pCodecContext->channels, m_desiredSampleFormat) < 0 )
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{
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- /* copy to a contiguous buffer */
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- uint8_t *out_planes[m_pCodecContext->channels];
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- if (m_dllAvUtil.av_samples_fill_arrays(out_planes, NULL, m_pBufferOutput, m_pCodecContext->channels, m_pFrame1->nb_samples, m_desiredSampleFormat, 1) < 0 ||
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- m_dllAvUtil.av_samples_copy(out_planes, m_pFrame1->data, 0, 0, m_pFrame1->nb_samples, m_pCodecContext->channels, m_desiredSampleFormat) < 0 )
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- {
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- outputSize = 0;
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- }
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+ outputSize = 0;
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}
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}
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+ int desired_size = AUDIO_DECODE_OUTPUT_BUFFER * (m_pCodecContext->channels * GetBitsPerSample()) >> (rounded_up_channels_shift[m_pCodecContext->channels] + 4);
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if (m_bFirstFrame)
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{
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- CLog::Log(LOGDEBUG, "COMXAudioCodecOMX::GetData size=%d/%d line=%d/%d cont=%d buf=%p", inputSize, outputSize, inLineSize, outLineSize, cont, *dst);
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+ CLog::Log(LOGDEBUG, "COMXAudioCodecOMX::GetData size=%d/%d line=%d/%d buf=%p, desired=%d", inputSize, outputSize, inLineSize, outLineSize, *dst, desired_size);
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m_bFirstFrame = false;
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}
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- return outputSize;
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+ m_iBufferOutputUsed += outputSize;
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+
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+ // if next buffer submitted won't fit then flush it out
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+ if (m_iBufferOutputUsed + outputSize > desired_size || (m_frameSize && (int)m_frameSize != outputSize))
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+ {
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+ int ret = m_iBufferOutputUsed;
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+ m_bGotFrame = false;
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+ m_iBufferOutputUsed = 0;
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+ m_frameSize = outputSize;
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+ dts = m_dts;
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+ pts = m_pts;
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+ return ret;
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+ }
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+ return 0;
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}
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void COMXAudioCodecOMX::Reset()
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{
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if (m_pCodecContext) m_dllAvCodec.avcodec_flush_buffers(m_pCodecContext);
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m_bGotFrame = false;
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+ m_iBufferOutputUsed = 0;
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}
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int COMXAudioCodecOMX::GetChannels()
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diff --git a/xbmc/cores/omxplayer/OMXAudioCodecOMX.h b/xbmc/cores/omxplayer/OMXAudioCodecOMX.h
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index 343465c..4dc2371 100644
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--- a/xbmc/cores/omxplayer/OMXAudioCodecOMX.h
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+++ b/xbmc/cores/omxplayer/OMXAudioCodecOMX.h
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@@ -36,8 +36,8 @@ class COMXAudioCodecOMX
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virtual ~COMXAudioCodecOMX();
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bool Open(CDVDStreamInfo &hints);
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void Dispose();
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- int Decode(BYTE* pData, int iSize);
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- int GetData(BYTE** dst);
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+ int Decode(BYTE* pData, int iSize, double dts, double pts);
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+ int GetData(BYTE** dst, double &dts, double &pts);
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void Reset();
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int GetChannels();
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uint64_t GetChannelMap();
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@@ -45,6 +45,7 @@ class COMXAudioCodecOMX
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int GetBitsPerSample();
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static const char* GetName() { return "FFmpeg"; }
|
||||
int GetBitRate();
|
||||
+ unsigned int GetFrameSize() { return m_frameSize; }
|
||||
|
||||
protected:
|
||||
AVCodecContext* m_pCodecContext;
|
||||
@@ -55,6 +56,7 @@ class COMXAudioCodecOMX
|
||||
AVFrame* m_pFrame1;
|
||||
|
||||
BYTE *m_pBufferOutput;
|
||||
+ int m_iBufferOutputUsed;
|
||||
int m_iBufferOutputAlloced;
|
||||
|
||||
bool m_bOpenedCodec;
|
||||
@@ -63,6 +65,8 @@ class COMXAudioCodecOMX
|
||||
|
||||
bool m_bFirstFrame;
|
||||
bool m_bGotFrame;
|
||||
+ unsigned int m_frameSize;
|
||||
+ double m_dts, m_pts;
|
||||
DllAvCodec m_dllAvCodec;
|
||||
DllAvUtil m_dllAvUtil;
|
||||
DllSwResample m_dllSwResample;
|
||||
diff --git a/xbmc/cores/omxplayer/OMXPlayerAudio.cpp b/xbmc/cores/omxplayer/OMXPlayerAudio.cpp
|
||||
index 7d9081c..53336cb 100644
|
||||
--- a/xbmc/cores/omxplayer/OMXPlayerAudio.cpp
|
||||
+++ b/xbmc/cores/omxplayer/OMXPlayerAudio.cpp
|
||||
@@ -228,9 +228,10 @@ bool OMXPlayerAudio::Decode(DemuxPacket *pkt, bool bDropPacket)
|
||||
|
||||
if(!OMX_IS_RAW(m_format.m_dataFormat) && !bDropPacket)
|
||||
{
|
||||
+ double dts = pkt->dts, pts=pkt->pts;
|
||||
while(!m_bStop && data_len > 0)
|
||||
{
|
||||
- int len = m_pAudioCodec->Decode((BYTE *)data_dec, data_len);
|
||||
+ int len = m_pAudioCodec->Decode((BYTE *)data_dec, data_len, dts, pts);
|
||||
if( (len < 0) || (len > data_len) )
|
||||
{
|
||||
m_pAudioCodec->Reset();
|
||||
@@ -241,7 +242,7 @@ bool OMXPlayerAudio::Decode(DemuxPacket *pkt, bool bDropPacket)
|
||||
data_len -= len;
|
||||
|
||||
uint8_t *decoded;
|
||||
- int decoded_size = m_pAudioCodec->GetData(&decoded);
|
||||
+ int decoded_size = m_pAudioCodec->GetData(&decoded, dts, pts);
|
||||
|
||||
if(decoded_size <=0)
|
||||
continue;
|
||||
@@ -275,7 +276,7 @@ bool OMXPlayerAudio::Decode(DemuxPacket *pkt, bool bDropPacket)
|
||||
if(m_silence)
|
||||
memset(decoded, 0x0, decoded_size);
|
||||
|
||||
- ret = m_omxAudio.AddPackets(decoded, decoded_size, m_audioClock, m_audioClock);
|
||||
+ ret = m_omxAudio.AddPackets(decoded, decoded_size, dts, pts, m_pAudioCodec->GetFrameSize());
|
||||
|
||||
if(ret != decoded_size)
|
||||
{
|
||||
@@ -313,7 +314,7 @@ bool OMXPlayerAudio::Decode(DemuxPacket *pkt, bool bDropPacket)
|
||||
if(m_silence)
|
||||
memset(pkt->pData, 0x0, pkt->iSize);
|
||||
|
||||
- m_omxAudio.AddPackets(pkt->pData, pkt->iSize, m_audioClock, m_audioClock);
|
||||
+ m_omxAudio.AddPackets(pkt->pData, pkt->iSize, m_audioClock, m_audioClock, 0);
|
||||
}
|
||||
|
||||
m_audioStats.AddSampleBytes(pkt->iSize);
|
||||
--
|
||||
1.8.5.5
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user