From 8eeb20a3147d8829f99d74fa5d9153eb9894b553 Mon Sep 17 00:00:00 2001 From: Yuusei KUWANA Date: Thu, 23 Oct 2014 21:46:16 +0900 Subject: [PATCH 1/4] more protocol. --- OMXReader.cpp | 1 + 1 file changed, 1 insertion(+) diff --git a/OMXReader.cpp b/OMXReader.cpp index f419ffbf..914f5a21 100644 --- a/OMXReader.cpp +++ b/OMXReader.cpp @@ -194,6 +194,7 @@ bool OMXReader::Open(std::string filename, bool dump_format, bool live /* =false if(m_filename.substr(0,6) == "mms://" || m_filename.substr(0,7) == "mmsh://" || m_filename.substr(0,7) == "mmst://" || m_filename.substr(0,7) == "mmsu://" || m_filename.substr(0,7) == "http://" || m_filename.substr(0,8) == "https://" || + m_filename.substr(0,6) == "tcp://" || m_filename.substr(0,7) == "rtmp://" || m_filename.substr(0,6) == "udp://" || m_filename.substr(0,7) == "rtsp://" || m_filename.substr(0,6) == "rtp://" || m_filename.substr(0,6) == "ftp://" || m_filename.substr(0,7) == "sftp://" || From f1c65417b56fcb5bd685788038c67ade8c734b24 Mon Sep 17 00:00:00 2001 From: Yuusei KUWANA Date: Fri, 24 Oct 2014 21:59:33 +0900 Subject: [PATCH 2/4] priority use of fdk-aac --- DllAvCodec.h | 4 ++++ OMXAudioCodecOMX.cpp | 11 ++++++++++- 2 files changed, 14 insertions(+), 1 deletion(-) diff --git a/DllAvCodec.h b/DllAvCodec.h index b0d4e1f8..3cb781f8 100644 --- a/DllAvCodec.h +++ b/DllAvCodec.h @@ -73,6 +73,7 @@ class DllAvCodecInterface virtual void avcodec_flush_buffers(AVCodecContext *avctx)=0; virtual int avcodec_open2_dont_call(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options)=0; virtual AVCodec *avcodec_find_decoder(enum AVCodecID id)=0; + virtual AVCodec *avcodec_find_decoder_by_name(const char *name)=0; virtual AVCodec *avcodec_find_encoder(enum AVCodecID id)=0; virtual int avcodec_close_dont_call(AVCodecContext *avctx)=0; virtual AVFrame *av_frame_alloc(void)=0; @@ -125,6 +126,7 @@ class DllAvCodec : public DllDynamic, DllAvCodecInterface virtual int avcodec_open2_dont_call(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options) { *(int *)0x0 = 0; return 0; } virtual int avcodec_close_dont_call(AVCodecContext *avctx) { *(int *)0x0 = 0; return 0; } virtual AVCodec *avcodec_find_decoder(enum AVCodecID id) { return ::avcodec_find_decoder(id); } + virtual AVCodec *avcodec_find_decoder_by_name(const char *name) { return ::avcodec_find_decoder_by_name(name); } virtual AVCodec *avcodec_find_encoder(enum AVCodecID id) { return ::avcodec_find_encoder(id); } virtual int avcodec_close(AVCodecContext *avctx) { @@ -195,6 +197,7 @@ class DllAvCodec : public DllDynamic, DllAvCodecInterface DEFINE_METHOD0(void, avcodec_register_all_dont_call) DEFINE_METHOD1(AVCodec*, avcodec_find_decoder, (enum AVCodecID p1)) + DEFINE_METHOD1(AVCodec*, avcodec_find_decoder_by_name, (const char *p1)) DEFINE_METHOD1(AVCodec*, avcodec_find_encoder, (enum AVCodecID p1)) DEFINE_METHOD1(int, avcodec_close_dont_call, (AVCodecContext *p1)) DEFINE_METHOD0(AVFrame*, av_frame_alloc) @@ -218,6 +221,7 @@ class DllAvCodec : public DllDynamic, DllAvCodecInterface RESOLVE_METHOD_RENAME(avcodec_open2,avcodec_open2_dont_call) RESOLVE_METHOD_RENAME(avcodec_close,avcodec_close_dont_call) RESOLVE_METHOD(avcodec_find_decoder) + RESOLVE_METHOD(avcodec_find_decoder_by_name) RESOLVE_METHOD(avcodec_find_encoder) RESOLVE_METHOD(av_frame_alloc) RESOLVE_METHOD_RENAME(avcodec_register_all, avcodec_register_all_dont_call) diff --git a/OMXAudioCodecOMX.cpp b/OMXAudioCodecOMX.cpp index 58d75a9b..2c56fd35 100644 --- a/OMXAudioCodecOMX.cpp +++ b/OMXAudioCodecOMX.cpp @@ -69,7 +69,16 @@ bool COMXAudioCodecOMX::Open(COMXStreamInfo &hints, enum PCMLayout layout) m_dllAvCodec.avcodec_register_all(); - pCodec = m_dllAvCodec.avcodec_find_decoder(hints.codec); + if( hints.codec != AV_CODEC_ID_AAC ){ + pCodec = m_dllAvCodec.avcodec_find_decoder(hints.codec); + } else { + pCodec = m_dllAvCodec.avcodec_find_decoder_by_name("libfdk_aac"); + if (!pCodec){ + pCodec = m_dllAvCodec.avcodec_find_decoder(hints.codec); + } + } + printf( "[Audio codec is %s]\n", pCodec->long_name ); + if (!pCodec) { CLog::Log(LOGDEBUG,"COMXAudioCodecOMX::Open() Unable to find codec %d", hints.codec); From 0267c95ff98a075c2d1d565026ba47f3d8678af1 Mon Sep 17 00:00:00 2001 From: Yuusei KUWANA Date: Fri, 24 Oct 2014 22:01:59 +0900 Subject: [PATCH 3/4] more smooth playback when audio channel change --- OMXAudio.cpp | 232 ++++++++++++++++++++++++++++++++++++++++++++- OMXAudio.h | 6 ++ OMXPlayerAudio.cpp | 30 +++++- 3 files changed, 262 insertions(+), 6 deletions(-) diff --git a/OMXAudio.cpp b/OMXAudio.cpp index e5d63653..bddf8fbf 100644 --- a/OMXAudio.cpp +++ b/OMXAudio.cpp @@ -87,8 +87,34 @@ bool COMXAudio::PortSettingsChanged() if (m_settings_changed) { + /* setup mixer input */ + OMX_INIT_STRUCTURE(m_pcm_output); + m_pcm_output.nPortIndex = m_omx_decoder.GetOutputPort(); + omx_err = m_omx_decoder.GetParameter(OMX_IndexParamAudioPcm, &m_pcm_output); + if(omx_err != OMX_ErrorNone) + { + CLog::Log(LOGERROR, "%s::%s - error m_omx_decoder GetParameter omx_err(0x%08x)", CLASSNAME, __func__, omx_err); + } + + //// decoder output -> mixer input + memcpy(m_pcm_output.eChannelMapping, m_input_channels, sizeof(m_input_channels)); + // round up to power of 2 + m_pcm_output.nChannels = m_InputChannels > 4 ? 8 : m_InputChannels > 2 ? 4 : m_InputChannels; + /* limit samplerate (through resampling) if requested */ + m_pcm_output.nSamplingRate = std::min(std::max((int)m_pcm_output.nSamplingRate, 8000), 192000); + m_omx_decoder.DisablePort(m_omx_decoder.GetOutputPort(), true); - m_omx_decoder.EnablePort(m_omx_decoder.GetOutputPort(), true); + m_omx_mixer.DisablePort(m_omx_mixer.GetInputPort(), false); + + m_pcm_output.nPortIndex = m_omx_mixer.GetInputPort(); + omx_err = m_omx_mixer.SetParameter(OMX_IndexParamAudioPcm, &m_pcm_output); + if(omx_err != OMX_ErrorNone) + { + CLog::Log(LOGERROR, "%s::%s - error m_omx_mixer(input) SetParameter omx_err(0x%08x)", CLASSNAME, __func__, omx_err); + } + m_omx_decoder.EnablePort(m_omx_decoder.GetOutputPort(), false); + m_omx_mixer.EnablePort(m_omx_mixer.GetInputPort(), false); + return true; } @@ -466,7 +492,8 @@ bool COMXAudio::Initialize(OMXClock *clock, const OMXAudioConfig &config, uint64 // should be big enough that common formats (e.g. 6 channel DTS) fit in a single packet. // we don't mind less common formats being split (e.g. ape/wma output large frames) // 6 channel 32bpp float to 8 channel 16bpp in, so a full 48K input buffer will fit the output buffer - m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (m_InputChannels * m_BitsPerSample) >> (rounded_up_channels_shift[m_InputChannels] + 4); +// m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (m_InputChannels * m_BitsPerSample) >> (rounded_up_channels_shift[m_InputChannels] + 4); + m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (6 * m_BitsPerSample) >> (rounded_up_channels_shift[6] + 4); m_wave_header.Samples.wSamplesPerBlock = 0; m_wave_header.Format.nChannels = m_InputChannels; @@ -657,6 +684,207 @@ bool COMXAudio::Initialize(OMXClock *clock, const OMXAudioConfig &config, uint64 return true; } +bool COMXAudio::ChangeInputFormat(const CStdString& device, int iChannels, uint64_t channelMap, + COMXStreamInfo &hints, enum PCMLayout layout, unsigned int uiSampleRate, unsigned int uiBitsPerSample, bool boostOnDownmix, + OMXClock *clock, bool bUsePassthrough, bool bUseHWDecode, bool is_live, float fifo_size) +{ + CSingleLock lock (m_critSection); + OMX_ERRORTYPE omx_err; + + layout = PCM_LAYOUT_2_0; + + m_InputChannels = iChannels; + + if(m_InputChannels == 0) + return false; + +// if(hints.samplerate == 0) +// return false; + + memset(m_input_channels, 0x0, sizeof(m_input_channels)); +// memset(m_output_channels, 0x0, sizeof(m_output_channels)); + memset(&m_wave_header, 0x0, sizeof(m_wave_header)); + + m_wave_header.Format.nChannels = 2; + m_wave_header.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; + + // set the input format, and get the channel layout so we know what we need to open + if (!m_Passthrough && channelMap) + { + enum PCMChannels inLayout[OMX_AUDIO_MAXCHANNELS]; + enum PCMChannels outLayout[OMX_AUDIO_MAXCHANNELS]; + // force out layout to stereo if input is not multichannel - it gives the receiver a chance to upmix + if (channelMap == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT) || channelMap == AV_CH_FRONT_CENTER) + layout = PCM_LAYOUT_2_0; + BuildChannelMap(inLayout, channelMap); + m_OutputChannels = BuildChannelMapCEA(outLayout, GetChannelLayout(layout)); + CPCMRemap m_remap; + m_remap.Reset(); + /*outLayout = */m_remap.SetInputFormat (m_InputChannels, inLayout, uiBitsPerSample / 8, uiSampleRate, layout, !m_normalize_downmix); + m_remap.SetOutputFormat(m_OutputChannels, outLayout); + m_remap.GetDownmixMatrix(m_downmix_matrix); + m_wave_header.dwChannelMask = channelMap; + BuildChannelMapOMX(m_input_channels, channelMap); + BuildChannelMapOMX(m_output_channels, GetChannelLayout(layout)); + } + + m_SampleRate = uiSampleRate; + m_BitsPerSample = uiBitsPerSample; + + m_BytesPerSec = m_SampleRate * 2 << rounded_up_channels_shift[m_InputChannels]; + m_BufferLen = m_BytesPerSec * AUDIO_BUFFER_SECONDS; + m_InputBytesPerSec = m_SampleRate * m_BitsPerSample * m_InputChannels >> 3; + + // should be big enough that common formats (e.g. 6 channel DTS) fit in a single packet. + // we don't mind less common formats being split (e.g. ape/wma output large frames) + // 6 channel 32bpp float to 8 channel 16bpp in, so a full 48K input buffer will fit the output buffer +// m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (m_InputChannels * m_BitsPerSample) >> (rounded_up_channels_shift[m_InputChannels] + 4); + + m_wave_header.Samples.wSamplesPerBlock = 0; + m_wave_header.Format.nChannels = m_InputChannels; + m_wave_header.Format.nBlockAlign = m_InputChannels * (m_BitsPerSample >> 3); + // 0x8000 is custom format interpreted by GPU as WAVE_FORMAT_IEEE_FLOAT_PLANAR + m_wave_header.Format.wFormatTag = m_BitsPerSample == 32 ? 0x8000 : WAVE_FORMAT_PCM; + m_wave_header.Format.nSamplesPerSec = m_SampleRate; + m_wave_header.Format.nAvgBytesPerSec = m_BytesPerSec; + m_wave_header.Format.wBitsPerSample = m_BitsPerSample; + m_wave_header.Samples.wValidBitsPerSample = m_BitsPerSample; + m_wave_header.Format.cbSize = 0; + m_wave_header.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; + + OMX_INIT_STRUCTURE(m_pcm_input); + memcpy(m_pcm_input.eChannelMapping, m_input_channels, sizeof(m_input_channels)); + m_pcm_input.eNumData = OMX_NumericalDataSigned; + m_pcm_input.eEndian = OMX_EndianLittle; + m_pcm_input.bInterleaved = OMX_TRUE; + m_pcm_input.nBitPerSample = m_BitsPerSample; + m_pcm_input.ePCMMode = OMX_AUDIO_PCMModeLinear; + m_pcm_input.nChannels = m_InputChannels; + m_pcm_input.nSamplingRate = m_SampleRate; + + // set up the number/size of buffers for decoder input + OMX_PARAM_PORTDEFINITIONTYPE port_param; + OMX_INIT_STRUCTURE(port_param); + port_param.nPortIndex = m_omx_decoder.GetInputPort(); + + omx_err = m_omx_decoder.GetParameter(OMX_IndexParamPortDefinition, &port_param); + if(omx_err != OMX_ErrorNone) + { + CLog::Log(LOGERROR, "COMXAudio::ChangeInputFormat error get OMX_IndexParamPortDefinition (input) omx_err(0x%08x)\n", omx_err); + return false; + } + + port_param.format.audio.eEncoding = m_eEncoding; + port_param.nBufferSize = m_ChunkLen; + port_param.nBufferCountActual = std::max(port_param.nBufferCountMin, 16U); + + // set up the number/size of buffers for decoder output + OMX_INIT_STRUCTURE(port_param); + port_param.nPortIndex = m_omx_decoder.GetOutputPort(); + + omx_err = m_omx_decoder.GetParameter(OMX_IndexParamPortDefinition, &port_param); + if(omx_err != OMX_ErrorNone) + { + CLog::Log(LOGERROR, "COMXAudio::ChangeInputFormat error get OMX_IndexParamPortDefinition (output) omx_err(0x%08x)\n", omx_err); + return false; + } + + port_param.nBufferCountActual = std::max((unsigned int)port_param.nBufferCountMin, m_BufferLen / port_param.nBufferSize); + + if(m_eEncoding == OMX_AUDIO_CodingPCM) + { + /* setup decoder input */ + OMX_INIT_STRUCTURE(m_pcm_output); + m_pcm_output.nPortIndex = m_omx_decoder.GetOutputPort(); + omx_err = m_omx_decoder.GetParameter(OMX_IndexParamAudioPcm, &m_pcm_output); + if(omx_err != OMX_ErrorNone) + { + CLog::Log(LOGERROR, "%s::%s - error m_omx_decoder GetParameter omx_err(0x%08x)", CLASSNAME, __func__, omx_err); + } + + printf( "m_omx_decoder OMX_IndexParamAudioPcm nChannels %d\n", m_pcm_output.nChannels ); + + //// decoder output -> mixer input + memcpy(m_pcm_output.eChannelMapping, m_input_channels, sizeof(m_input_channels)); + // round up to power of 2 + m_pcm_output.nChannels = m_InputChannels > 4 ? 8 : m_InputChannels > 2 ? 4 : m_InputChannels; + /* limit samplerate (through resampling) if requested */ + m_pcm_output.nSamplingRate = std::min(std::max((int)m_pcm_output.nSamplingRate, 8000), 192000); + + printf( "m_omx_decoder OMX_IndexParamAudioPcm nChannels %d\n", m_pcm_output.nChannels ); + + m_omx_decoder.DisablePort(m_omx_decoder.GetInputPort(), false); + m_omx_decoder.EnablePort(m_omx_decoder.GetInputPort(), true); + + OMX_BUFFERHEADERTYPE *omx_buffer = m_omx_decoder.GetInputBuffer(); + if(omx_buffer == NULL) + { + CLog::Log(LOGERROR, "COMXAudio::ChangeInputFormat - buffer error 0x%08x", omx_err); + return false; + } + + omx_buffer->nOffset = 0; + omx_buffer->nFilledLen = sizeof(m_wave_header); + if(omx_buffer->nFilledLen > omx_buffer->nAllocLen) + { + CLog::Log(LOGERROR, "COMXAudio::ChangeInputFormat - omx_buffer->nFilledLen > omx_buffer->nAllocLen"); + return false; + } + memset((unsigned char *)omx_buffer->pBuffer, 0x0, omx_buffer->nAllocLen); + memcpy((unsigned char *)omx_buffer->pBuffer, &m_wave_header, omx_buffer->nFilledLen); + omx_buffer->nFlags = OMX_BUFFERFLAG_CODECCONFIG | OMX_BUFFERFLAG_ENDOFFRAME; + + omx_err = m_omx_decoder.EmptyThisBuffer(omx_buffer); + if (omx_err != OMX_ErrorNone) + { + CLog::Log(LOGERROR, "%s::%s - OMX_EmptyThisBuffer() failed with result(0x%x)\n", CLASSNAME, __func__, omx_err); + return false; + } + } + else if(m_HWDecode) + { + // send decoder config + if(m_extrasize > 0 && m_extradata != NULL) + { + OMX_BUFFERHEADERTYPE *omx_buffer = m_omx_decoder.GetInputBuffer(); + + if(omx_buffer == NULL) + { + CLog::Log(LOGERROR, "%s::%s - buffer error 0x%08x", CLASSNAME, __func__, omx_err); + return false; + } + + omx_buffer->nOffset = 0; + omx_buffer->nFilledLen = m_extrasize; + if(omx_buffer->nFilledLen > omx_buffer->nAllocLen) + { + CLog::Log(LOGERROR, "%s::%s - omx_buffer->nFilledLen > omx_buffer->nAllocLen", CLASSNAME, __func__); + return false; + } + + memset((unsigned char *)omx_buffer->pBuffer, 0x0, omx_buffer->nAllocLen); + memcpy((unsigned char *)omx_buffer->pBuffer, m_extradata, omx_buffer->nFilledLen); + omx_buffer->nFlags = OMX_BUFFERFLAG_CODECCONFIG | OMX_BUFFERFLAG_ENDOFFRAME; + + omx_err = m_omx_decoder.EmptyThisBuffer(omx_buffer); + if (omx_err != OMX_ErrorNone) + { + CLog::Log(LOGERROR, "%s::%s - OMX_EmptyThisBuffer() failed with result(0x%x)\n", CLASSNAME, __func__, omx_err); + return false; + } + } + } + + CLog::Log(LOGDEBUG, "COMXAudio::ChangeInputFormat Input bps %d samplerate %d channels %d buffer size %d bytes per second %d", + (int)m_pcm_input.nBitPerSample, (int)m_pcm_input.nSamplingRate, (int)m_pcm_input.nChannels, m_BufferLen, m_InputBytesPerSec); + PrintPCM(&m_pcm_input, std::string("input")); + CLog::Log(LOGDEBUG, "COMXAudio::ChangeInputFormat device %s passthrough %d hwdecode %d", + device.c_str(), m_Passthrough, m_HWDecode); + + return true; +} + + //*********************************************************************************************** bool COMXAudio::Deinitialize() { diff --git a/OMXAudio.h b/OMXAudio.h index 5b56e97c..9a69fbb6 100644 --- a/OMXAudio.h +++ b/OMXAudio.h @@ -87,6 +87,10 @@ class COMXAudio float GetMaxLevel(double &pts); COMXAudio(); bool Initialize(OMXClock *clock, const OMXAudioConfig &config, uint64_t channelMap, unsigned int uiBitsPerSample); + bool ChangeInputFormat(const CStdString& device, int iChannels, uint64_t channelMap, + COMXStreamInfo &hints, enum PCMLayout layout, unsigned int uiSamplesPerSec, unsigned int uiBitsPerSample, bool boostOnDownmix, + OMXClock *clock, bool bUsePassthrough = false, bool bUseHWDecode = false, bool is_live = false, float fifo_size = 0); + ~COMXAudio(); bool PortSettingsChanged(); @@ -102,6 +106,7 @@ class COMXAudio bool ApplyVolume(); void SubmitEOS(); bool IsEOS(); + bool ToggleMonoTrack(); void Flush(); @@ -146,6 +151,7 @@ class COMXAudio bool m_submitted_eos; bool m_failed_eos; OMXAudioConfig m_config; + int m_monotrack; OMX_AUDIO_CHANNELTYPE m_input_channels[OMX_AUDIO_MAXCHANNELS]; OMX_AUDIO_CHANNELTYPE m_output_channels[OMX_AUDIO_MAXCHANNELS]; diff --git a/OMXPlayerAudio.cpp b/OMXPlayerAudio.cpp index 703c23cf..3bc89959 100644 --- a/OMXPlayerAudio.cpp +++ b/OMXPlayerAudio.cpp @@ -188,9 +188,8 @@ bool OMXPlayerAudio::Decode(OMXPacket *pkt) } // for passthrough we only care about the codec and the samplerate - bool minor_change = channels != m_config.hints.channels || - pkt->hints.bitspersample != m_config.hints.bitspersample || - old_bitrate != new_bitrate; + bool minor_change = ( pkt->hints.bitspersample != m_hints.bitspersample ) || + ( old_bitrate != new_bitrate ); if(pkt->hints.codec != m_config.hints.codec || pkt->hints.samplerate != m_config.hints.samplerate || @@ -230,12 +229,30 @@ bool OMXPlayerAudio::Decode(OMXPacket *pkt) while(data_len > 0) { int len = m_pAudioCodec->Decode((BYTE *)data_dec, data_len, dts, pts); - if( (len < 0) || (len > data_len) ) + if( (len < 0) /*|| (len > data_len)*/ ) { m_pAudioCodec->Reset(); break; } + // check audio channel change + if( m_pAudioCodec->GetChannels() != 0 && m_pAudioCodec->GetChannels() < 7 && + m_hints.channels != m_pAudioCodec->GetChannels() ) + { + + m_hints = pkt->hints; + m_hints.channels = m_pAudioCodec->GetChannels(); + channels = m_pAudioCodec->GetChannels(); + + bool bAudioRenderOpen = false; + bAudioRenderOpen = m_decoder->ChangeInputFormat(m_device, m_hints.channels, m_pAudioCodec->GetChannelMap(), + m_hints, m_layout, m_hints.samplerate, m_pAudioCodec->GetBitsPerSample(), m_boost_on_downmix, + m_av_clock, m_passthrough, m_hw_decode, m_live, m_fifo_size); + if(!bAudioRenderOpen) + CLog::Log(LOGINFO, "m_decoder->ChangeInputFormat() failed"); + + } + data_dec+= len; data_len -= len; @@ -501,3 +518,8 @@ bool OMXPlayerAudio::IsEOS() return m_packets.empty() && (!m_decoder || m_decoder->IsEOS()); } +bool OMXPlayerAudio::ToggleMonoTrack() +{ + if(m_decoder) + m_decoder->ToggleMonoTrack(); +} From be9e90b7070e1895c39e715760c141ef4a777e1e Mon Sep 17 00:00:00 2001 From: Yuusei KUWANA Date: Wed, 29 Oct 2014 21:58:53 +0900 Subject: [PATCH 4/4] revert buffer allocation --- OMXAudio.cpp | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/OMXAudio.cpp b/OMXAudio.cpp index bddf8fbf..b93cee75 100644 --- a/OMXAudio.cpp +++ b/OMXAudio.cpp @@ -492,8 +492,7 @@ bool COMXAudio::Initialize(OMXClock *clock, const OMXAudioConfig &config, uint64 // should be big enough that common formats (e.g. 6 channel DTS) fit in a single packet. // we don't mind less common formats being split (e.g. ape/wma output large frames) // 6 channel 32bpp float to 8 channel 16bpp in, so a full 48K input buffer will fit the output buffer -// m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (m_InputChannels * m_BitsPerSample) >> (rounded_up_channels_shift[m_InputChannels] + 4); - m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (6 * m_BitsPerSample) >> (rounded_up_channels_shift[6] + 4); + m_ChunkLen = AUDIO_DECODE_OUTPUT_BUFFER * (m_InputChannels * m_BitsPerSample) >> (rounded_up_channels_shift[m_InputChannels] + 4); m_wave_header.Samples.wSamplesPerBlock = 0; m_wave_header.Format.nChannels = m_InputChannels; @@ -691,8 +690,6 @@ bool COMXAudio::ChangeInputFormat(const CStdString& device, int iChannels, uint6 CSingleLock lock (m_critSection); OMX_ERRORTYPE omx_err; - layout = PCM_LAYOUT_2_0; - m_InputChannels = iChannels; if(m_InputChannels == 0)