From ab15e340321aa6cfd2fe444dbcaae8f7d3ee78ff Mon Sep 17 00:00:00 2001 From: Armin Novak Date: Fri, 9 Nov 2018 10:15:14 +0100 Subject: [PATCH] Refactored freerdp_dsp_resample, checks now done internally. --- libfreerdp/codec/dsp.c | 68 +++++++++++++++++++++++++++++++----------- 1 file changed, 50 insertions(+), 18 deletions(-) diff --git a/libfreerdp/codec/dsp.c b/libfreerdp/codec/dsp.c index ea1ea5a7d..d5cbb6c87 100644 --- a/libfreerdp/codec/dsp.c +++ b/libfreerdp/codec/dsp.c @@ -29,6 +29,7 @@ #include #include +#include #include #if !defined(WITH_DSP_FFMPEG) @@ -109,29 +110,65 @@ struct _FREERDP_DSP_CONTEXT static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context, const BYTE* src, size_t size, - const AUDIO_FORMAT* srcFormat) + const AUDIO_FORMAT* srcFormat, + const BYTE** data, size_t* length) { BYTE* p; size_t sframes, rframes; size_t rsize; size_t i, j; size_t sbytes, rbytes; - size_t srcBytePerFrame; + size_t srcBytesPerFrame, dstBytesPerFrame; + size_t srcChannels, dstChannels; + AUDIO_FORMAT format; - if (!context || !src || !srcFormat) + if (!context || !src || !srcFormat || !data || !length) return FALSE; - assert(srcFormat->wFormatTag == WAVE_FORMAT_PCM); - srcBytePerFrame = (srcFormat->wBitsPerSample > 8) ? 2 : 1; - sbytes = srcFormat->nChannels * srcBytePerFrame; + if (srcFormat->wFormatTag != WAVE_FORMAT_PCM) + { + WLog_ERR(TAG, "%s requires %s for sample input, got %s", __FUNCTION__, + audio_format_get_tag_string(WAVE_FORMAT_PCM), + audio_format_get_tag_string(srcFormat->wFormatTag)); + return FALSE; + } + + srcChannels = srcFormat->nChannels; + dstChannels = context->format.nChannels; + srcBytesPerFrame = (srcFormat->wBitsPerSample > 8) ? 2 : 1; + dstBytesPerFrame = (context->format.wBitsPerSample > 8) ? 2 : 1; + /* We want to ignore differences of source and destination format. */ + format = *srcFormat; + format.wFormatTag = WAVE_FORMAT_UNKNOWN; + + if (audio_format_compatible(&format, &context->format)) + return TRUE; + + /* TODO: Currently sample size conversion is not implemented. */ + if (srcFormat->wBitsPerSample != context->format.wBitsPerSample) + { + WLog_ERR(TAG, "%s does not support changing sample bit width [in=%"PRId16", out=%"PRId16"]", + __FUNCTION__, srcFormat->wBitsPerSample, context->format.wBitsPerSample); + return FALSE; + } + + /* TODO: Currently sample size conversion is not implemented. */ + if (srcChannels != dstChannels) + { + WLog_ERR(TAG, "%s does not support changing number of channels [in=%"PRIdz", out=%"PRIdz"]", + __FUNCTION__, srcChannels, dstChannels); + return FALSE; + } + + sbytes = srcChannels * srcBytesPerFrame; sframes = size / sbytes; - rbytes = srcBytePerFrame * context->format.nChannels; + rbytes = dstBytesPerFrame * dstChannels; /* Integer rounding correct division */ rframes = (sframes * context->format.nSamplesPerSec + (srcFormat->nSamplesPerSec + 1) / 2) / srcFormat->nSamplesPerSec; - rsize = rbytes * rframes; + rsize = rframes * rbytes; - if (!Stream_EnsureCapacity(context->resample, rsize + 1024)) + if (!Stream_EnsureCapacity(context->resample, rsize)) return FALSE; p = Stream_Buffer(context->resample); @@ -159,6 +196,8 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context, Stream_SetPointer(context->resample, p); Stream_SealLength(context->resample); + *data = Stream_Buffer(context->resample); + *length = Stream_Length(context->resample); return TRUE; } @@ -1049,14 +1088,8 @@ BOOL freerdp_dsp_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* srcFor if (!context || !context->encoder || !srcFormat || !data || !out) return FALSE; - if (srcFormat->nSamplesPerSec != context->format.nSamplesPerSec) - { - if (!freerdp_dsp_resample(context, data, length, srcFormat)) - return FALSE; - - data = Stream_Buffer(context->resample); - length = Stream_GetPosition(context->resample); - } + if (!freerdp_dsp_resample(context, data, length, srcFormat, &data, &length)) + return FALSE; switch (context->format.wFormatTag) { @@ -1196,7 +1229,6 @@ BOOL freerdp_dsp_supports_format(const AUDIO_FORMAT* format, BOOL encode) #endif } - BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* targetFormat) { #if defined(WITH_DSP_FFMPEG)