| Message ID | 7580B1EB-3AD3-4D90-AEDD-43F98F9BBC53@rastageeks.org |
|---|---|
| State | New |
| Headers | show |
| Series | [FFmpeg-devel,v6,01/03] libavdevice/avfoundation.m: use AudioConvert, extend supported formats | expand |
| Context | Check | Description |
|---|---|---|
| andriy/configurex86 | warning | Failed to apply patch |
| andriy/configureppc | warning | Failed to apply patch |
On Tue, Dec 28, 2021 at 2:50 PM Romain Beauxis <toots@rastageeks.org> wrote: > This is the first patch of a series of 3 that fix, cleanup and enhance the > avfoundation implementation for libavdevice. > > The patches have been submitted a couple of times now and have > received very nice feedback for the last two however but they do not seem > to have been considered for inclusion thus far. > > These patches come from an actual user-facing application relying on > libavdevice’s implementation of avfoundation audio input. Without them, > Avfoundation is practically unusable as it will: > * Refuse to process certain specific audio input format that are actually > returned by the OS for some users (packed PCM audio) > * Drop audio frames, resulting in corrupted audio input. This might have > been > unnoticed with video frames but this makes avfoundation essentially > unusable > for audio. > > The patches are now being included in our production build so they are > tested > and usable in production. > > So, this bares the question: is avfoundation still supported and actively > maintained > in libavdevice? It feels that such important bugs should have been noticed > by now > and also generated a little more interest in fixing them. Thanks for working on this, and addressing all the feedback so far. The patchset LGTM, and I think it should be applied. Looks like MAINTAINERS lists Thilo for avfoundation.m. I'm not sure if he's seen this yet, so I'm cc'ing on this reply. If we don't hear in the next couple weeks, I can apply these changes. Aman > > Thanks for y’all feedback! > — Romain > ----- > > Changes: > * v2: None > * v3: None > * v4: None > * v5: Fix indentation/wrapping > * v6: None > > * Implement support for AudioConverter > * Switch to AudioConverter's API to convert unsupported PCM > formats (non-interleaved, non-packed) to supported formats > * Minimize data copy. > > This fixes: https://trac.ffmpeg.org/ticket/9502 > > API ref: > > https://developer.apple.com/documentation/audiotoolbox/audio_converter_services > > Signed-off-by: Romain Beauxis <toots@rastageeks.org> > --- > libavdevice/avfoundation.m | 250 +++++++++++++++++++++---------------- > 1 file changed, 144 insertions(+), 106 deletions(-) > > diff --git a/libavdevice/avfoundation.m b/libavdevice/avfoundation.m > index 0cd6e646d5..79c9207cfa 100644 > --- a/libavdevice/avfoundation.m > +++ b/libavdevice/avfoundation.m > @@ -111,16 +111,10 @@ > > int num_video_devices; > > - int audio_channels; > - int audio_bits_per_sample; > - int audio_float; > - int audio_be; > - int audio_signed_integer; > - int audio_packed; > - int audio_non_interleaved; > - > - int32_t *audio_buffer; > - int audio_buffer_size; > + UInt32 audio_buffers; > + UInt32 audio_channels; > + UInt32 bytes_per_sample; > + AudioConverterRef audio_converter; > > enum AVPixelFormat pixel_format; > > @@ -299,7 +293,10 @@ static void destroy_context(AVFContext* ctx) > ctx->avf_delegate = NULL; > ctx->avf_audio_delegate = NULL; > > - av_freep(&ctx->audio_buffer); > + if (ctx->audio_converter) { > + AudioConverterDispose(ctx->audio_converter); > + ctx->audio_converter = NULL; > + } > > pthread_mutex_destroy(&ctx->frame_lock); > > @@ -673,6 +670,10 @@ static int get_audio_config(AVFormatContext *s) > AVFContext *ctx = (AVFContext*)s->priv_data; > CMFormatDescriptionRef format_desc; > AVStream* stream = avformat_new_stream(s, NULL); > + AudioStreamBasicDescription output_format = {0}; > + int audio_bits_per_sample, audio_float, audio_be; > + int audio_signed_integer, audio_packed, audio_non_interleaved; > + int must_convert = 0; > > if (!stream) { > return 1; > @@ -690,60 +691,95 @@ static int get_audio_config(AVFormatContext *s) > avpriv_set_pts_info(stream, 64, 1, avf_time_base); > > format_desc = > CMSampleBufferGetFormatDescription(ctx->current_audio_frame); > - const AudioStreamBasicDescription *basic_desc = > CMAudioFormatDescriptionGetStreamBasicDescription(format_desc); > + const AudioStreamBasicDescription *input_format = > CMAudioFormatDescriptionGetStreamBasicDescription(format_desc); > > - if (!basic_desc) { > + if (!input_format) { > unlock_frames(ctx); > av_log(s, AV_LOG_ERROR, "audio format not available\n"); > return 1; > } > > + if (input_format->mFormatID != kAudioFormatLinearPCM) { > + unlock_frames(ctx); > + av_log(s, AV_LOG_ERROR, "only PCM audio format are supported at > the moment\n"); > + return 1; > + } > + > stream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; > - stream->codecpar->sample_rate = basic_desc->mSampleRate; > - stream->codecpar->channels = basic_desc->mChannelsPerFrame; > + stream->codecpar->sample_rate = input_format->mSampleRate; > + stream->codecpar->channels = input_format->mChannelsPerFrame; > stream->codecpar->channel_layout = > av_get_default_channel_layout(stream->codecpar->channels); > > - ctx->audio_channels = basic_desc->mChannelsPerFrame; > - ctx->audio_bits_per_sample = basic_desc->mBitsPerChannel; > - ctx->audio_float = basic_desc->mFormatFlags & > kAudioFormatFlagIsFloat; > - ctx->audio_be = basic_desc->mFormatFlags & > kAudioFormatFlagIsBigEndian; > - ctx->audio_signed_integer = basic_desc->mFormatFlags & > kAudioFormatFlagIsSignedInteger; > - ctx->audio_packed = basic_desc->mFormatFlags & > kAudioFormatFlagIsPacked; > - ctx->audio_non_interleaved = basic_desc->mFormatFlags & > kAudioFormatFlagIsNonInterleaved; > - > - if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_float && > - ctx->audio_bits_per_sample == 32 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? > AV_CODEC_ID_PCM_F32BE : AV_CODEC_ID_PCM_F32LE; > - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_signed_integer && > - ctx->audio_bits_per_sample == 16 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? > AV_CODEC_ID_PCM_S16BE : AV_CODEC_ID_PCM_S16LE; > - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_signed_integer && > - ctx->audio_bits_per_sample == 24 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? > AV_CODEC_ID_PCM_S24BE : AV_CODEC_ID_PCM_S24LE; > - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_signed_integer && > - ctx->audio_bits_per_sample == 32 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? > AV_CODEC_ID_PCM_S32BE : AV_CODEC_ID_PCM_S32LE; > + audio_bits_per_sample = input_format->mBitsPerChannel; > + audio_float = input_format->mFormatFlags & > kAudioFormatFlagIsFloat; > + audio_be = input_format->mFormatFlags & > kAudioFormatFlagIsBigEndian; > + audio_signed_integer = input_format->mFormatFlags & > kAudioFormatFlagIsSignedInteger; > + audio_packed = input_format->mFormatFlags & > kAudioFormatFlagIsPacked; > + audio_non_interleaved = input_format->mFormatFlags & > kAudioFormatFlagIsNonInterleaved; > + > + ctx->bytes_per_sample = input_format->mBitsPerChannel >> 3; > + ctx->audio_channels = input_format->mChannelsPerFrame; > + > + if (audio_non_interleaved) { > + ctx->audio_buffers = input_format->mChannelsPerFrame; > } else { > - unlock_frames(ctx); > - av_log(s, AV_LOG_ERROR, "audio format is not supported\n"); > - return 1; > + ctx->audio_buffers = 1; > + } > + > + if (audio_non_interleaved || !audio_packed) { > + must_convert = 1; > + } > + > + output_format.mBitsPerChannel = input_format->mBitsPerChannel; > + output_format.mChannelsPerFrame = ctx->audio_channels; > + output_format.mFramesPerPacket = 1; > + output_format.mBytesPerFrame = output_format.mChannelsPerFrame * > ctx->bytes_per_sample; > + output_format.mBytesPerPacket = output_format.mFramesPerPacket * > output_format.mBytesPerFrame; > + output_format.mFormatFlags = kAudioFormatFlagIsPacked | audio_be; > + output_format.mFormatID = kAudioFormatLinearPCM; > + output_format.mReserved = 0; > + output_format.mSampleRate = input_format->mSampleRate; > + > + if (audio_float && > + audio_bits_per_sample == 32) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_F32BE : > AV_CODEC_ID_PCM_F32LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsFloat; > + } else if (audio_float && > + audio_bits_per_sample == 64) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_F64BE : > AV_CODEC_ID_PCM_F64LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsFloat; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 8) { > + stream->codecpar->codec_id = AV_CODEC_ID_PCM_S8; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 16) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S16BE : > AV_CODEC_ID_PCM_S16LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 24) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S24BE : > AV_CODEC_ID_PCM_S24LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 32) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S32BE : > AV_CODEC_ID_PCM_S32LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 64) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S64BE : > AV_CODEC_ID_PCM_S64LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S32BE : > AV_CODEC_ID_PCM_S32LE; > + output_format.mBitsPerChannel = 32; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + must_convert = 1; > } > > - if (ctx->audio_non_interleaved) { > - CMBlockBufferRef block_buffer = > CMSampleBufferGetDataBuffer(ctx->current_audio_frame); > - ctx->audio_buffer_size = > CMBlockBufferGetDataLength(block_buffer); > - ctx->audio_buffer = av_malloc(ctx->audio_buffer_size); > - if (!ctx->audio_buffer) { > + if (must_convert) { > + OSStatus ret = AudioConverterNew(input_format, &output_format, > &ctx->audio_converter); > + if (ret != noErr) { > unlock_frames(ctx); > - av_log(s, AV_LOG_ERROR, "error allocating audio buffer\n"); > + av_log(s, AV_LOG_ERROR, "Error while allocating audio > converter\n"); > return 1; > } > } > @@ -1048,6 +1084,7 @@ static int copy_cvpixelbuffer(AVFormatContext *s, > > static int avf_read_packet(AVFormatContext *s, AVPacket *pkt) > { > + OSStatus ret; > AVFContext* ctx = (AVFContext*)s->priv_data; > > do { > @@ -1091,7 +1128,7 @@ static int avf_read_packet(AVFormatContext *s, > AVPacket *pkt) > status = copy_cvpixelbuffer(s, image_buffer, pkt); > } else { > status = 0; > - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, > 0, pkt->size, pkt->data); > + ret = CMBlockBufferCopyDataBytes(block_buffer, 0, > pkt->size, pkt->data); > if (ret != kCMBlockBufferNoErr) { > status = AVERROR(EIO); > } > @@ -1105,82 +1142,83 @@ static int avf_read_packet(AVFormatContext *s, > AVPacket *pkt) > } > } else if (ctx->current_audio_frame != nil) { > CMBlockBufferRef block_buffer = > CMSampleBufferGetDataBuffer(ctx->current_audio_frame); > - int block_buffer_size = > CMBlockBufferGetDataLength(block_buffer); > > - if (!block_buffer || !block_buffer_size) { > - unlock_frames(ctx); > - return AVERROR(EIO); > - } > + size_t input_size = CMBlockBufferGetDataLength(block_buffer); > + int buffer_size = input_size / ctx->audio_buffers; > + int nb_samples = input_size / (ctx->audio_channels * > ctx->bytes_per_sample); > + int output_size = buffer_size; > > - if (ctx->audio_non_interleaved && block_buffer_size > > ctx->audio_buffer_size) { > + UInt32 size = sizeof(output_size); > + ret = AudioConverterGetProperty(ctx->audio_converter, > kAudioConverterPropertyCalculateOutputBufferSize, &size, &output_size); > + if (ret != noErr) { > unlock_frames(ctx); > - return AVERROR_BUFFER_TOO_SMALL; > + return AVERROR(EIO); > } > > - if (av_new_packet(pkt, block_buffer_size) < 0) { > + if (av_new_packet(pkt, output_size) < 0) { > unlock_frames(ctx); > return AVERROR(EIO); > } > > - CMItemCount count; > - CMSampleTimingInfo timing_info; > + if (ctx->audio_converter) { > + size_t input_buffer_size = offsetof(AudioBufferList, > mBuffers[0]) + (sizeof(AudioBuffer) * ctx->audio_buffers); > + AudioBufferList *input_buffer = > av_malloc(input_buffer_size); > > - if > (CMSampleBufferGetOutputSampleTimingInfoArray(ctx->current_audio_frame, 1, > &timing_info, &count) == noErr) { > - AVRational timebase_q = av_make_q(1, > timing_info.presentationTimeStamp.timescale); > - pkt->pts = pkt->dts = > av_rescale_q(timing_info.presentationTimeStamp.value, timebase_q, > avf_time_base_q); > - } > + input_buffer->mNumberBuffers = ctx->audio_buffers; > > - pkt->stream_index = ctx->audio_stream_index; > - pkt->flags |= AV_PKT_FLAG_KEY; > + for (int c = 0; c < ctx->audio_buffers; c++) { > + input_buffer->mBuffers[c].mNumberChannels = 1; > > - if (ctx->audio_non_interleaved) { > - int sample, c, shift, num_samples; > + ret = CMBlockBufferGetDataPointer(block_buffer, c * > buffer_size, (size_t *)&input_buffer->mBuffers[c].mDataByteSize, NULL, > (void *)&input_buffer->mBuffers[c].mData); > > - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, > 0, pkt->size, ctx->audio_buffer); > - if (ret != kCMBlockBufferNoErr) { > - unlock_frames(ctx); > - return AVERROR(EIO); > + if (ret != kCMBlockBufferNoErr) { > + av_free(input_buffer); > + unlock_frames(ctx); > + return AVERROR(EIO); > + } > } > > - num_samples = pkt->size / (ctx->audio_channels * > (ctx->audio_bits_per_sample >> 3)); > - > - // transform decoded frame into output format > - #define INTERLEAVE_OUTPUT(bps) > \ > - { > \ > - int##bps##_t **src; > \ > - int##bps##_t *dest; > \ > - src = av_malloc(ctx->audio_channels * > sizeof(int##bps##_t*)); \ > - if (!src) { > \ > - unlock_frames(ctx); > \ > - return AVERROR(EIO); > \ > - } > \ > - > \ > - for (c = 0; c < ctx->audio_channels; c++) { > \ > - src[c] = ((int##bps##_t*)ctx->audio_buffer) + c * > num_samples; \ > - } > \ > - dest = (int##bps##_t*)pkt->data; > \ > - shift = bps - ctx->audio_bits_per_sample; > \ > - for (sample = 0; sample < num_samples; sample++) > \ > - for (c = 0; c < ctx->audio_channels; c++) > \ > - *dest++ = src[c][sample] << shift; > \ > - av_freep(&src); > \ > - } > + AudioBufferList output_buffer = { > + .mNumberBuffers = 1, > + .mBuffers[0] = { > + .mNumberChannels = ctx->audio_channels, > + .mDataByteSize = pkt->size, > + .mData = pkt->data > + } > + }; > > - if (ctx->audio_bits_per_sample <= 16) { > - INTERLEAVE_OUTPUT(16) > - } else { > - INTERLEAVE_OUTPUT(32) > - } > - } else { > - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, > 0, pkt->size, pkt->data); > - if (ret != kCMBlockBufferNoErr) { > + ret = > AudioConverterConvertComplexBuffer(ctx->audio_converter, nb_samples, > input_buffer, &output_buffer); > + av_free(input_buffer); > + > + if (ret != noErr) { > unlock_frames(ctx); > return AVERROR(EIO); > } > + > + pkt->size = output_buffer.mBuffers[0].mDataByteSize; > + } else { > + ret = CMBlockBufferCopyDataBytes(block_buffer, 0, > pkt->size, pkt->data); > + if (ret != kCMBlockBufferNoErr) { > + unlock_frames(ctx); > + return AVERROR(EIO); > + } > } > > + CMItemCount count; > + CMSampleTimingInfo timing_info; > + > + if > (CMSampleBufferGetOutputSampleTimingInfoArray(ctx->current_audio_frame, 1, > &timing_info, &count) == noErr) { > + AVRational timebase_q = av_make_q(1, > timing_info.presentationTimeStamp.timescale); > + pkt->pts = pkt->dts = > av_rescale_q(timing_info.presentationTimeStamp.value, timebase_q, > avf_time_base_q); > + } > + > + pkt->stream_index = ctx->audio_stream_index; > + pkt->flags |= AV_PKT_FLAG_KEY; > + > CFRelease(ctx->current_audio_frame); > ctx->current_audio_frame = nil; > + > + unlock_frames(ctx); > } else { > pkt->data = NULL; > unlock_frames(ctx); > -- > 2.32.0 (Apple Git-132) > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". >
> On Dec 28, 2021, at 6:54 PM, Aman Karmani <ffmpeg@tmm1.net> wrote: > > > > On Tue, Dec 28, 2021 at 2:50 PM Romain Beauxis <toots@rastageeks.org> wrote: > This is the first patch of a series of 3 that fix, cleanup and enhance the > avfoundation implementation for libavdevice. > > The patches have been submitted a couple of times now and have > received very nice feedback for the last two however but they do not seem > to have been considered for inclusion thus far. > > These patches come from an actual user-facing application relying on > libavdevice’s implementation of avfoundation audio input. Without them, > Avfoundation is practically unusable as it will: > * Refuse to process certain specific audio input format that are actually > returned by the OS for some users (packed PCM audio) > * Drop audio frames, resulting in corrupted audio input. This might have been > unnoticed with video frames but this makes avfoundation essentially unusable > for audio. > > The patches are now being included in our production build so they are tested > and usable in production. > > So, this bares the question: is avfoundation still supported and actively maintained > in libavdevice? It feels that such important bugs should have been noticed by now > and also generated a little more interest in fixing them. > > Thanks for working on this, and addressing all the feedback so far. > > The patchset LGTM, and I think it should be applied. > > Looks like MAINTAINERS lists Thilo for avfoundation.m. I'm not sure if he's seen this yet, so I'm cc'ing on this reply. > > If we don't hear in the next couple weeks, I can apply these changes. Thank you, this is much appreciated! We discovered a bug in the audio converter patch, I’m posting a new updated series right away & will CC everyone here. Thanks! > > > Thanks for y’all feedback! > — Romain > ----- > > Changes: > * v2: None > * v3: None > * v4: None > * v5: Fix indentation/wrapping > * v6: None > > * Implement support for AudioConverter > * Switch to AudioConverter's API to convert unsupported PCM > formats (non-interleaved, non-packed) to supported formats > * Minimize data copy. > > This fixes: https://trac.ffmpeg.org/ticket/9502 > > API ref: > https://developer.apple.com/documentation/audiotoolbox/audio_converter_services > > Signed-off-by: Romain Beauxis <toots@rastageeks.org> > --- > libavdevice/avfoundation.m | 250 +++++++++++++++++++++---------------- > 1 file changed, 144 insertions(+), 106 deletions(-) > > diff --git a/libavdevice/avfoundation.m b/libavdevice/avfoundation.m > index 0cd6e646d5..79c9207cfa 100644 > --- a/libavdevice/avfoundation.m > +++ b/libavdevice/avfoundation.m > @@ -111,16 +111,10 @@ > > int num_video_devices; > > - int audio_channels; > - int audio_bits_per_sample; > - int audio_float; > - int audio_be; > - int audio_signed_integer; > - int audio_packed; > - int audio_non_interleaved; > - > - int32_t *audio_buffer; > - int audio_buffer_size; > + UInt32 audio_buffers; > + UInt32 audio_channels; > + UInt32 bytes_per_sample; > + AudioConverterRef audio_converter; > > enum AVPixelFormat pixel_format; > > @@ -299,7 +293,10 @@ static void destroy_context(AVFContext* ctx) > ctx->avf_delegate = NULL; > ctx->avf_audio_delegate = NULL; > > - av_freep(&ctx->audio_buffer); > + if (ctx->audio_converter) { > + AudioConverterDispose(ctx->audio_converter); > + ctx->audio_converter = NULL; > + } > > pthread_mutex_destroy(&ctx->frame_lock); > > @@ -673,6 +670,10 @@ static int get_audio_config(AVFormatContext *s) > AVFContext *ctx = (AVFContext*)s->priv_data; > CMFormatDescriptionRef format_desc; > AVStream* stream = avformat_new_stream(s, NULL); > + AudioStreamBasicDescription output_format = {0}; > + int audio_bits_per_sample, audio_float, audio_be; > + int audio_signed_integer, audio_packed, audio_non_interleaved; > + int must_convert = 0; > > if (!stream) { > return 1; > @@ -690,60 +691,95 @@ static int get_audio_config(AVFormatContext *s) > avpriv_set_pts_info(stream, 64, 1, avf_time_base); > > format_desc = CMSampleBufferGetFormatDescription(ctx->current_audio_frame); > - const AudioStreamBasicDescription *basic_desc = CMAudioFormatDescriptionGetStreamBasicDescription(format_desc); > + const AudioStreamBasicDescription *input_format = CMAudioFormatDescriptionGetStreamBasicDescription(format_desc); > > - if (!basic_desc) { > + if (!input_format) { > unlock_frames(ctx); > av_log(s, AV_LOG_ERROR, "audio format not available\n"); > return 1; > } > > + if (input_format->mFormatID != kAudioFormatLinearPCM) { > + unlock_frames(ctx); > + av_log(s, AV_LOG_ERROR, "only PCM audio format are supported at the moment\n"); > + return 1; > + } > + > stream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; > - stream->codecpar->sample_rate = basic_desc->mSampleRate; > - stream->codecpar->channels = basic_desc->mChannelsPerFrame; > + stream->codecpar->sample_rate = input_format->mSampleRate; > + stream->codecpar->channels = input_format->mChannelsPerFrame; > stream->codecpar->channel_layout = av_get_default_channel_layout(stream->codecpar->channels); > > - ctx->audio_channels = basic_desc->mChannelsPerFrame; > - ctx->audio_bits_per_sample = basic_desc->mBitsPerChannel; > - ctx->audio_float = basic_desc->mFormatFlags & kAudioFormatFlagIsFloat; > - ctx->audio_be = basic_desc->mFormatFlags & kAudioFormatFlagIsBigEndian; > - ctx->audio_signed_integer = basic_desc->mFormatFlags & kAudioFormatFlagIsSignedInteger; > - ctx->audio_packed = basic_desc->mFormatFlags & kAudioFormatFlagIsPacked; > - ctx->audio_non_interleaved = basic_desc->mFormatFlags & kAudioFormatFlagIsNonInterleaved; > - > - if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_float && > - ctx->audio_bits_per_sample == 32 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_F32BE : AV_CODEC_ID_PCM_F32LE; > - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_signed_integer && > - ctx->audio_bits_per_sample == 16 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_S16BE : AV_CODEC_ID_PCM_S16LE; > - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_signed_integer && > - ctx->audio_bits_per_sample == 24 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_S24BE : AV_CODEC_ID_PCM_S24LE; > - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && > - ctx->audio_signed_integer && > - ctx->audio_bits_per_sample == 32 && > - ctx->audio_packed) { > - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_S32BE : AV_CODEC_ID_PCM_S32LE; > + audio_bits_per_sample = input_format->mBitsPerChannel; > + audio_float = input_format->mFormatFlags & kAudioFormatFlagIsFloat; > + audio_be = input_format->mFormatFlags & kAudioFormatFlagIsBigEndian; > + audio_signed_integer = input_format->mFormatFlags & kAudioFormatFlagIsSignedInteger; > + audio_packed = input_format->mFormatFlags & kAudioFormatFlagIsPacked; > + audio_non_interleaved = input_format->mFormatFlags & kAudioFormatFlagIsNonInterleaved; > + > + ctx->bytes_per_sample = input_format->mBitsPerChannel >> 3; > + ctx->audio_channels = input_format->mChannelsPerFrame; > + > + if (audio_non_interleaved) { > + ctx->audio_buffers = input_format->mChannelsPerFrame; > } else { > - unlock_frames(ctx); > - av_log(s, AV_LOG_ERROR, "audio format is not supported\n"); > - return 1; > + ctx->audio_buffers = 1; > + } > + > + if (audio_non_interleaved || !audio_packed) { > + must_convert = 1; > + } > + > + output_format.mBitsPerChannel = input_format->mBitsPerChannel; > + output_format.mChannelsPerFrame = ctx->audio_channels; > + output_format.mFramesPerPacket = 1; > + output_format.mBytesPerFrame = output_format.mChannelsPerFrame * ctx->bytes_per_sample; > + output_format.mBytesPerPacket = output_format.mFramesPerPacket * output_format.mBytesPerFrame; > + output_format.mFormatFlags = kAudioFormatFlagIsPacked | audio_be; > + output_format.mFormatID = kAudioFormatLinearPCM; > + output_format.mReserved = 0; > + output_format.mSampleRate = input_format->mSampleRate; > + > + if (audio_float && > + audio_bits_per_sample == 32) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_F32BE : AV_CODEC_ID_PCM_F32LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsFloat; > + } else if (audio_float && > + audio_bits_per_sample == 64) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_F64BE : AV_CODEC_ID_PCM_F64LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsFloat; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 8) { > + stream->codecpar->codec_id = AV_CODEC_ID_PCM_S8; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 16) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S16BE : AV_CODEC_ID_PCM_S16LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 24) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S24BE : AV_CODEC_ID_PCM_S24LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 32) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S32BE : AV_CODEC_ID_PCM_S32LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else if (audio_signed_integer && > + audio_bits_per_sample == 64) { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S64BE : AV_CODEC_ID_PCM_S64LE; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + } else { > + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S32BE : AV_CODEC_ID_PCM_S32LE; > + output_format.mBitsPerChannel = 32; > + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; > + must_convert = 1; > } > > - if (ctx->audio_non_interleaved) { > - CMBlockBufferRef block_buffer = CMSampleBufferGetDataBuffer(ctx->current_audio_frame); > - ctx->audio_buffer_size = CMBlockBufferGetDataLength(block_buffer); > - ctx->audio_buffer = av_malloc(ctx->audio_buffer_size); > - if (!ctx->audio_buffer) { > + if (must_convert) { > + OSStatus ret = AudioConverterNew(input_format, &output_format, &ctx->audio_converter); > + if (ret != noErr) { > unlock_frames(ctx); > - av_log(s, AV_LOG_ERROR, "error allocating audio buffer\n"); > + av_log(s, AV_LOG_ERROR, "Error while allocating audio converter\n"); > return 1; > } > } > @@ -1048,6 +1084,7 @@ static int copy_cvpixelbuffer(AVFormatContext *s, > > static int avf_read_packet(AVFormatContext *s, AVPacket *pkt) > { > + OSStatus ret; > AVFContext* ctx = (AVFContext*)s->priv_data; > > do { > @@ -1091,7 +1128,7 @@ static int avf_read_packet(AVFormatContext *s, AVPacket *pkt) > status = copy_cvpixelbuffer(s, image_buffer, pkt); > } else { > status = 0; > - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); > + ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); > if (ret != kCMBlockBufferNoErr) { > status = AVERROR(EIO); > } > @@ -1105,82 +1142,83 @@ static int avf_read_packet(AVFormatContext *s, AVPacket *pkt) > } > } else if (ctx->current_audio_frame != nil) { > CMBlockBufferRef block_buffer = CMSampleBufferGetDataBuffer(ctx->current_audio_frame); > - int block_buffer_size = CMBlockBufferGetDataLength(block_buffer); > > - if (!block_buffer || !block_buffer_size) { > - unlock_frames(ctx); > - return AVERROR(EIO); > - } > + size_t input_size = CMBlockBufferGetDataLength(block_buffer); > + int buffer_size = input_size / ctx->audio_buffers; > + int nb_samples = input_size / (ctx->audio_channels * ctx->bytes_per_sample); > + int output_size = buffer_size; > > - if (ctx->audio_non_interleaved && block_buffer_size > ctx->audio_buffer_size) { > + UInt32 size = sizeof(output_size); > + ret = AudioConverterGetProperty(ctx->audio_converter, kAudioConverterPropertyCalculateOutputBufferSize, &size, &output_size); > + if (ret != noErr) { > unlock_frames(ctx); > - return AVERROR_BUFFER_TOO_SMALL; > + return AVERROR(EIO); > } > > - if (av_new_packet(pkt, block_buffer_size) < 0) { > + if (av_new_packet(pkt, output_size) < 0) { > unlock_frames(ctx); > return AVERROR(EIO); > } > > - CMItemCount count; > - CMSampleTimingInfo timing_info; > + if (ctx->audio_converter) { > + size_t input_buffer_size = offsetof(AudioBufferList, mBuffers[0]) + (sizeof(AudioBuffer) * ctx->audio_buffers); > + AudioBufferList *input_buffer = av_malloc(input_buffer_size); > > - if (CMSampleBufferGetOutputSampleTimingInfoArray(ctx->current_audio_frame, 1, &timing_info, &count) == noErr) { > - AVRational timebase_q = av_make_q(1, timing_info.presentationTimeStamp.timescale); > - pkt->pts = pkt->dts = av_rescale_q(timing_info.presentationTimeStamp.value, timebase_q, avf_time_base_q); > - } > + input_buffer->mNumberBuffers = ctx->audio_buffers; > > - pkt->stream_index = ctx->audio_stream_index; > - pkt->flags |= AV_PKT_FLAG_KEY; > + for (int c = 0; c < ctx->audio_buffers; c++) { > + input_buffer->mBuffers[c].mNumberChannels = 1; > > - if (ctx->audio_non_interleaved) { > - int sample, c, shift, num_samples; > + ret = CMBlockBufferGetDataPointer(block_buffer, c * buffer_size, (size_t *)&input_buffer->mBuffers[c].mDataByteSize, NULL, (void *)&input_buffer->mBuffers[c].mData); > > - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, ctx->audio_buffer); > - if (ret != kCMBlockBufferNoErr) { > - unlock_frames(ctx); > - return AVERROR(EIO); > + if (ret != kCMBlockBufferNoErr) { > + av_free(input_buffer); > + unlock_frames(ctx); > + return AVERROR(EIO); > + } > } > > - num_samples = pkt->size / (ctx->audio_channels * (ctx->audio_bits_per_sample >> 3)); > - > - // transform decoded frame into output format > - #define INTERLEAVE_OUTPUT(bps) \ > - { \ > - int##bps##_t **src; \ > - int##bps##_t *dest; \ > - src = av_malloc(ctx->audio_channels * sizeof(int##bps##_t*)); \ > - if (!src) { \ > - unlock_frames(ctx); \ > - return AVERROR(EIO); \ > - } \ > - \ > - for (c = 0; c < ctx->audio_channels; c++) { \ > - src[c] = ((int##bps##_t*)ctx->audio_buffer) + c * num_samples; \ > - } \ > - dest = (int##bps##_t*)pkt->data; \ > - shift = bps - ctx->audio_bits_per_sample; \ > - for (sample = 0; sample < num_samples; sample++) \ > - for (c = 0; c < ctx->audio_channels; c++) \ > - *dest++ = src[c][sample] << shift; \ > - av_freep(&src); \ > - } > + AudioBufferList output_buffer = { > + .mNumberBuffers = 1, > + .mBuffers[0] = { > + .mNumberChannels = ctx->audio_channels, > + .mDataByteSize = pkt->size, > + .mData = pkt->data > + } > + }; > > - if (ctx->audio_bits_per_sample <= 16) { > - INTERLEAVE_OUTPUT(16) > - } else { > - INTERLEAVE_OUTPUT(32) > - } > - } else { > - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); > - if (ret != kCMBlockBufferNoErr) { > + ret = AudioConverterConvertComplexBuffer(ctx->audio_converter, nb_samples, input_buffer, &output_buffer); > + av_free(input_buffer); > + > + if (ret != noErr) { > unlock_frames(ctx); > return AVERROR(EIO); > } > + > + pkt->size = output_buffer.mBuffers[0].mDataByteSize; > + } else { > + ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); > + if (ret != kCMBlockBufferNoErr) { > + unlock_frames(ctx); > + return AVERROR(EIO); > + } > } > > + CMItemCount count; > + CMSampleTimingInfo timing_info; > + > + if (CMSampleBufferGetOutputSampleTimingInfoArray(ctx->current_audio_frame, 1, &timing_info, &count) == noErr) { > + AVRational timebase_q = av_make_q(1, timing_info.presentationTimeStamp.timescale); > + pkt->pts = pkt->dts = av_rescale_q(timing_info.presentationTimeStamp.value, timebase_q, avf_time_base_q); > + } > + > + pkt->stream_index = ctx->audio_stream_index; > + pkt->flags |= AV_PKT_FLAG_KEY; > + > CFRelease(ctx->current_audio_frame); > ctx->current_audio_frame = nil; > + > + unlock_frames(ctx); > } else { > pkt->data = NULL; > unlock_frames(ctx); > -- > 2.32.0 (Apple Git-132) > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
diff --git a/libavdevice/avfoundation.m b/libavdevice/avfoundation.m index 0cd6e646d5..79c9207cfa 100644 --- a/libavdevice/avfoundation.m +++ b/libavdevice/avfoundation.m @@ -111,16 +111,10 @@ int num_video_devices; - int audio_channels; - int audio_bits_per_sample; - int audio_float; - int audio_be; - int audio_signed_integer; - int audio_packed; - int audio_non_interleaved; - - int32_t *audio_buffer; - int audio_buffer_size; + UInt32 audio_buffers; + UInt32 audio_channels; + UInt32 bytes_per_sample; + AudioConverterRef audio_converter; enum AVPixelFormat pixel_format; @@ -299,7 +293,10 @@ static void destroy_context(AVFContext* ctx) ctx->avf_delegate = NULL; ctx->avf_audio_delegate = NULL; - av_freep(&ctx->audio_buffer); + if (ctx->audio_converter) { + AudioConverterDispose(ctx->audio_converter); + ctx->audio_converter = NULL; + } pthread_mutex_destroy(&ctx->frame_lock); @@ -673,6 +670,10 @@ static int get_audio_config(AVFormatContext *s) AVFContext *ctx = (AVFContext*)s->priv_data; CMFormatDescriptionRef format_desc; AVStream* stream = avformat_new_stream(s, NULL); + AudioStreamBasicDescription output_format = {0}; + int audio_bits_per_sample, audio_float, audio_be; + int audio_signed_integer, audio_packed, audio_non_interleaved; + int must_convert = 0; if (!stream) { return 1; @@ -690,60 +691,95 @@ static int get_audio_config(AVFormatContext *s) avpriv_set_pts_info(stream, 64, 1, avf_time_base); format_desc = CMSampleBufferGetFormatDescription(ctx->current_audio_frame); - const AudioStreamBasicDescription *basic_desc = CMAudioFormatDescriptionGetStreamBasicDescription(format_desc); + const AudioStreamBasicDescription *input_format = CMAudioFormatDescriptionGetStreamBasicDescription(format_desc); - if (!basic_desc) { + if (!input_format) { unlock_frames(ctx); av_log(s, AV_LOG_ERROR, "audio format not available\n"); return 1; } + if (input_format->mFormatID != kAudioFormatLinearPCM) { + unlock_frames(ctx); + av_log(s, AV_LOG_ERROR, "only PCM audio format are supported at the moment\n"); + return 1; + } + stream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; - stream->codecpar->sample_rate = basic_desc->mSampleRate; - stream->codecpar->channels = basic_desc->mChannelsPerFrame; + stream->codecpar->sample_rate = input_format->mSampleRate; + stream->codecpar->channels = input_format->mChannelsPerFrame; stream->codecpar->channel_layout = av_get_default_channel_layout(stream->codecpar->channels); - ctx->audio_channels = basic_desc->mChannelsPerFrame; - ctx->audio_bits_per_sample = basic_desc->mBitsPerChannel; - ctx->audio_float = basic_desc->mFormatFlags & kAudioFormatFlagIsFloat; - ctx->audio_be = basic_desc->mFormatFlags & kAudioFormatFlagIsBigEndian; - ctx->audio_signed_integer = basic_desc->mFormatFlags & kAudioFormatFlagIsSignedInteger; - ctx->audio_packed = basic_desc->mFormatFlags & kAudioFormatFlagIsPacked; - ctx->audio_non_interleaved = basic_desc->mFormatFlags & kAudioFormatFlagIsNonInterleaved; - - if (basic_desc->mFormatID == kAudioFormatLinearPCM && - ctx->audio_float && - ctx->audio_bits_per_sample == 32 && - ctx->audio_packed) { - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_F32BE : AV_CODEC_ID_PCM_F32LE; - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && - ctx->audio_signed_integer && - ctx->audio_bits_per_sample == 16 && - ctx->audio_packed) { - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_S16BE : AV_CODEC_ID_PCM_S16LE; - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && - ctx->audio_signed_integer && - ctx->audio_bits_per_sample == 24 && - ctx->audio_packed) { - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_S24BE : AV_CODEC_ID_PCM_S24LE; - } else if (basic_desc->mFormatID == kAudioFormatLinearPCM && - ctx->audio_signed_integer && - ctx->audio_bits_per_sample == 32 && - ctx->audio_packed) { - stream->codecpar->codec_id = ctx->audio_be ? AV_CODEC_ID_PCM_S32BE : AV_CODEC_ID_PCM_S32LE; + audio_bits_per_sample = input_format->mBitsPerChannel; + audio_float = input_format->mFormatFlags & kAudioFormatFlagIsFloat; + audio_be = input_format->mFormatFlags & kAudioFormatFlagIsBigEndian; + audio_signed_integer = input_format->mFormatFlags & kAudioFormatFlagIsSignedInteger; + audio_packed = input_format->mFormatFlags & kAudioFormatFlagIsPacked; + audio_non_interleaved = input_format->mFormatFlags & kAudioFormatFlagIsNonInterleaved; + + ctx->bytes_per_sample = input_format->mBitsPerChannel >> 3; + ctx->audio_channels = input_format->mChannelsPerFrame; + + if (audio_non_interleaved) { + ctx->audio_buffers = input_format->mChannelsPerFrame; } else { - unlock_frames(ctx); - av_log(s, AV_LOG_ERROR, "audio format is not supported\n"); - return 1; + ctx->audio_buffers = 1; + } + + if (audio_non_interleaved || !audio_packed) { + must_convert = 1; + } + + output_format.mBitsPerChannel = input_format->mBitsPerChannel; + output_format.mChannelsPerFrame = ctx->audio_channels; + output_format.mFramesPerPacket = 1; + output_format.mBytesPerFrame = output_format.mChannelsPerFrame * ctx->bytes_per_sample; + output_format.mBytesPerPacket = output_format.mFramesPerPacket * output_format.mBytesPerFrame; + output_format.mFormatFlags = kAudioFormatFlagIsPacked | audio_be; + output_format.mFormatID = kAudioFormatLinearPCM; + output_format.mReserved = 0; + output_format.mSampleRate = input_format->mSampleRate; + + if (audio_float && + audio_bits_per_sample == 32) { + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_F32BE : AV_CODEC_ID_PCM_F32LE; + output_format.mFormatFlags |= kAudioFormatFlagIsFloat; + } else if (audio_float && + audio_bits_per_sample == 64) { + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_F64BE : AV_CODEC_ID_PCM_F64LE; + output_format.mFormatFlags |= kAudioFormatFlagIsFloat; + } else if (audio_signed_integer && + audio_bits_per_sample == 8) { + stream->codecpar->codec_id = AV_CODEC_ID_PCM_S8; + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; + } else if (audio_signed_integer && + audio_bits_per_sample == 16) { + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S16BE : AV_CODEC_ID_PCM_S16LE; + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; + } else if (audio_signed_integer && + audio_bits_per_sample == 24) { + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S24BE : AV_CODEC_ID_PCM_S24LE; + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; + } else if (audio_signed_integer && + audio_bits_per_sample == 32) { + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S32BE : AV_CODEC_ID_PCM_S32LE; + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; + } else if (audio_signed_integer && + audio_bits_per_sample == 64) { + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S64BE : AV_CODEC_ID_PCM_S64LE; + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; + } else { + stream->codecpar->codec_id = audio_be ? AV_CODEC_ID_PCM_S32BE : AV_CODEC_ID_PCM_S32LE; + output_format.mBitsPerChannel = 32; + output_format.mFormatFlags |= kAudioFormatFlagIsSignedInteger; + must_convert = 1; } - if (ctx->audio_non_interleaved) { - CMBlockBufferRef block_buffer = CMSampleBufferGetDataBuffer(ctx->current_audio_frame); - ctx->audio_buffer_size = CMBlockBufferGetDataLength(block_buffer); - ctx->audio_buffer = av_malloc(ctx->audio_buffer_size); - if (!ctx->audio_buffer) { + if (must_convert) { + OSStatus ret = AudioConverterNew(input_format, &output_format, &ctx->audio_converter); + if (ret != noErr) { unlock_frames(ctx); - av_log(s, AV_LOG_ERROR, "error allocating audio buffer\n"); + av_log(s, AV_LOG_ERROR, "Error while allocating audio converter\n"); return 1; } } @@ -1048,6 +1084,7 @@ static int copy_cvpixelbuffer(AVFormatContext *s, static int avf_read_packet(AVFormatContext *s, AVPacket *pkt) { + OSStatus ret; AVFContext* ctx = (AVFContext*)s->priv_data; do { @@ -1091,7 +1128,7 @@ static int avf_read_packet(AVFormatContext *s, AVPacket *pkt) status = copy_cvpixelbuffer(s, image_buffer, pkt); } else { status = 0; - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); + ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); if (ret != kCMBlockBufferNoErr) { status = AVERROR(EIO); } @@ -1105,82 +1142,83 @@ static int avf_read_packet(AVFormatContext *s, AVPacket *pkt) } } else if (ctx->current_audio_frame != nil) { CMBlockBufferRef block_buffer = CMSampleBufferGetDataBuffer(ctx->current_audio_frame); - int block_buffer_size = CMBlockBufferGetDataLength(block_buffer); - if (!block_buffer || !block_buffer_size) { - unlock_frames(ctx); - return AVERROR(EIO); - } + size_t input_size = CMBlockBufferGetDataLength(block_buffer); + int buffer_size = input_size / ctx->audio_buffers; + int nb_samples = input_size / (ctx->audio_channels * ctx->bytes_per_sample); + int output_size = buffer_size; - if (ctx->audio_non_interleaved && block_buffer_size > ctx->audio_buffer_size) { + UInt32 size = sizeof(output_size); + ret = AudioConverterGetProperty(ctx->audio_converter, kAudioConverterPropertyCalculateOutputBufferSize, &size, &output_size); + if (ret != noErr) { unlock_frames(ctx); - return AVERROR_BUFFER_TOO_SMALL; + return AVERROR(EIO); } - if (av_new_packet(pkt, block_buffer_size) < 0) { + if (av_new_packet(pkt, output_size) < 0) { unlock_frames(ctx); return AVERROR(EIO); } - CMItemCount count; - CMSampleTimingInfo timing_info; + if (ctx->audio_converter) { + size_t input_buffer_size = offsetof(AudioBufferList, mBuffers[0]) + (sizeof(AudioBuffer) * ctx->audio_buffers); + AudioBufferList *input_buffer = av_malloc(input_buffer_size); - if (CMSampleBufferGetOutputSampleTimingInfoArray(ctx->current_audio_frame, 1, &timing_info, &count) == noErr) { - AVRational timebase_q = av_make_q(1, timing_info.presentationTimeStamp.timescale); - pkt->pts = pkt->dts = av_rescale_q(timing_info.presentationTimeStamp.value, timebase_q, avf_time_base_q); - } + input_buffer->mNumberBuffers = ctx->audio_buffers; - pkt->stream_index = ctx->audio_stream_index; - pkt->flags |= AV_PKT_FLAG_KEY; + for (int c = 0; c < ctx->audio_buffers; c++) { + input_buffer->mBuffers[c].mNumberChannels = 1; - if (ctx->audio_non_interleaved) { - int sample, c, shift, num_samples; + ret = CMBlockBufferGetDataPointer(block_buffer, c * buffer_size, (size_t *)&input_buffer->mBuffers[c].mDataByteSize, NULL, (void *)&input_buffer->mBuffers[c].mData); - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, ctx->audio_buffer); - if (ret != kCMBlockBufferNoErr) { - unlock_frames(ctx); - return AVERROR(EIO); + if (ret != kCMBlockBufferNoErr) { + av_free(input_buffer); + unlock_frames(ctx); + return AVERROR(EIO); + } } - num_samples = pkt->size / (ctx->audio_channels * (ctx->audio_bits_per_sample >> 3)); - - // transform decoded frame into output format - #define INTERLEAVE_OUTPUT(bps) \ - { \ - int##bps##_t **src; \ - int##bps##_t *dest; \ - src = av_malloc(ctx->audio_channels * sizeof(int##bps##_t*)); \ - if (!src) { \ - unlock_frames(ctx); \ - return AVERROR(EIO); \ - } \ - \ - for (c = 0; c < ctx->audio_channels; c++) { \ - src[c] = ((int##bps##_t*)ctx->audio_buffer) + c * num_samples; \ - } \ - dest = (int##bps##_t*)pkt->data; \ - shift = bps - ctx->audio_bits_per_sample; \ - for (sample = 0; sample < num_samples; sample++) \ - for (c = 0; c < ctx->audio_channels; c++) \ - *dest++ = src[c][sample] << shift; \ - av_freep(&src); \ - } + AudioBufferList output_buffer = { + .mNumberBuffers = 1, + .mBuffers[0] = { + .mNumberChannels = ctx->audio_channels, + .mDataByteSize = pkt->size, + .mData = pkt->data + } + }; - if (ctx->audio_bits_per_sample <= 16) { - INTERLEAVE_OUTPUT(16) - } else { - INTERLEAVE_OUTPUT(32) - } - } else { - OSStatus ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); - if (ret != kCMBlockBufferNoErr) { + ret = AudioConverterConvertComplexBuffer(ctx->audio_converter, nb_samples, input_buffer, &output_buffer); + av_free(input_buffer); + + if (ret != noErr) { unlock_frames(ctx); return AVERROR(EIO); } + + pkt->size = output_buffer.mBuffers[0].mDataByteSize; + } else { + ret = CMBlockBufferCopyDataBytes(block_buffer, 0, pkt->size, pkt->data); + if (ret != kCMBlockBufferNoErr) { + unlock_frames(ctx); + return AVERROR(EIO); + } } + CMItemCount count; + CMSampleTimingInfo timing_info; + + if (CMSampleBufferGetOutputSampleTimingInfoArray(ctx->current_audio_frame, 1, &timing_info, &count) == noErr) { + AVRational timebase_q = av_make_q(1, timing_info.presentationTimeStamp.timescale); + pkt->pts = pkt->dts = av_rescale_q(timing_info.presentationTimeStamp.value, timebase_q, avf_time_base_q); + } + + pkt->stream_index = ctx->audio_stream_index; + pkt->flags |= AV_PKT_FLAG_KEY; + CFRelease(ctx->current_audio_frame); ctx->current_audio_frame = nil; + + unlock_frames(ctx); } else { pkt->data = NULL; unlock_frames(ctx);
This is the first patch of a series of 3 that fix, cleanup and enhance the avfoundation implementation for libavdevice. The patches have been submitted a couple of times now and have received very nice feedback for the last two however but they do not seem to have been considered for inclusion thus far. These patches come from an actual user-facing application relying on libavdevice’s implementation of avfoundation audio input. Without them, Avfoundation is practically unusable as it will: * Refuse to process certain specific audio input format that are actually returned by the OS for some users (packed PCM audio) * Drop audio frames, resulting in corrupted audio input. This might have been unnoticed with video frames but this makes avfoundation essentially unusable for audio. The patches are now being included in our production build so they are tested and usable in production. So, this bares the question: is avfoundation still supported and actively maintained in libavdevice? It feels that such important bugs should have been noticed by now and also generated a little more interest in fixing them. Thanks for y’all feedback! — Romain ----- Changes: * v2: None * v3: None * v4: None * v5: Fix indentation/wrapping * v6: None * Implement support for AudioConverter * Switch to AudioConverter's API to convert unsupported PCM formats (non-interleaved, non-packed) to supported formats * Minimize data copy. This fixes: https://trac.ffmpeg.org/ticket/9502 API ref: https://developer.apple.com/documentation/audiotoolbox/audio_converter_services Signed-off-by: Romain Beauxis <toots@rastageeks.org> --- libavdevice/avfoundation.m | 250 +++++++++++++++++++++---------------- 1 file changed, 144 insertions(+), 106 deletions(-)